- sr-users - kamailio.org

[Users] RTPPRXY or MEDIA Proxy
by ram 21 Jun '06

21 Jun '06

Hi all i was reading the documentation just i was confused to use Media proxy or rtp proxy when i run openser its using default rtpproxy what is the difference between then what is the nest to use for NAT Clients. suggestion will be appriciated. ram

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[Serusers] (no subject)
by Aman Sachdeva 21 Jun '06

21 Jun '06

sir, thanks for ur replly. sit i have problem that when i see the database of ser i could not get any entry in acc table.i call many time but it could not store in acc table. location of user stred in ser database but user calls is not stored in ser database .please help me so that i can solve this problem. thanks sachdeva.aman(a)gmail.com

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[Serusers] could not rev. resolve IP......
by Shenen Shenen 21 Jun '06

21 Jun '06

Hi!This my openser.cfg;I've 2 asteriskathome with openser installed;I want to copy the authentificated sip registration from one box to another and have always the same Sip registered in all 2 box, so if one box falls down the other can run without reinitialized all the sips (like Phoner..cubix...and also IP phones...)....I'm using vrrpd...12 is the master, 11 is the slave, 200 is the virtual IP(this point to the master in the first case..and all Sip proxies like the phoner or cubix points to the master..(I can set only 1 proxy.. 192.168.251.200 ..in x-lite I can set more proxies and I'vent problems..)(for this I must use openser..) This is the openser on the "slave" 192.168.251.11 # SCRIPT PER COPIARE LO STATO DELLE REGISTRAZIONI DEI SIP DAL .11 al .12 # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=no # (cmd line: -E) listen=192.168.251.12 listen=192.168.251.11 listen=192.1668.251.200 port=5060 children=5 dns=no rev_dns=no # ------------------ module loading ---------------------------------- loadmodule "modules/mysql/mysql.so" loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/usrloc/usrloc.so" loadmodule "modules/registrar/registrar.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); return; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # verify credentials if (!www_authorize("", "subscriber")) { www_challenge("", "0"); return; }; # if ok, update contacts and ... save("location"); # ... if this REGISTER is not a replica from our # peer server, replicate to the peer server if (!src_ip== 192.168.251.12) { t_replicate("192.168.251.12", "5060"); }; return; }; # do whatever else appropriate for your domain log("non-REGISTER\n"); }; } This is the openser on the "master" 192.168.251.12 # SCRIPT PER COPIARE LO STATO DELLE REGISTRAZIONI DEI SIP DAL .12 al .11 # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) listen= 192.168.251.12 listen=192.168.251.11 listen=192.1668.251.200 port=5060 children=5 dns=no rev_dns=no # ------------------ module loading ---------------------------------- loadmodule "modules/mysql/mysql.so" loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/usrloc/usrloc.so" loadmodule "modules/registrar/registrar.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); return; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # verify credentials if (!www_authorize("", "subscriber")) { www_challenge("", "0"); return; }; # if ok, update contacts and ... save("location"); # ... if this REGISTER is not a replica from our # peer server, replicate to the peer server if (!src_ip== 192.168.251.11) { t_replicate("192.168.251.11", "5060"); }; return; }; # do whatever else appropriate for your domain log("non-REGISTER\n"); }; } If I run the following command... I've problems with resolution of IP addresses... how can I resolve this problem?????????? 100000000000000000000000000000000000000000000000000000 thanks.....................help me please!!!!!I'm becoming crazy... [root@asterisk12 ~]# tail -f /var/log/messages Jun 21 03:33:43 asterisk12 openser: WARNING: fix_socket_list: could not rev. resolve 192.168.251.11 Jun 21 03:33:43 asterisk12 openser: WARNING: fix_socket_list: could not rev. resolve 192.168.251.200 Jun 21 03:33:43 asterisk12 openser[19037]: Maxfwd module- initializing Jun 21 03:33:43 asterisk12 openser[19037]: AUTH module - initializing Jun 21 03:33:43 asterisk12 openser[19037]: AUTH_DB module - initializing Jun 21 03:33:43 asterisk12 openser[19037]: INFO: udp_init: SO_RCVBUF is initially 110592 Jun 21 03:33:43 asterisk12 openser[19037]: INFO: udp_init: SO_RCVBUF is finally 221184 Jun 21 03:33:43 asterisk12 openser[19037]: INFO: udp_init: SO_RCVBUF is initially 110592 Jun 21 03:33:43 asterisk12 openser[19037]: INFO: udp_init: SO_RCVBUF is finally 221184 Jun 21 03:33:43 asterisk12 openser[19037]: ERROR: udp_init: bind(6, 0x8119a1c, 16) on 192.168.251.11 : Cannot assign requested address Jun 21 03:34:01 asterisk12 crond(pam_unix)[19040]: session opened for user root by (uid=0) Jun 21 03:34:01 asterisk12 crond(pam_unix)[19040]: session closed for user root

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RE: [Serusers] checking source ip and dest ip
by İlker Aktuna (Koç.net) 21 Jun '06

21 Jun '06

Hi, In fact my network is not 10.0.0.0 ; that was just an example (bad example). The network addresses are specific public network addresses (lets say 3 different C Class network) and has no relation to NAT. That's why I need to check the IP adrresses (from and to) of the SIP message before deciding if it should use mediaproxy or not. Any idea how to do that ? Soruce address can be checked as : if (src_ip="195.1.1.+") { } but that won't be same for destination address as "dst_ip" is the SER server ip. How can I check the ip address of the client that the SIP message will be forwarded ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 10:26 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Yes, and your network is in the RFC1918 (private) address space, so you can use the same logic (even without change as you want to proxy when the address is on your network). g-) İlker Aktuna (Koç.net) wrote: In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, In fact my network is not 10.0.0.0 ; that was just an example (bad example). The network addresses are specific public network addresses (lets say 3 different C Class network) and has no relation to NAT. That's why I need to check the IP adrresses (from and to) of the SIP message before deciding if it should use mediaproxy or not. Any idea how to do that ? Soruce address can be checked as : if (src_ip="195.1.1.+") { } but that won't be same for destination address as "dst_ip" is the SER server ip. How can I check the ip address of the client that the SIP message will be forwarded ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 10:26 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Yes, and your network is in the RFC1918 (private) address space, so you can use the same logic (even without change as you want to proxy when the address is on your network). g-) İlker Aktuna (Koç.net) wrote: In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/0B01060D/07044C00/09080544/001…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. 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RE: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723
by Benjamin.George＠t-systems.com 21 Jun '06

21 Jun '06

Hi, In my understanding, you can't configure media proxy for g729 as it won't support g729 codec. It is a patented codec and SER media proxy as well as SEMS supports only the patent free codecs like g711a/u or iLBC, etc. As a result of this you will get some choppy sound only if you use g729. So I feel, you have to modify the code in order to incorporate g729 with media proxy. Regards, Benjamin. _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ravi reddy Sent: Wednesday, June 21, 2006 1:02 PM To: serusers(a)lists.iptel.org Subject: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723 Hi , I am using ser-0.9.6 and mediaproxy-1.4.2 and the are working fine til now but when i use the sip-sip phone (or) sip - pstn in the peak hours i cant hear voice very clearly its choppy sound , when i tried to call grandstream sip phone to pstn i made grandstream phone to use only g-729 but mediaproxy is showing session in active can any body tell me how to change mediaproxy's g-711a to some thing like g-729 or g-723 so that i can hear clear voice in peak hours too thanks you very much regards ravi.

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[Serusers] How to start a VOIP business ?
by Concy Pereira 21 Jun '06

21 Jun '06

Thanks for your inputs, Does anybody has a complete document to setup all this.what I mean a step by step guide. Reggards, Concy Message: 16 Date: Wed, 21 Jun 2006 10:21:15 +1000 From: Nick Hoffman <nick.hoffman(a)altcall.com> Subject: Re: [Serusers] How to start a VOIP business ? To: serusers(a)lists.iptel.org Message-ID: <200606211021.16081.nick.hoffman(a)altcall.com> Content-Type: text/plain; charset="iso-8859-1" > Concy Pereira wrote: > > Sir, > > > > I would like to know how can start a successful SIP base VOIP > > business. I have being going through some documentation of Open server > > software but could not come to a conclusion. > > > > Anybody can suggest me a complete solution for running PC to Phone > > service Using GPL products, with accounting, authentication and > > authorization with different rate Plans etc. > > > > Here is what I have decided to do. > > > > SIP Server (Hardware) > > 1. Linux RedHat 9.0 > > 2. OpenSer or SIP express router > > 3. Radius client > > > > Accounting /CDR server ( Hardware) > > 1. Linux Redhat 9.0 > > 2. CDRTool by ag-projects http://www.ag-projects.com/cdrtool.html > > 3. MySQL > > > > 1.Carrirer > > Subcription to any International whole sale VOIP carrier > > > > Please suggest if I have missed anything out. > > > > Regards, > > Concy On Mon June 19 2006 18:21, "Greger V. Teigre" <greger(a)teigre.com> wrote: > It's a fairly big questions you're asking. But basically you're onto it > ;-) Ad 1. You should go for something newer than 9.0 > Ad 2. You need to decide on your maintenance schedule and policies for > upgrades/patching etc. Many larger-scale deployments are based on SER > 0.9.x because a) it has proven extremely stable b) it is currently > maintained for bug and security fixes and will be for a while. Others > choose openser to get more features, but you need to make sure that you > can handle testing and deployment of new versions more often. > Ad 3. You need a RADIUS server with a user database or you can use mysql > (you need mysql regardless of using RADIUS or not) > > g-) Hi there Greger. I'm curious as to why a RADIUS server is required. Is this for billing purposes? -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ------------------------------ Message: 17 Date: Wed, 21 Jun 2006 10:23:49 +1000 From: Nick Hoffman <nick.hoffman(a)altcall.com> Subject: Re: [Serusers] How to start a VOIP business ? To: serusers(a)lists.iptel.org Cc: Benjamin.George(a)t-systems.com Message-ID: <200606211023.49237.nick.hoffman(a)altcall.com> Content-Type: text/plain; charset="iso-8859-1" On Mon June 19 2006 18:48, Benjamin.George(a)t-systems.com wrote: > Hi, > > I too can provide you some inputs on this apart from the things which > are already mentioned: > > 1. You may require a media server for playing prompts, announcements, > etc 2. Voice mail server to store and play back voice mails > 3. Signaling + Media Gateway if you want to connect to PSTN world > 4. Any good performance management/monitoring tool (normally part of > OSS) to measure and manage the performance of the entire infrastructure > to assure zero/minimum downtime > > Regards, > Benjamin. Hi Benjamin. What would you recommend for your 3rd point? Cheers, -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ------------------------------ Message: 18 Date: Wed, 21 Jun 2006 09:05:20 +0800 From: Ryan Pagquil <rpagquil(a)philonline.com> Subject: Re: [Serusers] How to start a VOIP business ? To: nick.hoffman@altcall.com,serusers@lists.iptel.org Message-ID: <7.0.1.0.0.20060621090427.01b0fa20(a)philonline.com> Content-Type: text/plain; charset="us-ascii"; format=flowed Hi, To easily manage accounting and authorization use radius. I'm using radius for these purposes. regards, Ryan At 08:21 AM 6/21/2006, Nick Hoffman wrote: > > Concy Pereira wrote: > > > Sir, > > > > > > I would like to know how can start a successful SIP base VOIP > > > business. I have being going through some documentation of Open server > > > software but could not come to a conclusion. > > > > > > Anybody can suggest me a complete solution for running PC to Phone > > > service Using GPL products, with accounting, authentication and > > > authorization with different rate Plans etc. > > > > > > Here is what I have decided to do. > > > > > > SIP Server (Hardware) > > > 1. Linux RedHat 9.0 > > > 2. OpenSer or SIP express router > > > 3. Radius client > > > > > > Accounting /CDR server ( Hardware) > > > 1. Linux Redhat 9.0 > > > 2. CDRTool by ag-projects http://www.ag-projects.com/cdrtool.html > > > 3. MySQL > > > > > > 1.Carrirer > > > Subcription to any International whole sale VOIP carrier > > > > > > Please suggest if I have missed anything out. > > > > > > Regards, > > > Concy > > >On Mon June 19 2006 18:21, "Greger V. Teigre" <greger(a)teigre.com> wrote: > > It's a fairly big questions you're asking. But basically you're onto it > > ;-) Ad 1. You should go for something newer than 9.0 > > Ad 2. You need to decide on your maintenance schedule and policies for > > upgrades/patching etc. Many larger-scale deployments are based on SER > > 0.9.x because a) it has proven extremely stable b) it is currently > > maintained for bug and security fixes and will be for a while. Others > > choose openser to get more features, but you need to make sure that you > > can handle testing and deployment of new versions more often. > > Ad 3. You need a RADIUS server with a user database or you can use mysql > > (you need mysql regardless of using RADIUS or not) > > > > g-) > > >Hi there Greger. I'm curious as to why a RADIUS server is required. Is this >for billing purposes? >-- Nick >e: nick.hoffman(a)altcall.com >p: +61 7 5591 3588 >f: +61 7 5591 6588 > >If you receive this email by mistake, please notify us and do not make any >use of the email. We do not waive any privilege, confidentiality or >copyright associated with it. >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers ------------------------------ Message: 19 Date: Wed, 21 Jun 2006 07:14:43 +0200 From: Benjamin.George(a)t-systems.com Subject: RE: [Serusers] How to start a VOIP business ? To: nick.hoffman(a)altcall.com, serusers(a)lists.iptel.org Message-ID: <5BDFB528F5AD4744B933D3C6186DA27F02CD11CB(a)S4DE8PSAADL.t-systems.com> Content-Type: text/plain Hi, I would recommend any of the following for the aforesaid: 1. Cisco media gateway (Router can be converted to a media gateway using voice interface cards). We use Cisco 2600 and 7200 (with MPLS support) series for this functionality. 2. NMS media gateway Some other market leading products are also available for this purpose, for example products from Lucent, Alcatel, etc. Regards, Benjamin. -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Nick Hoffman Sent: Wednesday, June 21, 2006 5:54 AM To: serusers(a)lists.iptel.org Cc: George, Benjamin Subject: Re: [Serusers] How to start a VOIP business ? On Mon June 19 2006 18:48, Benjamin.George(a)t-systems.com wrote: > Hi, > > I too can provide you some inputs on this apart from the things which > are already mentioned: > > 1. You may require a media server for playing prompts, announcements, > etc 2. Voice mail server to store and play back voice mails > 3. Signaling + Media Gateway if you want to connect to PSTN world > 4. Any good performance management/monitoring tool (normally part of > OSS) to measure and manage the performance of the entire infrastructure > to assure zero/minimum downtime > > Regards, > Benjamin. Hi Benjamin. What would you recommend for your 3rd point? Cheers, -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ------------------------------ Message: 20 Date: Wed, 21 Jun 2006 08:42:14 +0200 From: "Greger V. Teigre" <greger(a)teigre.com> Subject: Re: [Serusers] How to start a VOIP business ? To: Ryan Pagquil <rpagquil(a)philonline.com> Cc: serusers(a)lists.iptel.org Message-ID: <4498EA46.5090605(a)teigre.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Yes, RADIUS is also quite efficient for authentication and is a very good interface into a multitude of user databases ranging from unix user accounts, Active Directory, LDAP, any sql database etc etc. (And Concy listed a RADIUS *client* as a requirement. I just corrected that to a server. I'm not saying that you should use RADIUS...) g-) Ryan Pagquil wrote: > Hi, > To easily manage accounting and authorization use radius. I'm > using radius for these purposes. > > regards, > Ryan > > At 08:21 AM 6/21/2006, Nick Hoffman wrote: >> > Concy Pereira wrote: >> > > Sir, >> > > >> > > I would like to know how can start a successful SIP base VOIP >> > > business. I have being going through some documentation of Open >> server >> > > software but could not come to a conclusion. >> > > >> > > Anybody can suggest me a complete solution for running PC to Phone >> > > service Using GPL products, with accounting, authentication and >> > > authorization with different rate Plans etc. >> > > >> > > Here is what I have decided to do. >> > > >> > > SIP Server (Hardware) >> > > 1. Linux RedHat 9.0 >> > > 2. OpenSer or SIP express router >> > > 3. Radius client >> > > >> > > Accounting /CDR server ( Hardware) >> > > 1. Linux Redhat 9.0 >> > > 2. CDRTool by ag-projects http://www.ag-projects.com/cdrtool.html >> > > 3. MySQL >> > > >> > > 1.Carrirer >> > > Subcription to any International whole sale VOIP carrier >> > > >> > > Please suggest if I have missed anything out. >> > > >> > > Regards, >> > > Concy >> >> >> On Mon June 19 2006 18:21, "Greger V. Teigre" <greger(a)teigre.com> wrote: >> > It's a fairly big questions you're asking. But basically you're >> onto it >> > ;-) Ad 1. You should go for something newer than 9.0 >> > Ad 2. You need to decide on your maintenance schedule and policies for >> > upgrades/patching etc. Many larger-scale deployments are based on SER >> > 0.9.x because a) it has proven extremely stable b) it is currently >> > maintained for bug and security fixes and will be for a while. Others >> > choose openser to get more features, but you need to make sure that >> you >> > can handle testing and deployment of new versions more often. >> > Ad 3. You need a RADIUS server with a user database or you can use >> mysql >> > (you need mysql regardless of using RADIUS or not) >> > >> > g-) >> >> >> Hi there Greger. I'm curious as to why a RADIUS server is required. >> Is this >> for billing purposes? >> -- Nick >> e: nick.hoffman(a)altcall.com >> p: +61 7 5591 3588 >> f: +61 7 5591 6588 >> >> If you receive this email by mistake, please notify us and do not >> make any >> use of the email. We do not waive any privilege, confidentiality or >> copyright associated with it. >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ------------------------------

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RE: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723
by İlker Aktuna (Koç.net) 21 Jun '06

21 Jun '06

I think mediaproxy can't do transcoding. Can anyone confirm this ? Thanks, ilker -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kouprianov Sent: Wednesday, June 21, 2006 11:31 AM To: serusers(a)iptel.org Subject: Re: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723 How about making clients change the default codec to smth else? On 6/21/06, Benjamin.George(a)t-systems.com <Benjamin.George(a)t-systems.com> wrote: > > > > Hi, > > In my understanding, you can't configure media proxy for g729 as it > won't support g729 codec. It is a patented codec and SER media proxy > as well as SEMS supports only the patent free codecs like g711a/u or > iLBC, etc. As a result of this you will get some choppy sound only if > you use g729. So I feel, you have to modify the code in order to > incorporate g729 with media proxy. > > Regards, > > Benjamin. > > > ________________________________ > > > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ravi reddy > Sent: Wednesday, June 21, 2006 1:02 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] How to change Mediaproxy codec g-711a to g-729 or > g-723 > > > > > Hi , > I am using ser-0.9.6 and mediaproxy-1.4.2 and the are > working fine til now but when i use the sip-sip phone (or) sip - pstn > in the peak hours i cant hear voice very clearly its choppy sound , > > when i tried to call grandstream sip phone to pstn i made > grandstream phone to use only g-729 but mediaproxy is showing session > in active > > can any body tell me how to change mediaproxy's g-711a to some thing > like > g-729 or g-723 so that i can hear clear voice in peak hours too > > > thanks you very much > > regards > ravi. > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers <http://387555.sigclick.mailinfo.com/sigclick/000A0607/07074800/0203074E/001…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723
by ravi reddy 21 Jun '06

21 Jun '06

Hi , I am using ser-0.9.6 and mediaproxy-1.4.2 and the are working fine til now but when i use the sip-sip phone (or) sip - pstn in the peak hours i cant hear voice very clearly its choppy sound , when i tried to call grandstream sip phone to pstn i made grandstream phone to use only g-729 but mediaproxy is showing session in active can any body tell me how to change mediaproxy's g-711a to some thing like g-729 or g-723 so that i can hear clear voice in peak hours too thanks you very much regards ravi.

1 0

[Serusers] NOTIFY and WATCHERINFO_NOTIFY
by Alois Schafferhofer 21 Jun '06

21 Jun '06

Hi, I have some questions about PUBLISH handling. SER sends a lot of NOTIFYS when status changes. Additionally, SER sends more than one tuples in one Response (open and closed in one Response). The second question is about WATCHERINFO_NOTIFY. I have the line: modparam("pa", "watcherinfo_notify", 1) in my config file, but SER doesn't NOTIFY me, if someone SUBSCRIBEs to my Presence information. Regards, Alois 10.40.1.42:5060 -> 10.15.200.57:5060 SUBSCRIBE sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.40.1.42:5060;rport;branch=z9hG4bK92504..Max-Forwards: 70..To: <sip:PDA_User@fh-joan neum.at>..From: <sip:esca@fh-joanneum.at>;tag=z9hG4bK19686534..Call-ID: 035185567140@10.40.1.42..CSeq: 1 SUBSCRIBE..Contact: <sip:esca@10.40.1.42>..Ex pires: 3600..User-Agent: mjsip stack 1.6..Event: presence..Accept: application/cpim-pidf+xml..Content-Length: 0.... # 10.15.200.57:5060 -> 10.40.1.42:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.40.1.42:5060;rport=5060;branch=z9hG4bK92504..To: <sip:PDA_User@fhjoanneum. at>;tag=589230a54cd08b3768529724501fae2 8-51f5..From: <sip:esca@fh-joanneum.at>;tag=z9hG4bK19686534..Call-ID: 035185567140@10.40.1.42..CSeq: 1 SUBSCRIBE..Expires: 3600..Contact: <sip:10.15.2 00.57:5060>..Server: Sip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: 0..Warning: 392 10.15.200.57:5060 "Noisy feedback tells: pid=7288 req_src_ip=10.40.1.42 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1".... # 10.15.200.57:5060 -> 10.40.1.42:5060 NOTIFY sip:esca@10.40.1.42 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bKc1d8.7c27cd7.0..To: <sip:esca@fhjoanneum. at>;tag=z9hG4bK19686534..Fro m: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-51f5..CSeq: 1 NOTIFY..Call-ID: 035185567140@10.40.1.42..Content-Length: 305..Use r-Agent: Sip EXpress router(0.10.99-dev35-pa-4.1 (i386/linux))..Event: presence..Content-Type: application/cpim-pidf+xml..Contact: <sip:10.15.200.57:5 060>..Subscription-State: active;expires=3600....<?xml version="1.0" encoding="UTF-8"?>..<presence xmlns="urn:ietf:params:xml:ns:cpim-pidf" entity="pr es:PDA_User@fh-joanneum.at">...<tuple id="0x409c3328x557f3c98x44970bc3">....<status><basic>open</basic></status>....<contact priority="0.00">sip:pda_u ser(a)fh-joanneum.at</contact>...</tuple>..</presence>.. # 10.40.1.42:5060 -> 10.15.200.57:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bKc1d8.7c27cd7.0..To: <sip:esca@fh-joanneum.at>;tag=z9hG4bK19686534..From: <sip:PDA_User@fhjoanneum. at>;tag=589230a54cd08b3768529724501fae28-51f5..Call-ID: 035185567140@10.40.1.42..CSeq: 1 NOTIFY..Server: mjsip stack 1.6..Content-Length: 0. 10.0.0.3:5060 -> 10.0.0.11:5060 PUBLISH sip:pda_user@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1286239759..Max-Forwards: 70..From: <sip:pda_user@fh-joanneu m.at>;tag=264505054..To: <sip:pda_user@fh-joanneum.at>..Call-ID: 2090602137-1913544447-137194582..CSeq: 2 PUBLISH..Contact: <sip:pda_user@fh-joanneum. at>..Content-Type: application/cpim-pidf+xml..Content-Length: 219..Expires: 7200..Event: presence..User-Agent: SIP .NET 1.0 evaluation version, www.in dependentsoft.com....<?xml version="1.0" encoding="utf-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf"entity="pres:pda_user@fhjoanneum. at"><tuple i d="0x409c3328x557f3c98x44970bc3"><status><basic>closed</basic></status></tuple></presence>.. # 10.0.0.11:5060 -> 10.0.0.3:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1286239759..From: <sip:pda_user@fh-joanneum.at>;tag=264505054..To: <sip:pda_user@fh-joann eum.at>;tag=589230a54cd08b3768529724501fae28-f3b7..Call-ID: 2090602137-1913544447-137194582..CSeq: 2 PUBLISH..Expires: 120..SIPETag: 0x409c2714x56983 668x449711ab..Contact: <sip:10.0.0.11:5060>..Server: Sip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11 :5060 "Noisy feedback tells: pid=7293 req_src_ip=10.0.0.3 req_src_port=5060 in_uri=sip:pda_user@fh-joanneum.at out_uri=sip:pda_user@fhjoanneum. at vi a_cnt==1" 10.15.200.57:5060 -> 10.40.1.42:5060 NOTIFY sip:esca@10.40.1.42 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK91d8.2831e0b1.0..To: <sip:esca@fhjoanneum. at>;tag=z9hG4bK19686534..Fr om: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-51f5..CSeq: 2 NOTIFY..Call-ID: 035185567140@10.40.1.42..Content-Length: 441..Us er-Agent: Sip EXpress router(0.10.99-dev35-pa-4.1 (i386/linux))..Event: presence..Content-Type: application/cpim-pidf+xml..Contact: <sip:10.15.200.57: 5060>..Subscription-State: active;expires=3545....<?xml version="1.0" encoding="UTF-8"?>..<presence xmlns="urn:ietf:params:xml:ns:cpim-pidf" entity="p res:PDA_User@fh-joanneum.at">...<tuple id="0x409c45e0x5dd5903fx449711ab">....<status><basic>closed</basic></status>....<contact priority="0.00"></cont act>...</tuple>...<tuple id="0x409c3328x557f3c98x44970bc3">....<status><basic>open</basic></status>....<contact priority="0.00">sip:pda_user@fh-joanne um.at</contact>...</tuple>..</presence>.. # 10.15.200.57:5060 -> 10.40.1.42:5060 NOTIFY sip:esca@10.40.1.42 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK23e1.c6b3a835.0..To: <sip:esca@fhjoanneum. at>;tag=z9hG4bK13311114..Fr om: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-4a9d..CSeq: 6 NOTIFY..Call-ID: 480873203923@10.40.1.42..Content-Length: 441..Us er-Agent: Sip EXpress router(0.10.99-dev35-pa-4.1 (i386/linux))..Event: presence..Content-Type: application/cpim-pidf+xml..Contact: <sip:10.15.200.57: 5060>..Subscription-State: active;expires=2198....<?xml version="1.0" encoding="UTF-8"?>..<presence xmlns="urn:ietf:params:xml:ns:cpim-pidf" entity="p res:PDA_User@fh-joanneum.at">...<tuple id="0x409c45e0x5dd5903fx449711ab">....<status><basic>closed</basic></status>....<contact priority="0.00"></cont act>...</tuple>...<tuple id="0x409c3328x557f3c98x44970bc3">....<status><basic>open</basic></status>....<contact priority="0.00">sip:pda_user@fh-joanne um.at</contact>...</tuple>..</presence>.. # 10.15.200.57:5060 -> 10.40.1.42:5060 NOTIFY sip:esca@10.40.1.42 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bKe636.2a57bde3.0..To: <sip:esca@fhjoanneum. at>;tag=z9hG4bK49649677..Fr om: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-9d18..CSeq: 12 NOTIFY..Call-ID: 047322370278@10.40.1.42..Content-Length: 441..U ser-Agent: Sip EXpress router(0.10.99-dev35-pa-4.1 (i386/linux))..Event: presence..Content-Type: application/cpim-pidf+xml..Contact: <sip:10.15.200.57 :5060>..Subscription-State: active;expires=1614....<?xml version="1.0" encoding="UTF-8"?>..<presence xmlns="urn:ietf:params:xml:ns:cpim-pidf" entity=" pres:PDA_User@fh-joanneum.at">...<tuple id="0x409c45e0x5dd5903fx449711ab">....<status><basic>closed</basic></status>....<contact priority="0.00"></con tact>...</tuple>...<tuple id="0x409c3328x557f3c98x44970bc3">....<status><basic>open</basic></status>....<contact priority="0.00">sip:pda_user@fh-joann eum.at</contact>...</tuple>..</presence>.. # 10.15.200.57:5060 -> 10.40.1.42:5060 NOTIFY sip:esca@10.40.1.42 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bKe7b3.ac386935.0..To: <sip:esca@fhjoanneum. at>;tag=z9hG4bK24602857..Fr om: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-52ae..CSeq: 19 NOTIFY..Call-ID: 310599367527@10.40.1.42..Content-Length: 441..U ser-Agent: Sip EXpress router(0.10.99-dev35-pa-4.1 (i386/linux))..Event: presence..Content-Type: application/cpim-pidf+xml..Contact: <sip:10.15.200.57 :5060>..Subscription-State: active;expires=1236....<?xml version="1.0" encoding="UTF-8"?>..<presence xmlns="urn:ietf:params:xml:ns:cpim-pidf" entity=" pres:PDA_User@fh-joanneum.at">...<tuple id="0x409c45e0x5dd5903fx449711ab">....<status><basic>closed</basic></status>....<contact priority="0.00"></con tact>...</tuple>...<tuple id="0x409c3328x557f3c98x44970bc3">....<status><basic>open</basic></status>....<contact priority="0.00">sip:pda_user@fh-joann eum.at</contact>...</tuple>..</presence>.. # 10.40.1.42:5060 -> 10.15.200.57:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK91d8.2831e0b1.0..To: <sip:esca@fh-joanneum.at>;tag=z9hG4bK19686534..From: <sip:PDA_User@fh -joanneum.at>;tag=589230a54cd08b3768529724501fae28-51f5..Call-ID: 035185567140@10.40.1.42..CSeq: 2 NOTIFY..Server: mjsip stack 1.6..Content-Length: 0. ... # 10.40.1.42:5060 -> 10.15.200.57:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK23e1.c6b3a835.0..To: <sip:esca@fh-joanneum.at>;tag=z9hG4bK13311114..From: <sip:PDA_User@fh -joanneum.at>;tag=589230a54cd08b3768529724501fae28-4a9d..Call-ID: 480873203923@10.40.1.42..CSeq: 6 NOTIFY..Server: mjsip stack 1.6..Content-Length: 0. ... # 10.40.1.42:5060 -> 10.15.200.57:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bKe636.2a57bde3.0..To: <sip:esca@fh-joanneum.at>;tag=z9hG4bK49649677..From: <sip:PDA_User@fh -joanneum.at>;tag=589230a54cd08b3768529724501fae28-9d18..Call-ID: 047322370278@10.40.1.42..CSeq: 12 NOTIFY..Server: mjsip stack 1.6..Content-Length: 0 .... # 10.40.1.42:5060 -> 10.15.200.57:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bKe7b3.ac386935.0..To: <sip:esca@fh-joanneum.at>;tag=z9hG4bK24602857..From: <sip:PDA_User@fh -joanneum.at>;tag=589230a54cd08b3768529724501fae28-52ae..Call-ID: 310599367527@10.40.1.42..CSeq: 19 NOTIFY..Server: mjsip stack 1.6..Content-Length: 0 modparam("msilo","use_contact",0) modparam("msilo","expire_time",7200) # -- usrloc params -- # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("rls", "min_expiration", 200) modparam("rls", "max_expiration", 300) modparam("rls", "default_expiration", 300) modparam("rls", "auth", "none") #"implicit" modparam("rls", "xcap_root", "http://localhost/xcap") modparam("rls", "reduce_xcap_needs", 1) modparam("rls", "db_mode", 1) modparam("rls", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("pa", "default_expires", 3600) modparam("pa", "use_db", 1) #modparam("pa", "pa_domain", "fh-joanneum.at") modparam("pa", "use_offline_winfo", 1) # allow storing authorization requests for offline users into database modparam("pa", "offline_winfo_timer", 600) # how often try to remove old stored authorization requests modparam("pa", "offline_winfo_expiration", 600) # how long stored authorization requests live modparam("pa", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("pa", "auth", "none") # mode of PA authorization: none, implicit or xcap #modparam("pa", "auth_xcap_root", "http://localhost/xcap") modparam("pa", "winfo_auth", "none") # do not authorize watcherinfo subscriptions modparam("pa", "use_callbacks", 1) # use only published information if set to 0 modparam("pa", "accept_internal_subscriptions", 1) # don't accept internal subscriptions from RLS, ... modparam("pa", "watcherinfo_notify", 1) modparam("pa", "max_subscription_expiration", 3600) # maximum value of Expires for subscriptions modparam("pa", "max_publish_expiration", 3600) # maximum value of Expires for publications modparam("pa", "timer_interval", 5) # how often test if something changes and send NOTIFY modparam("presence_b2b", "presence_route", "<sip:127.0.0.1;transport=tcp;lr>") # route for generated SUBSCRIBE requests for presence modparam("presence_b2b", "on_error_retry_time", 60) # waiting time from error to new attepmt about SUBSCRIBE modparam("presence_b2b", "wait_for_term_notify", 33) # how long wait for NOTIFY with Subscription-Status=terminated after unsubscribe modparam("presence_b2b", "resubscribe_delta", 30) # how long before expiration send renewal SUBSCRIBE request modparam("presence_b2b", "min_resubscribe_time", 60) # minimal time to send renewal SUBSCRIBE request from receiving previous response modparam("presence_b2b", "default_expiration", 3600) # default expiration timeout modparam("presence_b2b", "handle_presence_subscriptions", 1) # process internal subscriptions to presence events modparam("usrloc", "db_mode", 1) modparam("domain", "db_mode", 1) modparam("domain|uri_db|acc|auth_db|usrloc|msilo", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("domain", "domain_table", "domain") #modparam("domain", "domain_column", "domain") modparam("fifo", "fifo_file", "/tmp/ser_fifo") if (uri==myself) { log(1, "URI == myself"); if (method=="SUBSCRIBE") { log(1, "SUBSCRIBE received"); if (!t_newtran()) { sl_reply_error(); break; }; # new subscription if (@to.tag=="") { log(1, "new SUBSCRIPTION"); if ((@msg.supported=~"eventlist")) { log(1, "msg.supported=eventlist"); # Supported header field # -> may be RLS subscription if (is_simple_rls_target("$uid-list")) { log(1, "it is simple subscription!\n"); # handle_rls_subscription("1"); # takes From UID and makes XCAP query # for user's list named "default" if (@to.tag=="") { # only for new subscriptions (with empty to tag if (!query_resource_list("default")) { t_reply("500", "XCAP query error"); break; } } } if (!have_flat_list()) { # query_resource_list failed or was not called # do standard RLS query acording to To/AOR query_rls_services(); } if (have_flat_list()) { handle_rls_subscription("1"); break; } } # SUBSCRIBE to existing user # xlog("L_ERR", "PA: handling subscription: %tu from: %fu\n"); log(1, "handle SUBSCRIBE request\n"); handle_subscription("registrar"); log(1, "handled subscription\n"); log(1, "break 1"); break; } else { # renewal subscription log(1, "else - renewal subscription"); if (!handle_rls_subscription("0")) { log(1, "!handle rls subscription"); handle_subscription("registrar"); log(1, "handled subscription 2"); } log(1, "break 2"); break; } } if (method=="REGISTER") { log(1, "REGISTER received"); # if (!www_authorize("fh-joanneum.at", "credentials")) { # www_challenge("fh-joanneum.at", "0"); # break; # }; save("location"); log(1, "saved location"); break; }; if (method=="PUBLISH") { log(1, "PUBLISH received"); if (!t_newtran()) { sl_reply_error(); break; }; log(1, "handle publish request\n"); handle_publish("registrar"); log(1, "handled publish\n"); break; }; if (method=="NOTIFY") { log(1, "NOTIFY received"); if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; if (!handle_notify()) { log(1, "unable to handle notification - lui"); t_reply("481", "Unable to handle notification"); } break; };

2 1

[Serusers] Sending NOTIFY messages from presence module to an SIP Proxy Server
by Anton Schula 21 Jun '06

21 Jun '06

Hello, I have a routing problem with SER and OpenSer regarding presence NOTIFY messages. I want to use OpenSer (OSER in the diagram below) as SIP registrar and for routing SIP messages (which is working perfectly). And I have a separate SER with the PA and RLS modules (called SER-PA_RLS). This one also works perfectly when used as standalone server. Now I have configured the OSER to forward PUBLISH and SUBSCRIBE messages to the SER-PA_RLS. This part is O.K., messages are routed from the client over the OSER to SER-PA_RLS, reply messages travel back from SER-PA_RLS to the OSER and from there to the client. That's O.K., so OSER apears to have correct routing. But NOTIFY messages originating at the SER-PA_RLS are sent directly to the client, using the clients contact address. How can I configure SER-PA_RLS to send the NOTIFY messages via the OSER? In fact, this is all that SER-PA_RLS shall send to the outside world. All routing decissions shall be done in the OSER. The SER-PA_RLS should not be directly accessed (a firewall is in place). Call Flows: =========== Messages from Client to Presence server are O.K. OSER routes them to the SER-PA_RLS. Responses travel back via OSER. ###### ######## ############## # UA # # OSER # # SER-PA_RLS # ###### ######## ############## | | | |----PUBLISH--->| | | |----PUBLISH---->| | |<-----OK------- | |<------OK------| | | | | | | | |---SUBSCRIBE-->| | | |---SUBSCRIBE--->| | |<-----OK------- | |<------OK------| | | | | | | | | | | Intended route of NOTIFY messages: ###### ######## ############## # UA # # OSER # # SER-PA_RLS # ###### ######## ############## | | | | |<----NOTIFY-----| |<----NOTIFY----| | |-------OK----->| | | |-------OK------>| | | | | | | Observed route of NOTIFY messages: ###### ######## ############## # UA # # OSER # # SER-PA_RLS # ###### ######## ############## | | | | | | |<-------------NOTIFY------------| |----------------OK------------->| | | | | | | (In fact I only see the UDP SIP NOTIFY leavin the SER-PA_RLS machine. The firewall blocks this message, so the client nevers receives it.) I tried several configurations in ser.cfg, including ... if (method=="NOTIFY") { log(1,"NOTIFY forwarded to OSER\n"); forward(OSER-hostname,5060); } ... I never saw the log message. I think, that sending out messages from the SER or at least the presence module is independend from routing in the ser.cfg. Can anybody tell me how to forward all outgoing messages to the OSER? Many thank's in advance! best regards A. Schula PS: For your information On OSER, I simply use this if (method=="SUBSCRIBE" and src_ip!=SER-PA_RLS_IP || method=="NOTIFY" and src_ip!=SER-PA_RLS_IP || method=="PUBLISH" and src_ip!=SER-PA_RLS_IP) { forward_udp(SER-PA_RLS_IP,5060); exit; }; -- Echte DSL-Flatrate dauerhaft für 0,- Euro*! "Feel free" mit GMX DSL! http://www.gmx.net/de/go/dsl

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[Serusers] How to start a VOIP business ?
by Concy Pereira 21 Jun '06

21 Jun '06

Sir, I would like to know how can start a successful SIP base VOIP business. I have being going through some documentation of Open server software but could not come to a conclusion. Anybody can suggest me a complete solution for running PC to Phone service Using GPL products, with accounting, authentication and authorization with different rate Plans etc. Here is what I have decided to do. SIP Server (Hardware) 1. Linux RedHat 9.0 2. OpenSer or SIP express router 3. Radius client Accounting /CDR server ( Hardware) 1. Linux Redhat 9.0 2. CDRTool by ag-projects http://www.ag-projects.com/cdrtool.html 3. MySQL 1.Carrirer Subcription to any International whole sale VOIP carrier Please suggest if I have missed anything out. Regards, Concy

4 4

[Users] please help me on Nat and rttpproxy..
by raviprakash sunkara 21 Jun '06

21 Jun '06

Hi Users, Hellow users,, i need help from U. SP1 and SP2 are softphones, which are in Behind the NAT. HP! and HP2 are CISCO ATA phones are not in NAT ( in LAN) SP1 <==> SP2 , the calls are going and speech is clear, this one when LOGin failed , When in login is its says " 408 timeout" SP1 <==> HP1 its says "408 timeout " HP 1 <==> HP2 .... audieo is failed... Please help me -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) ravi.sunkara <at> hyperion-tech.com Hyperion Technology www.hyperion-tech.com

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RE: [Serusers] How to start a VOIP business ?
by Benjamin.George＠t-systems.com 21 Jun '06

21 Jun '06

Hi, I would recommend any of the following for the aforesaid: 1. Cisco media gateway (Router can be converted to a media gateway using voice interface cards). We use Cisco 2600 and 7200 (with MPLS support) series for this functionality. 2. NMS media gateway Some other market leading products are also available for this purpose, for example products from Lucent, Alcatel, etc. Regards, Benjamin. -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Nick Hoffman Sent: Wednesday, June 21, 2006 5:54 AM To: serusers(a)lists.iptel.org Cc: George, Benjamin Subject: Re: [Serusers] How to start a VOIP business ? On Mon June 19 2006 18:48, Benjamin.George(a)t-systems.com wrote: > Hi, > > I too can provide you some inputs on this apart from the things which > are already mentioned: > > 1. You may require a media server for playing prompts, announcements, > etc 2. Voice mail server to store and play back voice mails > 3. Signaling + Media Gateway if you want to connect to PSTN world > 4. Any good performance management/monitoring tool (normally part of > OSS) to measure and manage the performance of the entire infrastructure > to assure zero/minimum downtime > > Regards, > Benjamin. Hi Benjamin. What would you recommend for your 3rd point? Cheers, -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] How to start a VOIP business ?
by Benjamin.George＠t-systems.com 21 Jun '06

21 Jun '06

Hi, I too can provide you some inputs on this apart from the things which are already mentioned: 1. You may require a media server for playing prompts, announcements, etc 2. Voice mail server to store and play back voice mails 3. Signaling + Media Gateway if you want to connect to PSTN world 4. Any good performance management/monitoring tool (normally part of OSS) to measure and manage the performance of the entire infrastructure to assure zero/minimum downtime Regards, Benjamin. _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, June 19, 2006 1:51 PM To: Concy Pereira Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How to start a VOIP business ? It's a fairly big questions you're asking. But basically you're onto it ;-) Ad 1. You should go for something newer than 9.0 Ad 2. You need to decide on your maintenance schedule and policies for upgrades/patching etc. Many larger-scale deployments are based on SER 0.9.x because a) it has proven extremely stable b) it is currently maintained for bug and security fixes and will be for a while. Others choose openser to get more features, but you need to make sure that you can handle testing and deployment of new versions more often. Ad 3. You need a RADIUS server with a user database or you can use mysql (you need mysql regardless of using RADIUS or not) g-) Concy Pereira wrote: Sir, I would like to know how can start a successful SIP base VOIP business. I have being going through some documentation of Open server software but could not come to a conclusion. Anybody can suggest me a complete solution for running PC to Phone service Using GPL products, with accounting, authentication and authorization with different rate Plans etc. Here is what I have decided to do. SIP Server (Hardware) 1. Linux RedHat 9.0 2. OpenSer or SIP express router 3. Radius client Accounting /CDR server ( Hardware) 1. Linux Redhat 9.0 2. CDRTool by ag-projects http://www.ag-projects.co <http://www.ag-projects.co> m/cdrtool.html 3. MySQL 1.Carrirer Subcription to any International whole sale VOIP carrier Please suggest if I have missed anything out. Regards, Concy _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers>

3 2

[Serusers] SER to OpenSER
by Ryan Pagquil 20 Jun '06

20 Jun '06

Hi, What are the things I need to do to make SER config work with OpenSER? I got openser running but I can't register to it no error? Thanks, Ryan

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[Users] Final ACK getting wrong IP address
by Justin Pearce 20 Jun '06

20 Jun '06

Hey guys, I apologize for the length of this message, but I am experiencing a frustrating problem. Let me first explain the situation: We have a custom video conference app installed on two separate computers within our network. We have our OpenSER server setup with a public address at our ISP. There is a NAT between the clients and the OpenSER server with both clients behind the same NAT. Both clients have port ranges mapped through the firewall to allow for audio and video. We were having some issues previously with the clients unable to initiate calls between one another through the server, however I did some work with the config file and managed to get it working. While I was out of town, the problems came back even though the config had not been changed. The issue that now seems to be the problem is that the call process seems to happen properly up to the point at which the client making the call acknowledges the ACK from the client being called. The acknowledgement of the ACK seems to be getting sent to the private IP address as opposed to the public address as all the other communications seem to be. I am not entirely certain where I have the problem, if it is with the server (easier fix) or with the actual clients (difficult fix). I have appended an excerpt of the packet capture (from Ethereal) from the point at which the client programs through the complete call process (without the unregister, since that would seem to be redundant). I have also appended my config file. Would someone be able to look at this and tell me where I screwed up (aside from inside my own head)? Thanks, Justin Pearce Information Technology/Programming Price Video Productions Justin(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com ----------------------------------------------------- # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=no */ #- # check_via # Determines whether or not the server should check for a local address # in the VIA header. Default: no or 0 #- check_via=no #- # dns # Determine if we wish to use DNS lookups on hosts. Very costly. # Default: no or 0 #- dns=no #- # rev_dns # Determine if we wish to do Reverse DNS lookups on hosts. Very costly. # Default: no or 0 #- rev_dns=no #- # syn_branch # This parameter determines whether OpneSER will attempt to # statefully sync ACKs with INVITES in their appropreate transaction. # This is slightly costly, but reliable. Default: 1 #- syn_branch=1 #- # port # This paramter specifies the port which will be listened on. # In this case we are using the standard SIP 5060 #- port=5060 #- # listen # This determines the address and/or interface to be listened on. It # can be specified by the following: # proto:address:port (udp:192.168.16.9:5060) # address:port (192.168.16.9:5060) # address (192.168.16.9) # interface (eth0) #- listen=eth0 children=4 #- # fifo: Determines the location of the FIFO file for the server # (useful for real-time server stats) # fifo_mode: Specifies the access mode of the FIFO (same as *NIX file access) #- fifo="/tmp/openser_fifo" fifo_mode=0666 # # uncomment the following lines for TLS support disable_tls = 1 #listen = tls:eth0:5060 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # MySQL Module loadmodule "/usr/local/lib/openser/modules/mysql.so" # Stateless Replier Module loadmodule "/usr/local/lib/openser/modules/sl.so" # Statefule Transaction Module loadmodule "/usr/local/lib/openser/modules/tm.so" # Record Route Module loadmodule "/usr/local/lib/openser/modules/rr.so" # Max Forwards Module loadmodule "/usr/local/lib/openser/modules/maxfwd.so" # User Location Module loadmodule "/usr/local/lib/openser/modules/usrloc.so" # Registrar Module loadmodule "/usr/local/lib/openser/modules/registrar.so" # Text Operations Module loadmodule "/usr/local/lib/openser/modules/textops.so" # Presence Module loadmodule "/usr/local/lib/openser/modules/pa.so" # Accounting Module loadmodule "/usr/local/lib/openser/modules/acc.so" # Group Module loadmodule "/usr/local/lib/openser/modules/group.so" # URI_DB Module loadmodule "/usr/local/lib/openser/modules/uri_db.so" # NatHelper Module loadmodule "/usr/local/lib/openser/modules/nathelper.so" # Authentication Module loadmodule "/usr/local/lib/openser/modules/auth.so" # Database Authentication Module loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # User Location Module # The following parameters allow for the storage of user location # information in a database for more persistent lookups. # It also allows for easy lookup of data by an outside program. #--------------------------------------- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:<password deleted>@localhost/openser") modparam("usrloc", "received_column", "received") #--------------------------------------- # -- auth params -- # Authentication Module # The following parameters allow for the persistent storage of user # credentials in a database for authentication and authorization # of users at the server level. This allows for more granular control # over user accounts and their access privledges. #--------------------------------------- modparam("auth_db", "db_url", "mysql://openser:<password deleted>@localhost/openser") modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "use_domain", 0) modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "ha1") modparam("auth_db", "password_column_2", "ha1b") #--------------------------------------- # -- rr params -- # Record Route Module # This parameter adds value to ;lr param to make some broken UAs happy #--------------------------------------- modparam("rr", "enable_full_lr", 1) #--------------------------------------- # -- acc params -- # Accounting Module # The following parameters allow for the logging of messages that pass # through the server for the purpose of doing call accounting and tracking. #--------------------------------------- modparam("acc", "log_level", 4) modparam("acc", "log_flag", 3) modparam("acc", "log_missed_flag", 4) modparam("acc", "db_flag", 1 ) modparam("acc", "db_missed_flag", 2) modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://openser:<password deleted>@localhost/openser") #--------------------------------------- # -- pa params -- # Presence Module # These parameters allow for the function of the unstable presence module. #--------------------------------------- modparam("pa", "default_expires", 60) modparam("pa", "db_url", "mysql://openser:<password deleted>@localhost/openser") #--------------------------------------- # -- group params -- # Groups Module # This parameter allows for lookups of user groups from a database. # It is currently used with authentication for user access control. #--------------------------------------- modparam("group", "db_url", "mysql://openser:<password deleted>@localhost/openser") #--------------------------------------- # -- uri_db params -- # URI_DB Module # This allows for the lookup and storage of URIs in a database. #--------------------------------------- modparam("uri_db", "db_url", "mysql://openser:<password deleted>@localhost/openser") #--------------------------------------- # --nathelper params -- # NatHelper Module # The folowing paramters allow for the use of the nathelper module to aid # client programs in the traversal of NATs #--------------------------------------- modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "received_avp", 42) #--------------------------------------- # -- registrar params -- # Registrar Module # The following paramters aid the registrar in interfacing with the nathelper # to aid client programs in the traversal of NATs #--------------------------------------- modparam("registrar", "nat_flag", 6) modparam("registrar", "sip_natping_flag", 7) modparam("registrar", "received_avp", 42) modparam("registrar", "received_param", "received") #--------------------------------------- # ------------------------- request routing logic ------------------- # Main Routing Logic # Past this point is the routing logic used by the server to route all # SIP packets from their source to their intended destination. Changes # to this logic affect the manner in which packets are handled. #--------------------------------------- route{ # Label all transactions for accounting, as per the # parameters for the Accounting Module setflag(1); setflag(2); setflag(3); setflag(4); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; #If the message is too big for SIP if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # We record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER"){ record_route(); }; # Subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if(!is_user_in("From", "active")) { # If the user is not in the "active" group, # reply to the user and exit the call process. force_rport(); sl_send_reply("403", "Forbidden - The user account has been deactivated."); exit; }; if (!www_authorize("tucan", "subscriber")) { force_rport(); www_challenge("tucan", "1"); exit; }; fix_nated_register(); # Fix REGISTERs broken by NAT fix_nated_contact(); force_rport(); # Force the return port in the firewall add_rcv_param(); # Add the "recieved" parameter to the location save("location"); # Save the location in the registrar acc_db_request("200", "acc"); # Log the 200 OK exit; }; lookup("aliases"); # Check the ALIASES table if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # If the user cannot be found, reply to the user # and exit the call process sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }else{ # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (method=="REGISTER") { #Uncomment this if you want to use digest authentication if(!is_user_in("From", "active")) { # If the user is not in the "active" group, # reply to the user and exit call process. force_rport(); sl_send_reply("403", "Forbidden - The user account has been deactivated."); exit; }; if (!www_authorize("tucan", "subscriber")) { force_rport(); www_challenge("tucan", "1"); exit; }; fix_nated_register(); # Fix the REGISTERs broken bt NAT fix_nated_contact(); force_rport(); # Force the return port in the firewall add_rcv_param(); # Add the "recieved' parameter to the location save("location"); # Save the location in the registrar acc_db_request("200", "acc"); # Log the 200 OK exit; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # If the user cannot be found, notify sender and exit. sl_send_reply("404", "Not Found"); exit; }; }; # Continue onto route #1 below... route(1); } route[1] { if(method=="INVITE"){ record_route(); }; # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { # If the stateful relay of a message cannot take place, # reply to the sender statelessly with the error. sl_reply_error(); }; exit; } ---------------------------------------------- No. Time Source Destination Protocol Info 1 0.000000 [public ip here] [server ip here] SIP/SDP Request: INVITE sip:admin@[server ip here], with session description Frame 1 (1136 bytes on wire, 1136 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:admin@[server ip here] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=06c469dd756f4379a66bbc73ed316b82 SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 06c469dd756f4379a66bbc73ed316b82 To: <sip:admin@[server ip here]> SIP to address: sip:admin@[server ip here] Call-ID: 84be41be956d4be09fb1f53921941efa(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 Message body No. Time Source Destination Protocol Info 2 0.000126 [server ip here] [public ip here] SIP Status: 404 Not Found Frame 2 (572 bytes on wire, 572 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 404 Not Found Status-Code: 404 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=06c469dd756f4379a66bbc73ed316b82 SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 06c469dd756f4379a66bbc73ed316b82 To: <sip:admin@[server ip here]>;tag=9a61ee6bd4b4e287efd9537354cc97a2.3058 SIP to address: sip:admin@[server ip here] SIP tag: 9a61ee6bd4b4e287efd9537354cc97a2.3058 Call-ID: 84be41be956d4be09fb1f53921941efa(a)192.168.16.50 CSeq: 1 INVITE Server: OpenSer (1.0.1-tls (i386/linux)) Content-Length: 0 Warning: 392 [server ip here]:5060 "Noisy feedback tells: pid=13061 req_src_ip=[public ip here] req_src_port=62630 in_uri=sip:admin@[server ip here] out_uri=sip:admin@[server ip here] via_cnt==1" No. Time Source Destination Protocol Info 3 0.064898 [public ip here] [server ip here] SIP Request: ACK sip:admin@[server ip here] Frame 3 (396 bytes on wire, 396 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:admin@[server ip here] SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=06c469dd756f4379a66bbc73ed316b82 SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 06c469dd756f4379a66bbc73ed316b82 To: <sip:admin@[server ip here]>;tag=9a61ee6bd4b4e287efd9537354cc97a2.3058 SIP to address: sip:admin@[server ip here] SIP tag: 9a61ee6bd4b4e287efd9537354cc97a2.3058 Call-ID: 84be41be956d4be09fb1f53921941efa(a)192.168.16.50 CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 4 2.579286 [public ip here] [server ip here] SIP Request: REGISTER sip:[server ip here] Frame 4 (534 bytes on wire, 534 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: REGISTER sip:[server ip here] SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.16.50:9436 Max-Forwards: 70 From: <sip:tucan@[server ip here]>;tag=3a73c7106b634aebad60464c52fa6117;epid=2c72c82c29 SIP from address: sip:tucan@[server ip here] SIP tag: 3a73c7106b634aebad60464c52fa6117 To: <sip:tucan@[server ip here]> SIP to address: sip:tucan@[server ip here] Call-ID: 3da6677ead72432aa934d609637ba733(a)192.168.16.50 CSeq: 1 REGISTER Contact: <sip:192.168.16.50:9436>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" User-Agent: RTC/1.2.4949 Event: registration Allow-Events: presence Content-Length: 0 No. Time Source Destination Protocol Info 5 2.579813 [server ip here] [public ip here] SIP Status: 401 Unauthorized (0 bindings) Frame 5 (708 bytes on wire, 708 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.16.50:9436;rport=62630;received=[public ip here] From: <sip:tucan@[server ip here]>;tag=3a73c7106b634aebad60464c52fa6117;epid=2c72c82c29 SIP from address: sip:tucan@[server ip here] SIP tag: 3a73c7106b634aebad60464c52fa6117 To: <sip:tucan@[server ip here]>;tag=9a61ee6bd4b4e287efd9537354cc97a2.a80d SIP to address: sip:tucan@[server ip here] SIP tag: 9a61ee6bd4b4e287efd9537354cc97a2.a80d Call-ID: 3da6677ead72432aa934d609637ba733(a)192.168.16.50 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="tucan", nonce="44987450d34f0c6bd30858bee369ed44197c34f9", qop="auth" Server: OpenSer (1.0.1-tls (i386/linux)) Content-Length: 0 Warning: 392 [server ip here]:5060 "Noisy feedback tells: pid=13057 req_src_ip=[public ip here] req_src_port=62630 in_uri=sip:[server ip here] out_uri=sip:[server ip here] via_cnt==1" No. Time Source Destination Protocol Info 6 2.648131 [public ip here] [server ip here] SIP Request: REGISTER sip:[server ip here] Frame 6 (789 bytes on wire, 789 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: REGISTER sip:[server ip here] SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.16.50:9436 Max-Forwards: 70 From: <sip:tucan@[server ip here]>;tag=3a73c7106b634aebad60464c52fa6117;epid=2c72c82c29 SIP from address: sip:tucan@[server ip here] SIP tag: 3a73c7106b634aebad60464c52fa6117 To: <sip:tucan@[server ip here]> SIP to address: sip:tucan@[server ip here] Call-ID: 3da6677ead72432aa934d609637ba733(a)192.168.16.50 CSeq: 2 REGISTER Contact: <sip:192.168.16.50:9436>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" User-Agent: RTC/1.2.4949 Authorization: Digest username="tucan", realm="tucan", qop=auth, algorithm=md5, uri="sip:[server ip here]", nonce="44987450d34f0c6bd30858bee369ed44197c34f9", nc=00000001, cnonce="72165331663044488051404247101819", response="31eeb043a8aada9191e Event: registration Allow-Events: presence Content-Length: 0 No. Time Source Destination Protocol Info 7 2.649178 [server ip here] [public ip here] SIP Status: 200 OK (1 bindings) Frame 7 (680 bytes on wire, 680 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.16.50:9436;rport=62630;received=[public ip here] From: <sip:tucan@[server ip here]>;tag=3a73c7106b634aebad60464c52fa6117;epid=2c72c82c29 SIP from address: sip:tucan@[server ip here] SIP tag: 3a73c7106b634aebad60464c52fa6117 To: <sip:tucan@[server ip here]>;tag=9a61ee6bd4b4e287efd9537354cc97a2.a80d SIP to address: sip:tucan@[server ip here] SIP tag: 9a61ee6bd4b4e287efd9537354cc97a2.a80d Call-ID: 3da6677ead72432aa934d609637ba733(a)192.168.16.50 CSeq: 2 REGISTER Contact: <sip:[public ip here]:62630>;expires=3600;received="sip:[public ip here]:62630" Server: OpenSer (1.0.1-tls (i386/linux)) Content-Length: 0 Warning: 392 [server ip here]:5060 "Noisy feedback tells: pid=13054 req_src_ip=[public ip here] req_src_port=62630 in_uri=sip:[server ip here] out_uri=sip:[server ip here] via_cnt==1" No. Time Source Destination Protocol Info 8 12.094078 [public ip here] [server ip here] SIP/SDP Request: INVITE sip:admin@[server ip here], with session description Frame 8 (1140 bytes on wire, 1140 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:admin@[server ip here] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62650 Max-Forwards: 70 From: "dheller" <sip:dheller@[public ip here]>;tag=1c031f5aed1941828ce8b0bb85d094ad SIP Display info: "dheller" SIP from address: sip:dheller@[public ip here] SIP tag: 1c031f5aed1941828ce8b0bb85d094ad To: <sip:admin@[server ip here]> SIP to address: sip:admin@[server ip here] Call-ID: cddec4984cee45cf96208ca44e841f3b(a)192.168.16.40 CSeq: 1 INVITE Contact: <sip:[public ip here]:62650> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 Message body No. Time Source Destination Protocol Info 9 12.094290 [server ip here] [public ip here] SIP Status: 404 Not Found Frame 9 (576 bytes on wire, 576 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62650 (62650) Session Initiation Protocol Status-Line: SIP/2.0 404 Not Found Status-Code: 404 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62650 From: "dheller" <sip:dheller@[public ip here]>;tag=1c031f5aed1941828ce8b0bb85d094ad SIP Display info: "dheller" SIP from address: sip:dheller@[public ip here] SIP tag: 1c031f5aed1941828ce8b0bb85d094ad To: <sip:admin@[server ip here]>;tag=9a61ee6bd4b4e287efd9537354cc97a2.3d1d SIP to address: sip:admin@[server ip here] SIP tag: 9a61ee6bd4b4e287efd9537354cc97a2.3d1d Call-ID: cddec4984cee45cf96208ca44e841f3b(a)192.168.16.40 CSeq: 1 INVITE Server: OpenSer (1.0.1-tls (i386/linux)) Content-Length: 0 Warning: 392 [server ip here]:5060 "Noisy feedback tells: pid=13057 req_src_ip=[public ip here] req_src_port=62650 in_uri=sip:admin@[server ip here] out_uri=sip:admin@[server ip here] via_cnt==1" No. Time Source Destination Protocol Info 10 12.108542 [public ip here] [server ip here] SIP Request: ACK sip:admin@[server ip here] Frame 10 (400 bytes on wire, 400 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:admin@[server ip here] SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62650 Max-Forwards: 70 From: "dheller" <sip:dheller@[public ip here]>;tag=1c031f5aed1941828ce8b0bb85d094ad SIP Display info: "dheller" SIP from address: sip:dheller@[public ip here] SIP tag: 1c031f5aed1941828ce8b0bb85d094ad To: <sip:admin@[server ip here]>;tag=9a61ee6bd4b4e287efd9537354cc97a2.3d1d SIP to address: sip:admin@[server ip here] SIP tag: 9a61ee6bd4b4e287efd9537354cc97a2.3d1d Call-ID: cddec4984cee45cf96208ca44e841f3b(a)192.168.16.40 CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 11 12.576634 [public ip here] [server ip here] SIP Request: REGISTER sip:[server ip here] Frame 11 (538 bytes on wire, 538 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: REGISTER sip:[server ip here] SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.16.40:9409 Max-Forwards: 70 From: <sip:dheller@[server ip here]>;tag=4e5b2756f82a46f4b0d79fbe65b4b9b3;epid=9a2a864d1a SIP from address: sip:dheller@[server ip here] SIP tag: 4e5b2756f82a46f4b0d79fbe65b4b9b3 To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: a1541c851e0343c1a02f426b08fea18b(a)192.168.16.40 CSeq: 1 REGISTER Contact: <sip:192.168.16.40:9409>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" User-Agent: RTC/1.2.4949 Event: registration Allow-Events: presence Content-Length: 0 No. Time Source Destination Protocol Info 12 12.577226 [server ip here] [public ip here] SIP Status: 401 Unauthorized (0 bindings) Frame 12 (712 bytes on wire, 712 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62650 (62650) Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.16.40:9409;rport=62650;received=[public ip here] From: <sip:dheller@[server ip here]>;tag=4e5b2756f82a46f4b0d79fbe65b4b9b3;epid=9a2a864d1a SIP from address: sip:dheller@[server ip here] SIP tag: 4e5b2756f82a46f4b0d79fbe65b4b9b3 To: <sip:dheller@[server ip here]>;tag=9a61ee6bd4b4e287efd9537354cc97a2.e360 SIP to address: sip:dheller@[server ip here] SIP tag: 9a61ee6bd4b4e287efd9537354cc97a2.e360 Call-ID: a1541c851e0343c1a02f426b08fea18b(a)192.168.16.40 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="tucan", nonce="4498745aac38af8eb2e26fd9b7002a2e952aa9b6", qop="auth" Server: OpenSer (1.0.1-tls (i386/linux)) Content-Length: 0 Warning: 392 [server ip here]:5060 "Noisy feedback tells: pid=13054 req_src_ip=[public ip here] req_src_port=62650 in_uri=sip:[server ip here] out_uri=sip:[server ip here] via_cnt==1" No. Time Source Destination Protocol Info 13 12.699751 [public ip here] [server ip here] SIP Request: REGISTER sip:[server ip here] Frame 13 (795 bytes on wire, 795 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: REGISTER sip:[server ip here] SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.16.40:9409 Max-Forwards: 70 From: <sip:dheller@[server ip here]>;tag=4e5b2756f82a46f4b0d79fbe65b4b9b3;epid=9a2a864d1a SIP from address: sip:dheller@[server ip here] SIP tag: 4e5b2756f82a46f4b0d79fbe65b4b9b3 To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: a1541c851e0343c1a02f426b08fea18b(a)192.168.16.40 CSeq: 2 REGISTER Contact: <sip:192.168.16.40:9409>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" User-Agent: RTC/1.2.4949 Authorization: Digest username="dheller", realm="tucan", qop=auth, algorithm=md5, uri="sip:[server ip here]", nonce="4498745aac38af8eb2e26fd9b7002a2e952aa9b6", nc=00000001, cnonce="35632750553445629356654004975943", response="852e582b04bd8525b Event: registration Allow-Events: presence Content-Length: 0 No. Time Source Destination Protocol Info 14 12.700823 [server ip here] [public ip here] SIP Status: 200 OK (1 bindings) Frame 14 (833 bytes on wire, 833 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62650 (62650) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.16.40:9409;rport=62650;received=[public ip here] From: <sip:dheller@[server ip here]>;tag=4e5b2756f82a46f4b0d79fbe65b4b9b3;epid=9a2a864d1a SIP from address: sip:dheller@[server ip here] SIP tag: 4e5b2756f82a46f4b0d79fbe65b4b9b3 To: <sip:dheller@[server ip here]>;tag=9a61ee6bd4b4e287efd9537354cc97a2.e360 SIP to address: sip:dheller@[server ip here] SIP tag: 9a61ee6bd4b4e287efd9537354cc97a2.e360 Call-ID: a1541c851e0343c1a02f426b08fea18b(a)192.168.16.40 CSeq: 2 REGISTER Contact: <sip:192.168.16.40:10853>;expires=669;received="sip:[public ip here]:61696", <sip:192.168.16.40:15922>;expires=2101;received="sip:[public ip here]:62650", <sip:[public ip here]:62650>;expires=3600;received="sip:[public ip here]:62650" Server: OpenSer (1.0.1-tls (i386/linux)) Content-Length: 0 Warning: 392 [server ip here]:5060 "Noisy feedback tells: pid=13061 req_src_ip=[public ip here] req_src_port=62650 in_uri=sip:[server ip here] out_uri=sip:[server ip here] via_cnt==1" No. Time Source Destination Protocol Info 15 24.009555 [public ip here] [server ip here] SIP/SDP Request: INVITE sip:dheller@[server ip here], with session description Frame 15 (1140 bytes on wire, 1140 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:dheller@[server ip here] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 Message body No. Time Source Destination Protocol Info 16 24.009983 [server ip here] [public ip here] SIP Status: 100 trying -- your call is important to us Frame 16 (564 bytes on wire, 564 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Server: OpenSer (1.0.1-tls (i386/linux)) Content-Length: 0 Warning: 392 [server ip here]:5060 "Noisy feedback tells: pid=13054 req_src_ip=[public ip here] req_src_port=62630 in_uri=sip:dheller@[server ip here] out_uri=sip:192.168.16.40:10853 via_cnt==1" No. Time Source Destination Protocol Info 17 24.010138 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:192.168.16.40:10853, with session description Frame 17 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 61696 (61696) Session Initiation Protocol Request-Line: INVITE sip:192.168.16.40:10853 SIP/2.0 Method: INVITE Resent Packet: False Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 18 24.010163 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:192.168.16.40:15922, with session description Frame 18 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62650 (62650) Session Initiation Protocol Request-Line: INVITE sip:192.168.16.40:15922 SIP/2.0 Method: INVITE Resent Packet: False Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 19 24.010184 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:[public ip here]:62650, with session description Frame 19 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62650 (62650) Session Initiation Protocol Request-Line: INVITE sip:[public ip here]:62650 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 18 Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.2 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 20 24.150609 [public ip here] [server ip here] SIP Status: 100 Trying Frame 20 (423 bytes on wire, 423 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 Resent Packet: False Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 21 24.164671 [public ip here] [server ip here] SIP Status: 180 Ringing Frame 21 (502 bytes on wire, 502 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 22 24.164812 [server ip here] [public ip here] SIP Status: 180 Ringing Frame 22 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 23 24.166314 [public ip here] [server ip here] SIP Status: 180 Ringing Frame 23 (502 bytes on wire, 502 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 24 24.166448 [server ip here] [public ip here] SIP Status: 180 Ringing Frame 24 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 25 24.229568 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:192.168.16.40:10853, with session description Frame 25 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 61696 (61696) Session Initiation Protocol Request-Line: INVITE sip:192.168.16.40:10853 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 17 Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 26 24.229594 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:[public ip here]:62650, with session description Frame 26 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62650 (62650) Session Initiation Protocol Request-Line: INVITE sip:[public ip here]:62650 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 18 Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.2 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 27 24.326429 [public ip here] [server ip here] SIP Status: 180 Ringing Frame 27 (502 bytes on wire, 502 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 28 24.326584 [server ip here] [public ip here] SIP Status: 180 Ringing Frame 28 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 29 26.233136 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:192.168.16.40:10853, with session description Frame 29 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 61696 (61696) Session Initiation Protocol Request-Line: INVITE sip:192.168.16.40:10853 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 17 Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 30 26.233163 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:[public ip here]:62650, with session description Frame 30 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62650 (62650) Session Initiation Protocol Request-Line: INVITE sip:[public ip here]:62650 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 18 Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.2 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 31 26.622444 [public ip here] [server ip here] SIP Status: 180 Ringing Frame 31 (502 bytes on wire, 502 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 32 26.622630 [server ip here] [public ip here] SIP Status: 180 Ringing Frame 32 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 33 30.240354 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:192.168.16.40:10853, with session description Frame 33 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 61696 (61696) Session Initiation Protocol Request-Line: INVITE sip:192.168.16.40:10853 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 17 Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 34 30.240379 [server ip here] [public ip here] SIP/SDP Request: INVITE sip:[public ip here]:62650, with session description Frame 34 (1302 bytes on wire, 1302 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62650 (62650) Session Initiation Protocol Request-Line: INVITE sip:[public ip here]:62650 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 18 Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.2 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]> SIP to address: sip:dheller@[server ip here] Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Contact: <sip:[public ip here]:62630> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 707 P-hint: usrloc applied Message body No. Time Source Destination Protocol Info 35 30.386377 [public ip here] [server ip here] SIP Status: 180 Ringing Frame 35 (502 bytes on wire, 502 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 36 30.386530 [server ip here] [public ip here] SIP Status: 180 Ringing Frame 36 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Status-Code: 180 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 37 34.093569 [public ip here] [server ip here] SIP/SDP Status: 200 OK, with session description Frame 37 (1244 bytes on wire, 1244 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: False Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 38 34.093812 [server ip here] [public ip here] SIP/SDP Status: 200 OK, with session description Frame 38 (1183 bytes on wire, 1183 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 39 34.118980 [public ip here] [server ip here] SIP Request: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on Frame 39 (464 bytes on wire, 464 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK Route: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 40 34.119656 [server ip here] 192.168.16.40 SIP Request: ACK sip:192.168.16.40:9409 Frame 40 (532 bytes on wire, 532 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: 192.168.16.40 (192.168.16.40) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 9409 (9409) Session Initiation Protocol Request-Line: ACK sip:192.168.16.40:9409 SIP/2.0 Method: ACK Resent Packet: False Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 41 34.586546 [public ip here] [server ip here] SIP/SDP Status: 200 OK, with session description Frame 41 (1244 bytes on wire, 1244 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 37 Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 42 34.586788 [server ip here] [public ip here] SIP/SDP Status: 200 OK, with session description Frame 42 (1183 bytes on wire, 1183 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 38 Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 43 34.670011 [public ip here] [server ip here] SIP Request: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on Frame 43 (464 bytes on wire, 464 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK Route: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 44 34.670212 [server ip here] 192.168.16.40 SIP Request: ACK sip:192.168.16.40:9409 Frame 44 (532 bytes on wire, 532 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: 192.168.16.40 (192.168.16.40) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 9409 (9409) Session Initiation Protocol Request-Line: ACK sip:192.168.16.40:9409 SIP/2.0 Method: ACK Resent Packet: False Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 45 35.590952 [public ip here] [server ip here] SIP/SDP Status: 200 OK, with session description Frame 45 (1244 bytes on wire, 1244 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 37 Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 46 35.591101 [server ip here] [public ip here] SIP/SDP Status: 200 OK, with session description Frame 46 (1183 bytes on wire, 1183 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 38 Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 47 35.612137 [public ip here] [server ip here] SIP Request: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on Frame 47 (464 bytes on wire, 464 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK Route: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 48 35.612323 [server ip here] 192.168.16.40 SIP Request: ACK sip:192.168.16.40:9409 Frame 48 (532 bytes on wire, 532 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: 192.168.16.40 (192.168.16.40) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 9409 (9409) Session Initiation Protocol Request-Line: ACK sip:192.168.16.40:9409 SIP/2.0 Method: ACK Resent Packet: False Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 49 37.609873 [public ip here] [server ip here] SIP/SDP Status: 200 OK, with session description Frame 49 (1244 bytes on wire, 1244 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 37 Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 50 37.610024 [server ip here] [public ip here] SIP/SDP Status: 200 OK, with session description Frame 50 (1183 bytes on wire, 1183 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 38 Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 51 37.805510 [public ip here] [server ip here] SIP Request: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on Frame 51 (464 bytes on wire, 464 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK Route: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 52 37.805903 [server ip here] 192.168.16.40 SIP Request: ACK sip:192.168.16.40:9409 Frame 52 (532 bytes on wire, 532 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: 192.168.16.40 (192.168.16.40) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 9409 (9409) Session Initiation Protocol Request-Line: ACK sip:192.168.16.40:9409 SIP/2.0 Method: ACK Resent Packet: False Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 53 41.584640 [public ip here] [server ip here] SIP/SDP Status: 200 OK, with session description Frame 53 (1244 bytes on wire, 1244 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 37 Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 54 41.584780 [server ip here] [public ip here] SIP/SDP Status: 200 OK, with session description Frame 54 (1183 bytes on wire, 1183 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 38 Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 55 41.630352 [public ip here] [server ip here] SIP Request: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on Frame 55 (464 bytes on wire, 464 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK Route: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 56 41.630524 [server ip here] 192.168.16.40 SIP Request: ACK sip:192.168.16.40:9409 Frame 56 (532 bytes on wire, 532 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: 192.168.16.40 (192.168.16.40) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 9409 (9409) Session Initiation Protocol Request-Line: ACK sip:192.168.16.40:9409 SIP/2.0 Method: ACK Resent Packet: False Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 57 49.593330 [public ip here] [server ip here] SIP/SDP Status: 200 OK, with session description Frame 57 (1244 bytes on wire, 1244 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62650 (62650), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 37 Message Header Via: SIP/2.0/UDP [server ip here];branch=z9hG4bK4bbe.c709a097.1 Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 58 49.593472 [server ip here] [public ip here] SIP/SDP Status: 200 OK, with session description Frame 58 (1183 bytes on wire, 1183 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: [public ip here] ([public ip here]) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62630 (62630) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: True Suspected resend of frame: 38 Message Header Via: SIP/2.0/UDP [public ip here]:62630 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 INVITE Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Contact: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 679 Message body No. Time Source Destination Protocol Info 59 49.969732 [public ip here] [server ip here] SIP Request: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on Frame 59 (464 bytes on wire, 464 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK Route: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Length: 0 No. Time Source Destination Protocol Info 60 49.970031 [server ip here] 192.168.16.40 SIP Request: ACK sip:192.168.16.40:9409 Frame 60 (532 bytes on wire, 532 bytes captured) Ethernet II, Src: 00:30:48:28:66:ad, Dst: 00:b0:8e:e6:a0:02 Internet Protocol, Src Addr: [server ip here] ([server ip here]), Dst Addr: 192.168.16.40 (192.168.16.40) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 9409 (9409) Session Initiation Protocol Request-Line: ACK sip:192.168.16.40:9409 SIP/2.0 Method: ACK Resent Packet: False Message Header Record-Route: <sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on> Via: SIP/2.0/UDP [server ip here];branch=0 Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 69 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 61 58.227421 [public ip here] [server ip here] SIP Request: BYE sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on Frame 61 (464 bytes on wire, 464 bytes captured) Ethernet II, Src: 00:b0:8e:e6:a0:02, Dst: 00:30:48:28:66:ad Internet Protocol, Src Addr: [public ip here] ([public ip here]), Dst Addr: [server ip here] ([server ip here]) User Datagram Protocol, Src Port: 62630 (62630), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:[server ip here];ftag=528def284594483d94678a66ecf035ca;lr=on SIP/2.0 Method: BYE Resent Packet: False Message Header Via: SIP/2.0/UDP [public ip here]:62630 Max-Forwards: 70 From: "tucan" <sip:tucan@[public ip here]>;tag=528def284594483d94678a66ecf035ca SIP Display info: "tucan" SIP from address: sip:tucan@[public ip here] SIP tag: 528def284594483d94678a66ecf035ca To: <sip:dheller@[server ip here]>;tag=7830b3dafe9140a6aae6b74e591eaca1 SIP to address: sip:dheller@[server ip here] SIP tag: 7830b3dafe9140a6aae6b74e591eaca1 Call-ID: d104a459394b4a8a93164cb5a6248755(a)192.168.16.50 CSeq: 2 BYE Route: <sip:192.168.16.40:9409> User-Agent: RTC/1.2 Content-Length: 0 ---------------------------------------------------

1 0

[Serusers] I must be stupid... but
by olivier.taylor 20 Jun '06

20 Jun '06

hi all, I have modparam("tm", "fr_inv_timer", 60) when a callee doesn't anwer, he often send a 487 (cancelled) before 60 seconds... It means timeout in fact or no answer... If the caller stop the call, I also get a 487... How to make the difference on a 487 generated by the caller and by the callee? It makes difference, if it's generated by the callee, I will forward the call to voicemail or another number, if it's generated by the caller, I will do nothing. When I set fr_inv_timer to 15, I always get a timeout and it's ok, but my customers dislikes that as well as the callees who needs to be better than Ben Jonhson for the 100meters :( Does exists a parameter to be send to the callee to tell him to wait 60 seconds + 1 before sending me a Cancel? Regards, Olivier

7 13

RE: [Users] Privacy (*67)
by Pablo Delgado 20 Jun '06

20 Jun '06

To other sip phones. Pablo Delgado Digiport Miami LLC Phone: (305) 424-0016 Web: www.digiportmiami.com Email: pablo(a)digiportmiami.com -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Tuesday, June 20, 2006 3:56 PM To: Pablo Delgado Cc: users(a)openser.org Subject: [Users] Privacy (*67) Pablo Delgado writes: > I am using openser to handle all calls within my organization and does a > great job. The only thing that users recently pointed out to me is that > *67 callerid block does not work. caller id block to where? pstn or another sip phone? -- juha

2 1

[Users] Privacy (*67)
by Pablo Delgado 20 Jun '06

20 Jun '06

This has probably been discussed here so if it has, point me in the right direction if not, then I guess this will benefits other. I am using openser to handle all calls within my organization and does a great job. The only thing that users recently pointed out to me is that *67 callerid block does not work. My question is how can I set a block the caller id. What I have done so far is, I know how to filter out the calls that begin with *67, but after hours of digging though docks I have not figured out a way to set privacy on. Thanks in advance for any help. Regards, Pablo Delgado

2 1

RE: [Serusers] checking source ip and dest ip
by İlker Aktuna (Koç.net) 20 Jun '06

20 Jun '06

In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/02090600/07064D00/01080049/001…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

3 2

[Users] Updating usrloc database
by Britton Payne 20 Jun '06

20 Jun '06

Hello, all. In our current configuration OpenSER is using a MySQL db for location information. Registrations made to openser get pushed into the db ok, but changes to the database directly (like additions) do not appear to make it to openser. Is there a way to make changes to the database become visible to openser like a fifo command or should I be using db_mode=1? We would like to be able to do this without restarting openser and dropping calls in progress. Thank you, Britton

5 4

[Users] RE: acc.so problem
by Edson 20 Jun '06

20 Jun '06

>From the first line message, it seems to me that You have some concern with Your MySQL server.... it seems that it run out of free socked... or that the user is not available... please, try to take a look on this... Edson. P.S: post to the list as well... the chances to a good help increases... ;) > -----Original Message----- > From: zhan li [mailto:zhan_li_2000@hotmail.com] > Sent: sexta-feira, 16 de junho de 2006 01:52 > To: bogdan(a)voice-system.ro > Cc: 4lists(a)gmail.com > Subject: acc.so problem > > Hello > > When I try to start the openser with acc modules, I got the following > error, > the v136545352 is very odd, I checked the version table of Location which > is 1001, I wonder what v136545352 means in error message. > > I search online, and found another people get the exactly same problem > with > mine. > > Thanks > > > > 0(0) db_init: Connection 'mysql://openser:openserrw@localhost/openser' not > found in pool > 0(0) new_connection: Opening MySQL connection: > mysql://openser:openserrw@localhost/openser > 0(0) new_connection: Connection type is Localhost via UNIX socket > 0(0) new_connection: Protocol version is 10 > 0(0) new_connection: Server version is 4.0.18-standard > 0(0) usrloc:preload_udomain: Wrong version v136545352 for table > <location>, > expected v1001 > 0(0) register_udomain(): Error while preloading domain 'location' > 0(0) pool_remove: Removing connection from the pool > 0(0) domain_fixup(): Error while registering domain > ERROR: error -1 while trying to fix configuration

2 1

Re: [Serusers] Re: mediaproxy problem
by Andrey Kouprianov 20 Jun '06

20 Jun '06

Thank you Hakan!! That's great news! I still cant get my hands free to deal with this issue. However, I looked up my FreeBSD ports and python mysqldb support lib is there. :) I'll try to run mediaproxy asap and let the list know about the outcome. Regards, Andrey. On 6/20/06, Hakan YASTI <hakanyasti(a)gmail.com> wrote: > Hi Andrey, > > It has worked. After I installed > py24-MySQLdb-1.2.1.p2 Access a MySQL database through Python > it is working now. > Thanks for your kind interest. > Best Regards, > Hakan. > > > ----- Original Message ----- > From: "Andrey Kouprianov" <andrey.kouprianov(a)gmail.com> > To: <serusers(a)iptel.org> > Sent: Sunday, June 18, 2006 2:58 PM > Subject: Re: [Serusers] Re: mediaproxy problem > > > > Will try!! :) > > > > Thanx a lot! > > > > On 6/18/06, rusl <rusl(a)jpfortune.org> wrote: > >> > >> You need to have the python " MySQLdb" module installed. > >> > >> On Sunday 18. June 2006 11:45, Andrey Kouprianov wrote: > >> > Yup. Me too. I hope to get some comments on this... :) > >> > > >> > Anyone? > >> > > >> > On 6/17/06, Hakan YASTI <hakanyasti(a)gmail.com> wrote: > >> > > Hi Andrey, > >> > > > >> > > I have the same problem with proxy dispatcher. Have got a solution > >> > > for > >> > > your problem. I couldn't see any solution for your problem. > >> > > > >> > > Hope to hera soon. > >> > > > >> > > Best Regards, > >> > > Hakan. > >> > > > >> > > > >> > > > >> > > > >> > > Hi, > >> > > >> > I have tried to install mediaproxy 1.7.2. So, i did. Configuration > >> > > >> > > was > >> > > >> > about the same as 1.4.2, however, "mediaproxy start", just > >> > > >> > > started > >> > > >> > mediaproxy.py, but not proxydispatcher.py. Version 1.4.2 started > >> > > >> > > them > >> > > >> > both w/o any problem. Trying to start proxydispatcher.py > >> > > >> > > independently > >> > > >> > did not give any results. /var/run/proxydispatcher.sock is > >> > > >> > > simply not > >> > > >> > there anymore. There are no logs or whatsoever, that I could show > >> > > >> > > you, > >> > > >> > because I did not find any. Anyway, here's the config > >> > > >> > > (NOTE: > >> > > >> > mediaproxy's running on the same machine as SER): > >> > > >> > [Dispatcher] > >> > start > >> > > >> > > = yes > >> > > >> > socket = /var/run/proxydispatcher.sock > >> > listen = <my IP address> > >> > group > >> > > >> > > = wheel > >> > > >> > defaultProxy = /var/run/mediaproxy.sock > >> > > >> > [MediaProxy] > >> > start = > >> > > >> > > yes > >> > > >> > socket = /var/run/mediaproxy.sock > >> > group = wheel > >> > listen = <my IP > >> > > >> > > address> > >> > > >> > allow = <my IP address>, 127.0.0.1 > >> > proxyIP = <my IP > >> > > >> > > address> > >> > > >> > portRange = 60000:65000 > >> > TOS = 0xb8 > >> > idleTimeout = 60 > >> > holdTimeout = > >> > > >> > > 3600 > >> > > >> > forceClose = 10800 > >> > > >> > [Accounting] > >> > dbaccounting = off > >> > > >> > //<my IP address> > >> > > >> > > is IP of SER/mediaproxy machine > >> > > >> > // the rest is irrelevant > >> > > >> > Please, also note > >> > > >> > > that I dont have a DNS SRV entry for media proxy. > >> > > >> > Could that cause the > >> > > >> > > problem? > >> > > >> > One way or another, I had to downgrade back to 1.4.2. Any > >> > > >> > > suggestions? > >> > > >> > Thanx in advance. > >> > > >> > Andrey. > >> > > >> > _______________________________________________ > >> > Serusers mailing list > >> > Serusers(a)lists.iptel.org > >> > http://lists.iptel.org/mailman/listinfo/serusers > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > >

1 1

[Serusers] sipsak says: error: cannot handle this redirect
by jp＠speakplease.com 20 Jun '06

20 Jun '06

I'm quite new with this whole SIP thing (surprise surprise) I'm trying to learn and build a system but i can't seem to get even the most simple redirects to work. I did get it to work earlier, but then it stopped working and now i can't get it to work again. any help please? since i'm not that hack with the linux either i reinstalled it in case i had made stupid mistakes (couldn't get phpmyadmin to work at that point but that's another story...) when i do this: sipsak -vv -s sip:7770088@192.168.1.102 sipsak says: message received: ** received redirect error: cannot handle this redirect: SIP/2.0 300 Redirect Via: SIP/2.0/UDP 192.168.1.111:1235;branch=z9hG4bK.457b3839;rport=1236;alias From: sip:sipsak@192.168.1.111:1235;tag=4b93b8d3 To: sip:7770088@192.168.1.102;tag=b27e1a1d33761e85846fc98f5f3a7e58.dd84 Call-ID: 1267972307(a)192.168.1.111 CSeq: 1 OPTIONS Contact: sip:7770088@sip.varphonex.com:5060 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.102:5060 "Noisy feedback tells: pid=30758 req_src_ip=192.168.1.111 req_src_port=1236 in_uri=sip:7770088@192.168.1.102 out_uri=sip:7770088@sip.varphonex.com:5060 via_cnt==1" the redirect i'm using is the redirect example from ser documentation (changed the rewrite uri to 7770088@sip.varphonex.com:5060) I'm trying to get my voip phone to connect to varphonex through ser. thanks in advance! -JP

2 1

[Serusers] Radius acc ERROR
by Kostas Marneris 20 Jun '06

20 Jun '06

Hi all, I've already successfully configured SER for syslog accounting, but now I'm trying to turn it to RADIUS accounting. - I changed the modules/acc/Makefile file (uncomment the following lines) : DEFS+=-DRAD_ACC include ../../Makefile.radius - re-compile & install (make & copy the new acc.so to the lib path). - ser.cfg : loadmodule "/opt/ser-0.9.6/lib/ser/modules/acc.so" modparam("acc", "radius_config", "/opt/radiusclient-ng/etc/radiusclient-ng/radiusclient.conf") When I try to restart SER (serctl restart) I get the following error mesg : Jun 20 14:45:12 xxx ser[19502]: ERROR: acc: can't get code for the Failed attribute value Jun 20 14:45:12 xxx ser[19502]: init_mod(): Error while initializing module acc any hint ? error on radiusclient conf / SER modparam conf ? thank you

1 0

[Serusers] checking source ip and dest ip
by İlker Aktuna (Koç.net) 20 Jun '06

20 Jun '06

Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/0B080506/07014D00/08080B4F/001…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

[Serusers] PSTN connectivity
by ravi reddy 20 Jun '06

20 Jun '06

Hi SER users , At present iam using SER-0.9.6 as proxy,registrar ,and the sip to sip calls i tested are o.k and is perfectly going and now when i tried to connect to PSTN it showing 500 internal server error i dont understand whether it is the mistake of cisco sip gateway or SER 's here is the sip messages which i get when i tried to call SIP to PSTN . I HAD GIVEN SIP TEST CLIENTS USER ID AS 32331000 AND THE LOCAL PSTN NO. ARE LIKE THIS :eg: 22458792 My Domain/Realm=81.21.33.35 and iam using onsip ser.cfg as the ser config file if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { # Domestic PSTN route(4); route(5); break; }; and the pstn connectivity lines are like this: and here are the sip messages when i tried to call pstn number my pstn gateway which iam testing now is 81.21.38.21 # U 81.21.34.34:5068 -> 81.21.33.35:5060 INVITE sip:22485400@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bKccadffff3fd60000. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>. Contact: <sip:32331001@192.168.0.74:5068>. Supported: replaces. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46532 INVITE. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 305. . v=0. o=32331001 8000 8000 IN IP4 192.168.0.74. s=SIP Call. c=IN IP4 192.168.0.74. t=0 0. m=audio 8000 RTP/AVP 8 0 18 4 2 99. a=sendrecv. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:99 iLBC/8000. a=fmtp:99 mode=20. a=ptime:20. # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bKccadffff3fd60000;received= 81.21.34.34. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=74961b5b71b6ddce908b9155b956083f.c903. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46532 INVITE. Proxy-Authenticate: Digest realm="81.21.33.35", nonce="4497c732a5a6f0ff3fb62bd9ac4561a02e4975c9". Content-Length: 0. Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27570 req_src_ip= 81.21.34.34 req_src_port=5068 in_uri=sip:22485400@81.21.33.35 out_uri= sip:22485400@81.21.33.35 via_cnt==1". . # U 81.21.34.34:5068 -> 81.21.33.35:5060 ACK sip:22485400@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bKccadffff3fd60000. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=74961b5b71b6ddce908b9155b956083f.c903. Contact: <sip:32331001@192.168.0.74:5068>. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46532 ACK. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 81.21.34.34:5068 -> 81.21.33.35:5060 INVITE sip:22485400@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>. Contact: <sip:32331001@192.168.0.74:5068>. Supported: replaces. Proxy-Authorization: Digest username="32331001", realm="81.21.33.35", algorithm=MD5, uri="sip:22485400@81.21.33.35", nonce="4497c732a5a6f0ff3fb62bd9ac4561a02e4975c9", response="af0c7b754d60bf28e493637995ac5675". Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46533 INVITE. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 305. . v=0. o=32331001 8000 8001 IN IP4 192.168.0.74. s=SIP Call. c=IN IP4 192.168.0.74. t=0 0. m=audio 8000 RTP/AVP 8 0 18 4 2 99. a=sendrecv. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:99 iLBC/8000. a=fmtp:99 mode=20. a=ptime:20. # U 81.21.34.6:5062 -> 81.21.33.35:5060 .... # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.74:5068 ;branch=z9hG4bK1be8ffffc8ecffff;rport=5068;received=81.21.34.34. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46533 INVITE. Content-Length: 0. Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27568 req_src_ip= 81.21.34.34 req_src_port=5068 in_uri=sip:22485400@81.21.33.35 out_uri= sip:22485400@81.21.38.21 via_cnt==1". . # U 81.21.33.35:5060 -> 81.21.38.21:5060 INVITE sip:22485400@81.21.38.21 SIP/2.0. Record-Route: <sip:22485400@81.21.33.35 :5060;nat=yes;ftag=26b000000d350000;lr=on>. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK7de4.ed7d9ff3.0. Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>. Contact: <sip:32331001@81.21.34.34:5068>. Supported: replaces. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46533 INVITE. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 16. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 321. . v=0. o=32331001 8000 8001 IN IP4 192.168.0.74. s=SIP Call. c=IN IP4 81.21.33.3581.21.33.35. t=0 0. m=audio 3505035050 RTP/AVP 8 0 18 4 2 99. a=sendrecv. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:99 iLBC/8000. a=fmtp:99 mode=20. a=ptime:20. # U 81.21.38.21:5060 -> 81.21.33.35:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK7de4.ed7d9ff3.0,SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 81.21.38.21:5060 -> 81.21.33.35:5060 SIP/2.0 500 Internal Server Error. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK7de4.ed7d9ff3.0,SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 81.21.33.35:5060 -> 81.21.38.21:5060 ACK sip:22485400@81.21.38.21 SIP/2.0. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK7de4.ed7d9ff3.0. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. Call-ID: 9ad9ffff49050000(a)192.168.0.74. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. CSeq: 46533 ACK. Content-Length: 0. . # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 500 Internal Server Error. Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 81.21.34.34:5068 -> 81.21.33.35:5060 ACK sip:22485400@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Contact: <sip:32331001@192.168.0.74:5068>. Proxy-Authorization: Digest username="32331001", realm="81.21.33.35", algorithm=MD5, uri="sip:22485400@81.21.33.35", nonce="4497c732a5a6f0ff3fb62bd9ac4561a02e4975c9", response="e09e9b23688cc3847829da66fe1fc145". Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46533 ACK. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 500 Internal Server Error. Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 500 Internal Server Error. Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. I hope you guys will help in this issue Thanks in Advance. Regards' Ravi

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[Serusers] Compiling problem
by Alois Schafferhofer 20 Jun '06

20 Jun '06

Hi. I want to compile SER 0.9.5. Therefore, I need the packages -- libmysqlclient-dev and -- libxml2-dev The problem is, that I can't find these packages for Suse 9.3. Can someone give me an address where I can download these packages? Regards, Alois

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[Users] Response to Nat problem
by Cseke Tamas 20 Jun '06

20 Jun '06

Hello! We've a problem with a provisional response. The invite is sent through NAT , and the response will be sent to incorrect port. The invite is sent from 193.77.152.140:53559 to ser, (it is a natted firewall. It not insert Via header, don't understand sip) The topmost via is: Via: SIP/2.0/UDP 10.10.1.2:5080;branch=z9hG4bK-a47f2525b20ab1a953a7c74174551819;received=193.77.152.140 The server recognize, that it is received from 193.77.152.140, but the reply will be sent to 193.77.152.140:5080 according to the RFC 3261 the port used in the response is , what present in the via header. How can i force that the response will be sent to the source port (52559)? Thanks, Tamas

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RE: [Serusers] Mediaproxy problem - how to disable P2P rtp connection
by İlker Aktuna (Koç.net) 20 Jun '06

20 Jun '06

Hi, I have found the problem. (in fact it was not a problem) The client was using the alternate media connection information which is peer-2-peer. How can I prevent P2P connection ? I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 6:13 PM To: serusers(a)iptel.org Subject: RE: [Serusers] Mediaproxy problem Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, I have found the problem. (in fact it was not a problem) The client was using the alternate media connection information which is peer-2-peer. How can I prevent P2P connection ? I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 6:13 PM To: serusers(a)iptel.org Subject: RE: [Serusers] Mediaproxy problem Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker <http://387555.sigclick.mailinfo.com/sigclick/03090E00/07074E03/00090C49/002…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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RE: [Serusers] How to start a VOIP business ?
by Concy Pereira 20 Jun '06

20 Jun '06

Thanks Prashant, Can you give me more details about the CDR tool that comes with Asterisk. My final aim is to provide PC-to-Phone service on a small scale. Will your solution help me. Note that I am replying from a new email address Regards Concy -----Original Message----- From: Prashant Bhalesain [mailto:prashb2000@yahoo.com] Sent: Monday, June 19, 2006 11:18 PM To: Benjamin.George(a)t-systems.com; greger(a)teigre.com; CPEREIRA(a)Human-soft.com Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] How to start a VOIP business ? Boils down to Asterisk+SER I guess. You can use asterisk as a voicemail server+media server+CDR and SER as a SIP router. (connection to PSTN/termination gateway?) Asteriskathome provides most software bundled. worth giving a try. Prashant --- Benjamin.George(a)t-systems.com wrote: > Hi, > > > > I too can provide you some inputs on this apart from > the things which are already mentioned: > > > > 1. You may require a media server for playing > prompts, announcements, etc > 2. Voice mail server to store and play back voice > mails > 3. Signaling + Media Gateway if you want to connect > to PSTN world > 4. Any good performance management/monitoring tool > (normally part of OSS) to measure and manage the > performance of the entire infrastructure to assure > zero/minimum downtime > > > > Regards, > > Benjamin. > > > > > > _____ > > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Monday, June 19, 2006 1:51 PM > To: Concy Pereira > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] How to start a VOIP business > ? > > > > It's a fairly big questions you're asking. But > basically you're onto it ;-) > Ad 1. You should go for something newer than 9.0 > Ad 2. You need to decide on your maintenance > schedule and policies for upgrades/patching etc. > Many larger-scale deployments are based on SER 0.9.x > because a) it has proven extremely stable b) it is > currently maintained for bug and security fixes and > will be for a while. Others choose openser to get > more features, but you need to make sure that you > can handle testing and deployment of new versions > more often. > Ad 3. You need a RADIUS server with a user database > or you can use mysql (you need mysql regardless of > using RADIUS or not) > > g-) > > > Concy Pereira wrote: > > Sir, > > > > I would like to know how can start a successful SIP > base VOIP business. I have being going through some > documentation of Open server software but could not > come to a conclusion. > > > > Anybody can suggest me a complete solution for > running PC to Phone service Using GPL products, with > accounting, authentication and authorization with > different rate Plans etc. > > > > Here is what I have decided to do. > > > > SIP Server (Hardware) > > > > 1. Linux RedHat 9.0 > > 2. OpenSer or SIP express router > > 3. Radius client > > > > Accounting /CDR server ( Hardware) > > > > 1. Linux Redhat 9.0 > > 2. CDRTool by ag-projects > http://www.ag-projects.co > <http://www.ag-projects.co> m/cdrtool.html > > 3. MySQL > > > > 1.Carrirer > > > > Subcription to any International whole sale VOIP > carrier > > > > Please suggest if I have missed anything out. > > > > Regards, > > > > Concy > > > > > > > > _____ > > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > <http://lists.iptel.org/mailman/listinfo/serusers> > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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Re: [Users] Dumb Newbie Question
by Daniel-Constantin Mierla 19 Jun '06

19 Jun '06

On 06/19/06 20:42, Kelvin Williams wrote: > > My OpenSER box receives an Invite to > sip:+11234567890@ip.add.re.ss:5060 – my XPro is registered as > Kelvin(a)my.domain.com <mailto:Kelvin@my.domain.com>. > > When I receive this call, to my number 11234567890 – how do I make my > XPro ring? > If you have only one account, then you can do it from the script: rewriteuser("Kelvin"); lookup("location"); t_relay(); If you have many, you better use aliases. So just point 11234567890 to kelvin in aliases table (see alias_db module http://openser.org/docs/modules/1.0.x/alias_db.html) and then perform user location lookup (see registrar/usrloc modules). It will translate in: alias_db_lookup("dbaliases"); lookup("location"); t_relay(); Cheers, Daniel > kw > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] Dumb Newbie Question
by Kelvin Williams 19 Jun '06

19 Jun '06

My OpenSER box receives an Invite to sip:+11234567890@ip.add.re.ss:5060 - my XPro is registered as Kelvin(a)my.domain.com. When I receive this call, to my number 11234567890 - how do I make my XPro ring? kw

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[Serusers] publish presentity: No tuple for ....
by Alois Schafferhofer 19 Jun '06

19 Jun '06

Hi, I want my SER(0.9.3) to store status and x, y coordinates into database and to generate a NOTIFY whenever the position changes. In the log-file, I can see this information, but SER doesn't store it in the database and it doesn't generate a NOTIFY. In the log file, I get the message: publish presentity: "No tuple for ....". I hope, someone can help me. Regards, Alois /usr/sbin/ser[25182]: handle publish request /usr/sbin/ser[25182]: get_pres_uri: _puri=sip:PDA_User@fhjoanneum.at /usr/sbin/ser[25182]: handle_publish -4- p_uri= p_uri.len=23 /usr/sbin/ser[25182]: handle_publish -5- presentity=0x4069c0f8 /usr/sbin/ser[25182]: parse_pidf: sipuri=0x821b230:sip:PDA_User@fh-joanneum.at contact=(nil):(null) basic=0x821b2c8:open location=(nil):(null) /usr/sbin/ser[25182]: parse_pidf: site=(nil):(null) floor=(nil):(null) room=(nil):(null) /usr/sbin/ser[25182]: parse_pidf: x=0x821be28:01524.3739 y=0x821be38:04704.0970 radius=(nil):(null) /usr/sbin/ser[25182]: publish_presentity: no tuple for PDA_User(a)fh-joanneum.at /usr/sbin/ser[25182]: publish_presentity: event_package=1 -1- /usr/sbin/ser[25182]: handle_publish -8- paerrno=0 /usr/sbin/ser[25182]: handled publish PUBLISH sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1443248938..Max-Forwards: 70..From: <sip:PDA_User@fh-joanneum.at>;tag=271646917..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: 1040576725-1127390674-656438565..CSeq: 5 PUBLISH..Contact: <sip:PDA_User@fh-joanneum. at>..Content-Type: application/cpim-pidf+xml..Content-Length: 308..Expires:7200..Event: presence..User-Agent: SIP .NET 1.0 evaluation version, www.independentsoft.com....<?xml version="1.0" encoding="utf-8"?> <presencexmlns="urn:ietf:params:xml:ns:pidf"entity="sip:PDA_User@fh-joanneum.at"> <tupleid="9r28r59"> <status> <basic>open</basic> <geopriv><locationinfo><civilAddress><x>01524.3595</x><y>04704.0990</y></civilAddress></locationinfo></geopriv> </status> </tuple> </presence>.. SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1443248938..From: <sip:PDA_User@fh-joanneum.at>;tag=271646917..To: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-4cd6..Call-ID: 1040576725-1127390674-656438565..CSeq: 5 PUBLISH..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells:pid=25182 req_src_ip=10.0.0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1"

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[Users] nat traversal
by Padmaja RV 19 Jun '06

19 Jun '06

Hi all, i need to implement openser for nat traversal. can anyone tell me if using nat helper module would be better or using media proxy. from so far i read, i understood that mediaproxy needs separate installation but does not need to make much change in the existing openser.cfg. where as if i use the nathelper, there is no need for separate media proxy installation but the openser cfg needs to be changed. Please suggest the best way to do it....? also please correct me if what i understood is not right...... Thanks so much, Padmaja

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RE: [Users] SIP Clients Can't REGISTER
by Hamid Ali Asgari 19 Jun '06

19 Jun '06

The client can't register, still it doesn't allow registration. Attached is the ngrep output you mentioned. Thanks, Hamid Ngrep Output: # U 85.15.7.17:3166 -> 85.15.7.18:5060 REGISTER sip:shatel.ir SIP/2.0..From: sip:101@shatel.ir;tag=5fa04040..To: sip:101@shatel.ir..Call-Id: 9401fb256987b29da88b308b1 1ce64ed..Cseq: 1 REGISTER..Contact: <sip:101@85.15.7.17;LINEID=6f5beb97d08670a1c8328ce1526ae928>..Expires: 300..Date: Mon, 19 J un 2006 11:02:02 GMT..Max-Forwards: 20..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Via: SIP/2.0/UDP 85.15.7.17;branch =z9hG4bK-a4ba80ab4920a96042a782f74c40944d..Content-Length: 0.... # U 85.15.7.17:3166 -> 85.15.7.18:5060 REGISTER sip:shatel.ir SIP/2.0..From: sip:101@shatel.ir;tag=5fa04040..To: sip:101@shatel.ir..Call-Id: 9401fb256987b29da88b308b1 1ce64ed..Cseq: 1 REGISTER..Contact: <sip:101@85.15.7.17;LINEID=6f5beb97d08670a1c8328ce1526ae928>..Expires: 300..Date: Mon, 19 J un 2006 11:02:02 GMT..Max-Forwards: 20..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Via: SIP/2.0/UDP 85.15.7.17;branch =z9hG4bK-a4ba80ab4920a96042a782f74c40944d..Content-Length: 0.... # U 85.15.7.17:3166 -> 85.15.7.18:5060 REGISTER sip:shatel.ir SIP/2.0..From: sip:101@shatel.ir;tag=5fa04040..To: sip:101@shatel.ir..Call-Id: 9401fb256987b29da88b308b1 1ce64ed..Cseq: 1 REGISTER..Contact: <sip:101@85.15.7.17;LINEID=6f5beb97d08670a1c8328ce1526ae928>..Expires: 300..Date: Mon, 19 J un 2006 11:02:02 GMT..Max-Forwards: 20..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Via: SIP/2.0/UDP 85.15.7.17;branch =z9hG4bK-a4ba80ab4920a96042a782f74c40944d..Content-Length: 0.... # U 85.15.7.18:5060 -> 85.15.7.17:5060 SIP/2.0 401 Unauthorized..From: sip:101@shatel.ir;tag=5fa04040..To: sip:101@shatel.ir;tag=a2c1e7c3808498f2cb2d193cd520e0fc.60d6 ..Call-Id: 9401fb256987b29da88b308b11ce64ed..Cseq: 1 REGISTER..Via: SIP/2.0/UDP 85.15.7.17;branch=z9hG4bK-a4ba80ab4920a96042a78 2f74c40944d..WWW-Authenticate: Digest realm="shatel.ir", nonce="449677461ef22770698d74f5dde4b0c2f00cf44f"..Server: OpenSer (1.0 .1 (i386/linux))..Content-Length: 0..Warning: 392 85.15.7.18:5060 "Noisy feedback tells: pid=13352 req_src_ip=85.15.7.17 req_s rc_port=3166 in_uri=sip:shatel.ir out_uri=sip:shatel.ir via_cnt==1".... # U 85.15.7.18:5060 -> 85.15.7.17:5060 SIP/2.0 401 Unauthorized..From: sip:101@shatel.ir;tag=5fa04040..To: sip:101@shatel.ir;tag=a2c1e7c3808498f2cb2d193cd520e0fc.60d6 ..Call-Id: 9401fb256987b29da88b308b11ce64ed..Cseq: 1 REGISTER..Via: SIP/2.0/UDP 85.15.7.17;branch=z9hG4bK-a4ba80ab4920a96042a78 2f74c40944d..WWW-Authenticate: Digest realm="shatel.ir", nonce="4496774acfd59b6e20c1968fc0bff804d526ce03"..Server: OpenSer (1.0 .1 (i386/linux))..Content-Length: 0..Warning: 392 85.15.7.18:5060 "Noisy feedback tells: pid=13352 req_src_ip=85.15.7.17 req_s rc_port=3166 in_uri=sip:shatel.ir out_uri=sip:shatel.ir via_cnt==1".... -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Monday, June 19, 2006 4:32 PM To: Hamid Ali Asgari Cc: users(a)openser.org Subject: Re: [Users] SIP Clients Can't REGISTER Hello, On 06/19/06 14:38, Hamid Ali Asgari wrote: > Hi, > Thanks for the reply. > I commented out the entry you mentioned: > > #101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended", > #Service-Type == "Group-Check" > > What happens now is that the radius will check for a group named SUSPENDED > and will reject because it won't match the 101 user: > > >From RADIUS debug: > users: Matched entry DEFAULT at line 152 > users: Matched entry DEFAULT at line 255 > ----------------------------------------------------------- > Users:152: DEFAULT Auth-Type = System > Fall-Through = 1 > Users:255: DEFAULT Auth-Type := Reject, Service-Type == "Group-Check" > > Any thoughts? > this is ok, it is the response that the user is not in the suspended group. The processing should go further in openser configuration script. Do you encounter other issues with calls/registrations? Cheers, Daniel > Hamid > > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Monday, June 19, 2006 12:54 PM > To: Hamid Ali Asgari > Cc: 'Edson'; users(a)openser.org > Subject: Re: [Users] SIP Clients Can't REGISTER > > If look carefully in the configuration file of the tutorial, it checks > for users belonging to "suspended" group. As you have in 'users' file: > > 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended", > Service-Type == "Group-Check" > > Reply-Message = "Authorized" > > means that user 101 is in suspended group and it is not allowed to > register or initiate calls. So remove this line for your 'users' file. > > Also, when you test, just watch the network traffic on you sip server, > it will reveal the reply code so you can spot easily which statement was > hit in the config file: ngrep -d any port 5060 > > Cheers, > Daniel > >

2 1

RE: [Serusers] Mediaproxy problem
by İlker Aktuna (Koç.net) 19 Jun '06

19 Jun '06

Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker <http://387555.sigclick.mailinfo.com/sigclick/010C0605/07024D06/0203064C/007…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

[Serusers] Mediaproxy problem
by İlker Aktuna (Koç.net) 19 Jun '06

19 Jun '06

Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker <http://387555.sigclick.mailinfo.com/sigclick/0A090604/07074E00/0906054D/001…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

Re: [Serusers] SER doesn't generate NOTIFY
by Vaclav Kubart 19 Jun '06

19 Jun '06

Look into "presence handbook" (on ftp://ftp.iptel.org/pub/ser/presence). If you want to throw authorization out, you can set PA module parameter "auth" to "none". Vaclav On Mon, Jun 19, 2006 at 03:21:39PM +0200, Alois Schafferhofer wrote: > And how can the subscription get authorized? > > Thank you! > > Alois > > > >From: Vaclav Kubart <vaclav.kubart(a)iptel.org> > >To: Alois Schafferhofer <schaffi_lui(a)hotmail.com> > >CC: serusers(a)lists.iptel.org > >Subject: Re: [Serusers] SER doesn't generate NOTIFY > >Date: Mon, 19 Jun 2006 15:13:43 +0200 > > > >Hi, > >it seems that the subscription is not authorized (response is "202 > >Accepted") and thus the subscriber is not notified on changes. > > Vaclav > > > >On Mon, Jun 19, 2006 at 02:45:41PM +0200, Alois Schafferhofer wrote: > >> Hi, > >> > >> I use Presence Snapshot 4.1 on Linux 9.3. > >> SER doesn't generate a NOTIFY when status changes. In MySQL database, I > >can > >> see the change: > >> table: Presentity_Contact > >> basic: online/offline > >> status:"" > >> > >> I wonder why basic is online/offline and status is "". Is this correct? > >In > >> my messages, i send basic=open/closed. > >> > >> In my config-file, I set the timer_interval to 5 seconds. > >> > >> What am I doing wrong? > >> > >> Regards, > >> Alois > >> > >> SUBSCRIBE sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP > >> 10.15.200.22:5060;rport;branch=z9hG4bK67417..Max-Forwards: 70..To: > >> <sip:PDA_User@fh-joanneum.at>..From: > >> <sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..Call-ID: > >> 720365631452@10.15.200.22..CSeq: 1 SUBSCRIBE..Contact: > >> <sip:esca@10.15.200.22>..Expires: 3600..User-Agent: mjsip stack > >1.6..Event: > >> presence..Accept: application/cpimpidf+xml..Content-Length: 0.... > >> > >> SIP/2.0 202 Accepted..Via: SIP/2.0/UDP > >> > >10.15.200.22:5060;rport=5060;branch=z9hG4bK67417..To:<sip:PDA_User@fhjoanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..From: > >> > ><sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..Call-ID:720365631452@10.15.200.22..CSeq: > >> 1 SUBSCRIBE..Expires: 3600..Contact: <sip:10.15.200.57:5060>..Server: > >Sip > >> EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: > >> 0..Warning: 392 10.15.200.57:5060 "Noisy feedba > >> ck tells: pid=17547 req_src_ip=10.15.200.22 req_src_port=5060 > >> in_uri=sip:PDA_User@fhjoanneum. > >> at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1".... > >> > >> NOTIFY sip:esca@10.15.200.22 SIP/2.0..Via: SIP/2.0/UDP > >> > >10.15.200.57;branch=z9hG4bK720f.274f52a6.0..To:<sip:esca@fhjoanneum.at>;tag=z9hG4bK64795000..From: > >> > ><sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..CSeq:1 > >> NOTIFY..Call-ID: 720365631452@10.15.200.22..Content-Length: 0.. > >> User-Agent: Sip EXpress router(0.10.99-dev35-pa-4.1 > >(i386/linux))..Event: > >> presence..Contact: > >> <sip:10.15.200.57:5060>..Subscription-State: pending;expires=3600.... > >> > >> U 10.15.200.22:5060 -> 10.15.200.57:5060 > >> SIP/2.0 200 OK..Via: SIP/2.0/UDP > >> 10.15.200.57;branch=z9hG4bK720f.274f52a6.0..To: > >> <sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..From: <sip:PDA_User@fh > >> -joanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..Call-ID: > >> 720365631452@10.15.200.22..CSeq: 1 NOTIFY..Server: mjsip stack > >> 1.6..Content-Length:0. > >> > >> > >> > >> PUBLISH sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP > >> 10.0.0.3:5060;branch=z9hG4bK843798674..Max-Forwards: 70..From: > >> <sip:PDA_User@fhjoanneum > >> .at>;tag=850624414..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: > >> 1836804380-820133259- > >> 1969969487..CSeq: 6 PUBLISH..Contact: <sip:PDA_User@fh-joanneum.a > >> t>..Content-Type: application/pidf+xml..Content-Length: 226..Expires: > >> 7200..Event: > >> presence..User-Agent: SIP .NET 1.0 evaluation version, www.independ > >> entsoft.com..SIP-If-Match: 0x409c3088x10cc41b0x4496b143....<?xml > >> version="1.0" > >> encoding="utf-8"?> > >> <presence xmlns="urn:ietf:params:xml:ns:pidf" > >> entity="pres:PDA_user@fh-joanneum.at"> > >> <tuple id="4e9a7335-5f7f-4a3c-91a7-fd761e9138b0"> > >> <status><basic>open</basic> > >> </status> > >> </tuple> > >> </presence>.. > >> > >> SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK843798674.. > >> From:<sip:PDA_User@fh-joanneum.at>;tag=850624414..To: > >> > ><sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-ed6b..Call-ID: > >> 1836804380-820133259- > >> 1969969487..CSeq: 6 PUBLISH..Expires: > >> 120..SIP-Tag:0x409c3088x10cc41b0x4496b143..Contact: > >> <sip:10.0.0.11:5060>..Server: Sip EXpress router > >> (0.10.99-dev35-pa-4.1(i386/linux))..Content-Length: 0..Warning: 392 > >> 10.0.0.11: > >> 5060 "Noisy feedback tells: pid=17551 req_src_ip=10.0.0.3 > >req_src_port=5060 > >> in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at > >> via_cnt==1" > >> > >> > >> > >> > >> # modparam("msilo","registrar","sip:registrar@test-domain.com") > >> modparam("msilo","use_contact",0) > >> modparam("msilo","expire_time",7200) > >> # -- usrloc params -- > >> # -- auth params -- > >> # Uncomment if you are using auth module > >> # > >> modparam("auth_db", "calculate_ha1", yes) > >> # > >> # If you set "calculate_ha1" parameter to yes (which true in this > >config), > >> # uncomment also the following parameter) > >> # > >> modparam("auth_db", "password_column", "password") > >> # -- rr params -- > >> # add value to ;lr param to make some broken UAs happy > >> modparam("rr", "enable_full_lr", 1) > >> modparam("rls", "min_expiration", 200) > >> modparam("rls", "max_expiration", 300) > >> modparam("rls", "default_expiration", 300) > >> modparam("rls", "auth", "none") #"implicit" > >> modparam("rls", "xcap_root", "http://localhost/xcap") > >> modparam("rls", "reduce_xcap_needs", 1) > >> modparam("rls", "db_mode", 1) > >> modparam("rls", "db_url", "mysql://ser:heslo@localhost:3306/ser") > >> modparam("pa", "default_expires", 3600) > >> modparam("pa", "use_db", 1) > >> #modparam("pa", "pa_domain", "fh-joanneum.at") > >> modparam("pa", "use_offline_winfo", 1) # allow storing authorization > >> requests for offline users into database > >> modparam("pa", "offline_winfo_timer", 600) # how often try to remove old > >> stored authorization requests > >> modparam("pa", "offline_winfo_expiration", 600) # how long stored > >> authorization requests live > >> modparam("pa", "db_url", "mysql://ser:heslo@localhost:3306/ser") > >> modparam("pa", "auth", "xcap") # mode of PA authorization: none, > >> implicit or xcap > >> modparam("pa", "auth_xcap_root", "http://localhost/xcap") > >> modparam("pa", "winfo_auth", "none") # do not authorize watcherinfo > >> subscriptions > >> modparam("pa", "use_callbacks", 1) # use only published information if > >set > >> to 0 > >> modparam("pa", "accept_internal_subscriptions", 1) # don't accept > >internal > >> subscriptions from RLS, ... > >> modparam("pa", "watcherinfo_notify", 1) > >> modparam("pa", "max_subscription_expiration", 3600) # maximum value of > >> Expires for subscriptions > >> modparam("pa", "max_publish_expiration", 120) # maximum value of Expires > >for > >> publications > >> modparam("pa", "timer_interval", 5) > >> > >> modparam("presence_b2b", "presence_route", > >> "<sip:127.0.0.1;transport=tcp;lr>") > >> # route for generated SUBSCRIBE requests for presence > >> modparam("presence_b2b", "on_error_retry_time", 60) # waiting time from > >> error to new attepmt about SUBSCRIBE > >> modparam("presence_b2b", "wait_for_term_notify", 33) # how long wait for > >> NOTIFY > >> with Subscription-Status=terminated after unsubscribe > >> modparam("presence_b2b", "resubscribe_delta", 30) # how long before > >> expiration send renewal SUBSCRIBE request > >> modparam("presence_b2b", "min_resubscribe_time", 60) # minimal time to > >send > >> renewal SUBSCRIBE request from receiving previous response > >> modparam("presence_b2b", "default_expiration", 3600) # default > >expiration > >> timeout > >> modparam("presence_b2b", "handle_presence_subscriptions", 1) # process > >> internal subscriptions to presence events > >> modparam("usrloc", "db_mode", 1) > >> modparam("domain", "db_mode", 1) > >> modparam("domain|uri_db|acc|auth_db|usrloc|msilo", "db_url", > >> "mysql://ser:heslo@localhost:3306/ser") > >> modparam("domain", "domain_table", "domain") > >> #modparam("domain", "domain_column", "domain") > >> modparam("fifo", "fifo_file", "/tmp/ser_fifo") > >> > >> > >> route{ > >> # XML RPC > >> if (method == "POST" || method == "GET") { > >> create_via(); > >> log(1, "Created via()"); > >> dispatch_rpc(); > >> log(1, "Dispatched rps()"); > >> break; > >> log(1, "break XML RPC"); > >> } > >> # initial sanity checks -- messages with > >> # max_forwards==0, or excessively long requests > >> if (!mf_process_maxfwd_header("10")) { > >> sl_send_reply("483","Too Many Hops"); > >> break; > >> }; > >> if (msg:len >= max_len ) { > >> sl_send_reply("513", "Message too big"); > >> break; > >> }; > >> # we record-route all messages -- to make sure that > >> # subsequent messages will go through our proxy; that's > >> # particularly good if upstream and downstream entities > >> # use different transport protocol > >> if (!method=="REGISTER") record_route(); > >> # subsequent messages withing a dialog should take the > >> # path determined by record-routing > >> if (loose_route()) { > >> # mark routing logic in request > >> append_hf("P-hint: rr-enforced\r\n"); > >> route(1); > >> break; > >> }; > >> if (uri==myself) { > >> log(1, "URI == myself"); > >> if (method=="SUBSCRIBE") { > >> log(1, "SUBSCRIBE received"); > >> if (!t_newtran()) { > >> sl_reply_error(); > >> break; > >> }; > >> # new subscription > >> if (@to.tag=="") { > >> log(1, "new SUBSCRIPTION"); > >> if ((@msg.supported=~"eventlist")) { > >> log(1, "msg.supported=eventlist"); > >> # Supported header field > >> # -> may be RLS subscription > >> if (is_simple_rls_target("$uid-list")) { > >> log(1, "it is simple > >> subscription!\n"); > >> # handle_rls_subscription("1"); > >> # takes From UID and makes XCAP > >> query > >> # for user's list named > >> "default" > >> if (@to.tag=="") { > >> # only for new > >> subscriptions (with empty to tag > >> if > >> (!query_resource_list("default")) { > >> t_reply("500", > >> "XCAP query error"); > >> break; > >> } > >> } > >> } > >> if (!have_flat_list()) { > >> # query_resource_list failed or > >> was not called > >> # do standard RLS query acording > >> to To/AOR > >> query_rls_services(); > >> } > >> if (have_flat_list()) { > >> handle_rls_subscription("1"); > >> break; > >> } > >> } > >> # SUBSCRIBE to existing user > >> # xlog("L_ERR", "PA: handling subscription: %tu > >> from: %fu\n"); > >> log(1, "handle SUBSCRIBE request\n"); > >> handle_subscription("registrar"); > >> log(1, "handled subscription\n"); > >> log(1, "break 1"); > >> break; > >> } > >> else { # renewal subscription > >> log(1, "else - renewal subscription"); > >> if (!handle_rls_subscription("0")) { > >> log(1, "!handle rls subscription"); > >> handle_subscription("registrar"); > >> log(1, "handled subscription 2"); > >> } > >> log(1, "break 2"); > >> break; > >> } > >> } > >> if (method=="REGISTER") { > >> log(1, "REGISTER received"); > >> # if (!www_authorize("fh-joanneum.at", "credentials")) { > >> # www_challenge("fh-joanneum.at", "0"); > >> # break; > >> # }; > >> save("location"); > >> log(1, "saved location"); > >> break; > >> }; > >> > >> > >> if (method=="PUBLISH") { > >> log(1, "PUBLISH received"); > >> if (!t_newtran()) { > >> sl_reply_error(); > >> break; > >> }; > >> log(1, "handle publish request\n"); > >> handle_publish("registrar"); > >> log(1, "handled publish\n"); > >> break; > >> }; > >> > >> if (method=="NOTIFY") { > >> log(1, "NOTIFY received"); > >> if (!t_newtran()) { > >> log(1, "newtran error\n"); > >> sl_reply_error(); > >> break; > >> }; > >> if (!handle_notify()) { > >> log(1, "unable to handle notification - lui"); > >> t_reply("481", "Unable to handle notification"); > >> } > >> break; > >> }; > >> > >> > >> # message authorization > >> if (method=="MESSAGE") { > >> log(1, "MESSAGE authorization\n"); > >> if (!authorize_message("http://localhost/xcap")) { > >> sl_reply("403", "Forbidden"); > >> break; > >> } > >> } > >> # native SIP destinations are handled using our USRLOC DB > >> if (!lookup("location")) { > >> log(1, "!lookup location gö"); > >> sl_send_reply("404", "Not Found"); > >> break; > >> }; > >> }; > >> # append_hf("P-hint: usrloc applied\r\n"); > >> log(1, "route (1)"); > >> route(1); > >> } > >> route[1] > >> { > >> # send it out now; use stateful forwarding as it works reliably > >> # even for UDP2TCP > >> if (!t_relay()) { > >> log(1, "!t_relay - sl reply error"); > >> sl_reply_error(); > >> }; > >> } > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] pa_domain
by Alois Schafferhofer 19 Jun '06

19 Jun '06

Hi. I installed the compiled version of SER 0.9.5 on Linux 9.3. I just copied the files into the right directories. Now I have a problem to connect to MySQL database. In the Log-message, I can see the line: pa_domain=sip.handhelds.org But I need another pa_domain and I think that this is the problem. How can I change pa_domain? ser[17044]: Maxfwd module- initializing ser[17044]: db_url=0x82077c0 ser[17044]: db_url=mysql://ser:heslo@127.0.0.1:3306/ser ser[17044]: db_url.len=36 ser[17044]: pa_domain=sip.handhelds.org ser[17044]: pa_mod: use_db=1 db_url.s=mysql://ser:heslo@127.0.0.1:3306/ser pa_domain=sip.handhelds.org ser[17044]: pa_mod_init done ser[17044]: subscribe_fixup: pdomain name is registrar ser[17044]: new_connection(): Client does not support authentication protocol requested by server; consider upgrading MySQL client ser[17044]: db_init(): Could not create a connection ser[17044]: ERROR: auth_db_ver: unable to open database connection ser[17044]: auth_db:str_fixup(): Error while querying table version ser[17044]: ERROR: fix_expr : fix_actions error ser[17044]: db_close(): Invalid parameter value Best regards, Alois

2 1

[Serusers] SER doesn't generate NOTIFY
by Alois Schafferhofer 19 Jun '06

19 Jun '06

Hi, I use Presence Snapshot 4.1 on Linux 9.3. SER doesn't generate a NOTIFY when status changes. In MySQL database, I can see the change: table: Presentity_Contact basic: online/offline status:"" I wonder why basic is online/offline and status is "". Is this correct? In my messages, i send basic=open/closed. In my config-file, I set the timer_interval to 5 seconds. What am I doing wrong? Regards, Alois SUBSCRIBE sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.15.200.22:5060;rport;branch=z9hG4bK67417..Max-Forwards: 70..To: <sip:PDA_User@fh-joanneum.at>..From: <sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..Call-ID: 720365631452@10.15.200.22..CSeq: 1 SUBSCRIBE..Contact: <sip:esca@10.15.200.22>..Expires: 3600..User-Agent: mjsip stack 1.6..Event: presence..Accept: application/cpimpidf+xml..Content-Length: 0.... SIP/2.0 202 Accepted..Via: SIP/2.0/UDP 10.15.200.22:5060;rport=5060;branch=z9hG4bK67417..To:<sip:PDA_User@fhjoanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..From: <sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..Call-ID:720365631452@10.15.200.22..CSeq: 1 SUBSCRIBE..Expires: 3600..Contact: <sip:10.15.200.57:5060>..Server: Sip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: 0..Warning: 392 10.15.200.57:5060 "Noisy feedba ck tells: pid=17547 req_src_ip=10.15.200.22 req_src_port=5060 in_uri=sip:PDA_User@fhjoanneum. at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1".... NOTIFY sip:esca@10.15.200.22 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK720f.274f52a6.0..To:<sip:esca@fhjoanneum.at>;tag=z9hG4bK64795000..From: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..CSeq:1 NOTIFY..Call-ID: 720365631452@10.15.200.22..Content-Length: 0.. User-Agent: Sip EXpress router(0.10.99-dev35-pa-4.1 (i386/linux))..Event: presence..Contact: <sip:10.15.200.57:5060>..Subscription-State: pending;expires=3600.... U 10.15.200.22:5060 -> 10.15.200.57:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK720f.274f52a6.0..To: <sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..From: <sip:PDA_User@fh -joanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..Call-ID: 720365631452@10.15.200.22..CSeq: 1 NOTIFY..Server: mjsip stack 1.6..Content-Length:0. PUBLISH sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK843798674..Max-Forwards: 70..From: <sip:PDA_User@fhjoanneum .at>;tag=850624414..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: 1836804380-820133259- 1969969487..CSeq: 6 PUBLISH..Contact: <sip:PDA_User@fh-joanneum.a t>..Content-Type: application/pidf+xml..Content-Length: 226..Expires: 7200..Event: presence..User-Agent: SIP .NET 1.0 evaluation version, www.independ entsoft.com..SIP-If-Match: 0x409c3088x10cc41b0x4496b143....<?xml version="1.0" encoding="utf-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:PDA_user@fh-joanneum.at"> <tuple id="4e9a7335-5f7f-4a3c-91a7-fd761e9138b0"> <status><basic>open</basic> </status> </tuple> </presence>.. SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK843798674.. From:<sip:PDA_User@fh-joanneum.at>;tag=850624414..To: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-ed6b..Call-ID: 1836804380-820133259- 1969969487..CSeq: 6 PUBLISH..Expires: 120..SIP-Tag:0x409c3088x10cc41b0x4496b143..Contact: <sip:10.0.0.11:5060>..Server: Sip EXpress router (0.10.99-dev35-pa-4.1(i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11: 5060 "Noisy feedback tells: pid=17551 req_src_ip=10.0.0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1" # modparam("msilo","registrar","sip:registrar@test-domain.com") modparam("msilo","use_contact",0) modparam("msilo","expire_time",7200) # -- usrloc params -- # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("rls", "min_expiration", 200) modparam("rls", "max_expiration", 300) modparam("rls", "default_expiration", 300) modparam("rls", "auth", "none") #"implicit" modparam("rls", "xcap_root", "http://localhost/xcap") modparam("rls", "reduce_xcap_needs", 1) modparam("rls", "db_mode", 1) modparam("rls", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("pa", "default_expires", 3600) modparam("pa", "use_db", 1) #modparam("pa", "pa_domain", "fh-joanneum.at") modparam("pa", "use_offline_winfo", 1) # allow storing authorization requests for offline users into database modparam("pa", "offline_winfo_timer", 600) # how often try to remove old stored authorization requests modparam("pa", "offline_winfo_expiration", 600) # how long stored authorization requests live modparam("pa", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("pa", "auth", "xcap") # mode of PA authorization: none, implicit or xcap modparam("pa", "auth_xcap_root", "http://localhost/xcap") modparam("pa", "winfo_auth", "none") # do not authorize watcherinfo subscriptions modparam("pa", "use_callbacks", 1) # use only published information if set to 0 modparam("pa", "accept_internal_subscriptions", 1) # don't accept internal subscriptions from RLS, ... modparam("pa", "watcherinfo_notify", 1) modparam("pa", "max_subscription_expiration", 3600) # maximum value of Expires for subscriptions modparam("pa", "max_publish_expiration", 120) # maximum value of Expires for publications modparam("pa", "timer_interval", 5) modparam("presence_b2b", "presence_route", "<sip:127.0.0.1;transport=tcp;lr>") # route for generated SUBSCRIBE requests for presence modparam("presence_b2b", "on_error_retry_time", 60) # waiting time from error to new attepmt about SUBSCRIBE modparam("presence_b2b", "wait_for_term_notify", 33) # how long wait for NOTIFY with Subscription-Status=terminated after unsubscribe modparam("presence_b2b", "resubscribe_delta", 30) # how long before expiration send renewal SUBSCRIBE request modparam("presence_b2b", "min_resubscribe_time", 60) # minimal time to send renewal SUBSCRIBE request from receiving previous response modparam("presence_b2b", "default_expiration", 3600) # default expiration timeout modparam("presence_b2b", "handle_presence_subscriptions", 1) # process internal subscriptions to presence events modparam("usrloc", "db_mode", 1) modparam("domain", "db_mode", 1) modparam("domain|uri_db|acc|auth_db|usrloc|msilo", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("domain", "domain_table", "domain") #modparam("domain", "domain_column", "domain") modparam("fifo", "fifo_file", "/tmp/ser_fifo") route{ # XML RPC if (method == "POST" || method == "GET") { create_via(); log(1, "Created via()"); dispatch_rpc(); log(1, "Dispatched rps()"); break; log(1, "break XML RPC"); } # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (uri==myself) { log(1, "URI == myself"); if (method=="SUBSCRIBE") { log(1, "SUBSCRIBE received"); if (!t_newtran()) { sl_reply_error(); break; }; # new subscription if (@to.tag=="") { log(1, "new SUBSCRIPTION"); if ((@msg.supported=~"eventlist")) { log(1, "msg.supported=eventlist"); # Supported header field # -> may be RLS subscription if (is_simple_rls_target("$uid-list")) { log(1, "it is simple subscription!\n"); # handle_rls_subscription("1"); # takes From UID and makes XCAP query # for user's list named "default" if (@to.tag=="") { # only for new subscriptions (with empty to tag if (!query_resource_list("default")) { t_reply("500", "XCAP query error"); break; } } } if (!have_flat_list()) { # query_resource_list failed or was not called # do standard RLS query acording to To/AOR query_rls_services(); } if (have_flat_list()) { handle_rls_subscription("1"); break; } } # SUBSCRIBE to existing user # xlog("L_ERR", "PA: handling subscription: %tu from: %fu\n"); log(1, "handle SUBSCRIBE request\n"); handle_subscription("registrar"); log(1, "handled subscription\n"); log(1, "break 1"); break; } else { # renewal subscription log(1, "else - renewal subscription"); if (!handle_rls_subscription("0")) { log(1, "!handle rls subscription"); handle_subscription("registrar"); log(1, "handled subscription 2"); } log(1, "break 2"); break; } } if (method=="REGISTER") { log(1, "REGISTER received"); # if (!www_authorize("fh-joanneum.at", "credentials")) { # www_challenge("fh-joanneum.at", "0"); # break; # }; save("location"); log(1, "saved location"); break; }; if (method=="PUBLISH") { log(1, "PUBLISH received"); if (!t_newtran()) { sl_reply_error(); break; }; log(1, "handle publish request\n"); handle_publish("registrar"); log(1, "handled publish\n"); break; }; if (method=="NOTIFY") { log(1, "NOTIFY received"); if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; if (!handle_notify()) { log(1, "unable to handle notification - lui"); t_reply("481", "Unable to handle notification"); } break; }; # message authorization if (method=="MESSAGE") { log(1, "MESSAGE authorization\n"); if (!authorize_message("http://localhost/xcap")) { sl_reply("403", "Forbidden"); break; } } # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log(1, "!lookup location gö"); sl_send_reply("404", "Not Found"); break; }; }; # append_hf("P-hint: usrloc applied\r\n"); log(1, "route (1)"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { log(1, "!t_relay - sl reply error"); sl_reply_error(); }; }

2 1

RE: [Serusers] How to send instant message programmatically
by srinivas.sriram＠wipro.com 19 Jun '06

19 Jun '06

Hi Greger I have checked with sipp , with that we can send messages. But What is my requirement is i have write the code for constructing SIP "MESSAGE" message for a given sip uri. This I have to call form one of the ser module. Total what I am trying is to send a IM message when there is a missed call for a particular SIP uri. In SER there is a code for writing into the data base of the sip uri , when there is a missed call for that Uri. At this point we have to send IM message. Thank you, Regards, Sriram Srinivas. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, June 19, 2006 5:35 PM To: Srinivas Sriram (WT01 - IP-Multimedia Carrier & Ent Networks) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to send instant message programmatically You are better off generating the message outside SER. SER is not a UAS. Have a look at sipsak or sipp (both found at sourceforge, I think). g-) srinivas.sriram(a)wipro.com wrote: Hello, I am using SER-0.9.6. How to send instant message programmatically, how to build sip message and which function can by used(like t_uac, t_request). Can you help where and which to be changed in ser code. Thank you, Regards, Sriram Srinivas. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

Re: [Users] SIP Clients Can't REGISTER
by Daniel-Constantin Mierla 19 Jun '06

19 Jun '06

Hello, On 06/19/06 14:38, Hamid Ali Asgari wrote: > Hi, > Thanks for the reply. > I commented out the entry you mentioned: > > #101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended", > #Service-Type == "Group-Check" > > What happens now is that the radius will check for a group named SUSPENDED > and will reject because it won't match the 101 user: > > >From RADIUS debug: > users: Matched entry DEFAULT at line 152 > users: Matched entry DEFAULT at line 255 > ----------------------------------------------------------- > Users:152: DEFAULT Auth-Type = System > Fall-Through = 1 > Users:255: DEFAULT Auth-Type := Reject, Service-Type == "Group-Check" > > Any thoughts? > this is ok, it is the response that the user is not in the suspended group. The processing should go further in openser configuration script. Do you encounter other issues with calls/registrations? Cheers, Daniel > Hamid > > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Monday, June 19, 2006 12:54 PM > To: Hamid Ali Asgari > Cc: 'Edson'; users(a)openser.org > Subject: Re: [Users] SIP Clients Can't REGISTER > > If look carefully in the configuration file of the tutorial, it checks > for users belonging to "suspended" group. As you have in 'users' file: > > 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended", > Service-Type == "Group-Check" > > Reply-Message = "Authorized" > > means that user 101 is in suspended group and it is not allowed to > register or initiate calls. So remove this line for your 'users' file. > > Also, when you test, just watch the network traffic on you sip server, > it will reveal the reply code so you can spot easily which statement was > hit in the config file: ngrep -d any port 5060 > > Cheers, > Daniel > > > > > On 06/18/06 12:23, Hamid Ali Asgari wrote: > >> Hi, >> >> I get the following: >> >> [root@localhost radiusclient-ng]# radiusclient -f >> /usr/local/etc/radiusclient-ng/radiusclient.conf >> 'User-Name=101(a)mydomain.com' 'Password=101';echo $? >> >> Reply-Message = 'Authenticated' >> >> 1 >> >> I guess the radiusclient is working fine since I am receiving the >> RADIUS packets on the freeradius debugs. >> >> I suspect the following entries in the debug: >> >> rad_recv: Access-Request packet from host 127.0.0.1:32901, id=197, >> length=64 >> >> User-Name = "101@mydomain" >> >> Sip-Group = "*suspended*" >> >> Service-Type = Group-Check >> >> NAS-Port = 0 >> >> NAS-IP-Address = 127.0.0.1 >> >> Why is openser setting the group to SUSPENDED???? Since there is no >> group named _suspended_ on the radius that may be the reason the >> radius is rejecting: >> >> RADIUS Users file: >> >> ### --- avps --- >> >> 101(a)mydomain.com Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" >> >> Sip-Avp += "#3#1", >> >> Sip-Avp += "#4:08:00", >> >> Sip-Avp += "#5:16:00", >> >> Sip-Avp += "#6:Mon,Wed,Thu,Fri" >> >> 102(a)mydomain.com Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" >> >> Sip-Avp += "#3#1", >> >> Sip-Avp += "#4:08:00", >> >> Sip-Avp += "#5:16:00", >> >> Sip-Avp += "#6:Mon,Wed,Thu,Free" >> >> DEFAULT Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" >> >> ### --- group checking --- >> >> ### --- user 101 --- >> >> 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "voip", >> Service-Type == "Group-Check" >> >> Reply-Message = "Authorized" >> >> 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended", >> Service-Type == "Group-Check" >> >> Reply-Message = "Authorized" >> >> 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "pstn", >> Service-Type == "Group-Check" >> >> Reply-Message = "Authorized" >> >> ### --- user 102 --- >> >> 102(a)mydomain.com Auth-Type := Accept, Sip-Group == "voip", >> Service-Type == "Group-Check" >> >> Reply-Message = "Authorized" >> >> DEFAULT Auth-Type := Reject, Service-Type == "Group-Check" >> >> ### --- user authentication --- >> >> 101(a)mydomain.com Auth-Type := Digest, User-Password == "101" >> >> Reply-Message = "Authenticated", >> >> Sip-Avp += "rpid:101", >> >> Sip-Avp += "#2:192.168.10.17", >> >> # Sip-Avp += "#2:192.168.2.11" >> >> 102(a)mydomain.com Auth-Type := Digest, User-Password == "102" >> >> Reply-Message = "Authenticated", >> >> Sip-Avp += "rpid:102", >> >> # Sip-Avp += "#2:192.168.2.12" >> >> ------------------------------------------------------------------------ >> >> *From:* Edson [mailto:4lists@gmail.com] >> *Sent:* Thursday, June 15, 2006 4:19 PM >> *To:* 'Hamid Ali Asgari'; users(a)openser.org >> *Subject:* RE: [Users] SIP Clients Can't REGISTER >> >> Take a look on the user credentials on Radius DB. Radius is rejecting >> the authentication, so there should be some mismatch between Your >> login credentials and what Radius has in its DB. >> >> Try to run this command and see what's the result (linux version): >> >> radiusclient -f <conf-dir-of-radiusclient-ng>/radiusclient.conf -p 123 >> 'User-Name=some(a)valid.user' 'Password=Guess';echo $? >> >> It should return: >> >> Reply-Message = 'Authenticated' >> >> Edson. >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

1 0

Re: [Serusers] Unresolvable Destination
by Greger V. Teigre 19 Jun '06

19 Jun '06

Oh yes, you just have to lookup the uri with domain name and translate it to an actual location (i.e. using lookup("location") or lookup("aliases")) Have a look at the SER Getting Started document from http://onsip.org/ (that one was for Olivier ;-) It will guide you through setting up SER. g-) Alois Schafferhofer wrote: > Thank you. > > Therefore I have two questions. > > Does it work without DNS server? > What about a Pocket PC? It can't be member of Active Directory. > > I hope, these questions don't seem too stupid for you. > > Regards, > Alois > > >> From: "Greger V. Teigre" <greger(a)teigre.com> >> To: Alois Schafferhofer <schaffi_lui(a)hotmail.com> >> CC: serusers(a)lists.iptel.org >> Subject: Re: [Serusers] Unresolvable Destination >> Date: Sat, 17 Jun 2006 20:58:46 +0200 >> >> This means that fh-joanneum.at is not possible to DNS resolve. A sip >> uri needs to be resolvable. g-) >> >> Alois Schafferhofer wrote: >>> Hi, >>> >>> I'm using Presence Snapshot 4.1. >>> >>> If I want to REGISTER my client, I get the error: Unresolvable >>> Destination. >>> >>> I have following lines in my log message: >>> >>> /usr/sbin/ser[8941]: Unknown domain to: sip:PDA_User@fh-joanneum.at >>> from: sip:PDA_User@fh-joanneum.at >>> /usr/sbin/ser[8941]: ERROR: mk_proxy: could not resolve hostname: >>> "fhjoanneum.at" >>> /usr/sbin/ser[8941]: ERROR: uri2proxy: bad host name in URI >>> <sip:PDA_User@fhjoanneum.at> >>> /usr/sbin/ser[8941]: ERROR: t_forward_nonack: failure to add branches >>> /usr/sbin/ser[8941]: sl_reply error() >>> /usr/sbin/ser[8941]: ERROR: sl_reply_error used: I'm terribly sorry, >>> server error occurred (1/SL) >>> >>> >>> >>> U 10.0.0.3:5060 -> 10.0.0.11:5060 >>> REGISTER sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP >>> 10.0.0.3:5060;branch=z9hG4bK479720758..Max-Forwards: 70..From: >>> <sip:PDA_User@fhjoanneu >>> m.at>;tag=1505826169..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: >>> 927318080- >>> 406265317-1110482780..CSeq: 1 REGISTER..Contact: >>> <sip:PDA_User@10.0.0.3>.. >>> Content-Length: 0..User-Agent: SIP .NET 1.0 evaluation version, >>> www.independentsoft.com.... >>> # >>> U 10.0.0.11:5060 -> 10.0.0.3:5060 >>> SIP/2.0 478 Unresolvable destination (478/TM)..Via: SIP/2.0/UDP >>> 10.0.0.3:5060;branch=z9hG4bK479720758..From: <sip:PDA_User@fhjoanneum. >>> at>;tag=1505826 >>> 169..To: >>> <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28- >>> 2449..Call-ID: 927318080-406265317-1110482780..CSeq: 1 >>> REGISTER..Server: S >>> ip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: >>> 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=8941 >>> req_src_ip=10.0 >>> .0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at >>> out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1".... >>> # >>> U 10.0.0.11:5060 -> 10.0.0.3:5060 >>> SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: >>> SIP/2.0/UDP >>> 10.0.0.3:5060;branch=z9hG4bK479720758..From: <sip:PDA_User@fh-joanneum. >>> at>;tag=1505826169..To: <sip:PDA_User@fhjoanneum. >>> at>;tag=56a704c9068c5b1b9228b761fe5fb2cc.2449..Call-ID: 927318080- >>> 406265317-1110482780..CSeq: 1 REGI >>> STER..Server: Sip EXpress router (0.10.99-dev35-pa-4.1 >>> (i386/linux))..Content- >>> Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=8941 >>> req_src_ip=10.0.0.3 req_src_port=5060 >>> in_uri=sip:PDA_User@fh-joanneum.at >>> out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1" >>> >>> I hope you can help me. >>> >>> Regards, >>> Alois >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> > > >

1 0

[Serusers] fr_inv_timer
by olivier.taylor 19 Jun '06

19 Jun '06

hi all, I have modparam("tm", "fr_inv_timer", 60) when a callee doesn't anwer, he often send a 487 (cancelled) before 60 seconds... It means timeout in fact or no answer... If the caller stop the call, I also get a 487... How to make the difference on a 487 generated by the caller and by the callee? It makes difference, if it's generated by the callee, I will forward the call to voicemail or another number, if it's generated by the caller, I will do nothing. When I set fr_inv_timer to 15, I always get a timeout and it's ok, but my customers dislikes that as well as the callees who needs to be better than Ben Jonhson for the 100meters :( Does exists a parameter to be send to the callee to tell him to wait 60 seconds + 1 before sending me a Cancel? Regards, Olivier

1 0

[Serusers] voip peering
by Mindaugas Kuprys 19 Jun '06

19 Jun '06

Hi, Is there anyone running ser for peering with other voip providers using IPERX, Neustar and Arbinet provided services? I'm thinking to do that or should better use other solution like enum server http://www.ag-projects.com/ENUM.html? Regards, Mindaugas

1 0

[Users] Sql possiblities of 1.0.1
by Fatih Cerit 19 Jun '06

19 Jun '06

Hi all Is it possible to do these with 1.0.1 SQL modules?? 1- is it possible to only insert the location queries instead of insert and update? 2- is it possible to lookup() from diffrent table (like a view) and save() location table? if possible which will queried from init_db. 3- is it possible to call function-select instead of a insert like select * from function_location('','','','','',''); thanks cerit

1 0

[Users] planet VIP 152 T
by Joao Pereira 19 Jun '06

19 Jun '06

Hello to all Im testing a Planet 152 T phone and Im having some problems. Can someone tell me if this phone does URI dialing? And does it work behind NAT (does it need any special configuration on the SIP server)? Thanks Joao Pereira

2 1

[Users] sip contact relay/forward/rewrite?
by Paul Smith 19 Jun '06

19 Jun '06

Hi, I am new to openser and trying to use it as a proxy i.e. I have a phone behind NAT, and another SIP server with all our account/registrar/routing details, I want to try mediaproxy as a far-end nat traversal solution to allow us to install phones without doing much (or anything) to the NAT firewalls.... The idea is that I set up phones to use Openser as Outbound proxy, and use symmetric RTP, and tell openser to nathelper to do regular keep alive pings. Then the phone should open SIP port to openser when registering, and OpenSER forwards the registration to the current sip server. Later invites from the current server should go back through Openser to the phone over the still open ip:port pair. MediaProxy on the openser should handle the RTP traffic between phone and old sip server. I have got ser and mediaproxy running and outgoing calls are fine. The problem is that incoming calls are not going through the proxy... the phone registers via the openser proxy, but when initiating an incoming call to the phone the registrar server sends the SIP invite straight to the NAT firewall, which rejects it... What do I need to do so that openSER tells the registrar to route SIP INVITES back through the openser server? I realise that at present I am looking at a very insecure open relay - I'll add more auth and domain checking when I get the basic outbound relay / media proxy bit working. PHONE ---- NAT ---- SER ----- SIP registrar Server with user details etc. Thanks.

2 1

[Users] Can't forward to PSTN gateway
by Kenny Chua 19 Jun '06

19 Jun '06

Hi all, I'm having trouble allowing calls to my pstn gateway. I get some errors. These are the errors I get: I get the 404 Not Found error Here is the information I got from ngrep: U 2006/06/17 02:58:31.372496 68.xxx.xxx.17:5060 -> 192.168.1.20:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..From: K enny <sip:201@mydomain.myftp.org>;tag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7(a)192.168.1.20.. CSeq: 60451 INVITE..Server: OpenSer (1.0.1 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.5:5060 "Noisy feedback tells: pid=2874 req_src_ip=1 92.168.1.1 req_src_port=5060 in_uri=sip:13373038435@mydomain.myftp.org out_uri=sip:13373038435@pstn.gateway.net:5060 via_cnt==1".... U 2006/06/17 02:58:31.487474 192.168.1.5:5060 -> 65.xxx.xxx.xxx:5060 INVITE sip:13373038435@pstn.gateway.net:5060 SIP/2.0..Record-Route: <sip:192.168.1.5;ftag=2300590181;lr=on>..Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK 7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..From: Kenny <sip:201@mydomain.myftp.org>;t ag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Contact: <sip:201@192.168.1.1:5060>..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7(a)192.168.1.20 ..CSeq: 60451 INVITE..Max-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 304....v=0..o=201 10653130 1065 3152 IN IP4 68.226.254.17..s=X-Lite..c=IN IP4 192.168.1.1..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rt pmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. U 2006/06/17 02:58:31.545023 65.200.189.132:5060 -> 192.168.1.5:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 68.226.254.17:5060;branch=z9hG4bK7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837 F1736B4E11832482B6D09012F5..From: Kenny <sip:201@mydomain.myftp.org>;tag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Call-ID: 4B570DFD-0E2E-4F 32-8B4B-05ADC419D0E7@192.168.1.20..CSeq: 60451 INVITE..Content-Length: 0.... U 2006/06/17 02:58:31.661353 65.200.189.132:5060 -> 192.168.1.5:5060 SIP/2.0 500 Server Internal Error..To: <sip:13373038435@mydomain.myftp.org>;tag=3359519831-260907..From: Kenny <sip:201@mydomain.myftp.org>;tag=23005901 81..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7@192.168.1.20..CSeq: 60451 INVITE..Contact: sip:13373038435@65.xxx.xxx.xxx:5060..Via: SIP/2.0/UDP 68.xx x.xxx.17:5060;branch=z9hG4bK7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..Content-Length : 0.... U 2006/06/17 02:58:31.661842 192.168.1.5:5060 -> xx.xxx.xxx.132:5060 ACK sip:13373038435@pstn.gateway.net:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7217.285e6331.0..From: Kenny <sip:201@mydomain.myftp.org>; tag=2300590181..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7@192.168.1.20..To: <sip:13373038435@mydomain.myftp.org>;tag=3359519831-260907..CSeq: 60451 ACK..User-Agent: OpenSer (1.0.1 (i386/linux))..Content-Length: 0.... Below is my openser.cfg # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ # put your server IP address here listen=192.168.1.5 alias=mydomain.myftp.org check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (method=="REGISTER") { xlog("Registering\n"); route(2); return; }; lookup("aliases"); if(method =="INVITE"){ if( uri=~"^sip:1[0-9]*@"){ xlog("International call\n"); route(4); exit; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("mydomain.myftp.org","subscriber")) { www_challenge("mydomain.myftp.org","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; xlog("Registered Successfully\n"); } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[4]{ xlog("Before forward to host\n"); rewritehostport("pstn.gateway.net:5060"); xlog("After forward to host\n"); route(1); } What am I doin wrong? Why am I not able to dial a PSTN line? My PSTN gateway allows international calls. Any help is appreciated. Thank you. --------------------------------- Ring'em or ping'em. Make PC-to-phone calls as low as 1¢/min with Yahoo! Messenger with Voice.

2 1

[Users] lawful legal intercept
by tele 19 Jun '06

19 Jun '06

hi, all is it possible to implement with an openser a legal interception? maybe with t_replicate() to another server. what about the media forking of the call? thanks :tele

2 3

[Users] SIP Clients Can't REGISTER
by Hamid Ali Asgari 19 Jun '06

19 Jun '06

Hi, I have followed the Intrsuctions on setting up FreeRADIUS with OpenSER. (http://openser.org/docs/openser-radius-1.0.x.html) But the server doesn't allow the registration: Config: ============================================================================ ==================== # $Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=udp:192.168.10.18 alias="openser.org" alias="mydomain.com" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- modparam("avpops", "avp_aliases", "day=i:101;time=i:102") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # check if user is suspended if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) { if (radius_is_user_in("From", "suspended")) { sl_send_reply("403", "Forbidden - suspended"); exit; }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (!uri==myself) { # check if user is allowed to do voip calls to other domains if(is_method("INVITE|MESSAGE")) { if (!radius_is_user_in("From", "voip")) { sl_send_reply("403", "Forbidden VoIP"); exit; }; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("mydomain.com")) { www_challenge("mydomain.com", "0"); exit; }; # check the src ip address if(!avp_check("i:2", "eq/$src_ip/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; # calls to pstn if(uri=~"sip:00[1-9][0-9]+@") { if(is_method("INVITE") && !has_totag()) { if (!radius_is_user_in("From", "pstn")) { sl_send_reply("403", "Forbidden PSTN"); exit; }; }; # set gateway address rewritehostport("192.168.10.18:5090"); route(1); }; # load callee's avps if(avp_load_radius("callee")) { # check if user has time filter enabled if(avp_check("i:3", "eq/i:1")) { # print time in an avp avp_printf("i:100", "$Tf"); # extract day avp_subst("i:100/i:101", "/(.{3}) .+/*\1*/"); if(!avp_check("i:6", "fm/$day")) { sl_send_reply("403", "Forbidden - day"); exit; }; # extract 'hours:minutes' avp_subst("i:100/i:102", "/(.{10}) (.{5}):.+/\2/"); if((is_avp_set("i:4") && avp_check("i:4", "gt/$time")) || (is_avp_set("i:5") && avp_check("i:5", "lt/$time"))) { sl_send_reply("403", "Forbidden - time"); exit; }; }; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # generic forward route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } # ============================================================================ ==================== Users file on RADIUS: ### --- avps --- 101(a)mydomain.com Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" Sip-Avp += "#3#1", Sip-Avp += "#4:08:00", Sip-Avp += "#5:16:00", Sip-Avp += "#6:Mon,Wed,Thu,Fri" 102(a)mydomain.com Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" Sip-Avp += "#3#1", Sip-Avp += "#4:08:00", Sip-Avp += "#5:16:00", Sip-Avp += "#6:Mon,Wed,Thu,Free" DEFAULT Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" ### --- group checking --- ### --- user 101 --- 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "voip", Service-Type == "Group-Check" Reply-Message = "Authorized" 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "pstn", Service-Type == "Group-Check" Reply-Message = "Authorized" ### --- user 102 --- 102(a)mydomain.com Auth-Type := Accept, Sip-Group == "voip", Service-Type == "Group-Check" Reply-Message = "Authorized" DEFAULT Auth-Type := Reject, Service-Type == "Group-Check" ### --- user authentication --- 101(a)mydomain.com Auth-Type := Digest, User-Password == "101" Reply-Message = "Authenticated", Sip-Avp += "rpid:101", Sip-Avp += "#2:192.168.10.17", # Sip-Avp += "#2:192.168.2.11" 102(a)mydomain.com Auth-Type := Digest, User-Password == "102" Reply-Message = "Authenticated", Sip-Avp += "rpid:102", # Sip-Avp += "#2:192.168.2.12" ============================================================================ ==================== And bellow is the dubug from radiusd -X and openser -E -ddddddd: OpenSER Debug: ***************************** 0(4084) SIP Request: 0(4084) method: <REGISTER> 0(4084) uri: <sip:mydomain.com> 0(4084) version: <SIP/2.0> 0(4084) parse_headers: flags=2 0(4084) DEBUG:parse_to:end of header reached, state=8 0(4084) DEBUG: get_hdr_field: <To> [19]; uri=[sip:101@mydomain.com] 0(4084) DEBUG: to body [sip:101@mydomain.com ] 0(4084) get_hdr_field: cseq <Cseq>: <2> <REGISTER> 0(4084) Found param type 232, <branch> = <z9hG4bK-eb137d5b6c8524d7c778e5629555a97e>; state=16 0(4084) end of header reached, state=5 0(4084) parse_headers: Via found, flags=2 0(4084) parse_headers: this is the first via 0(4084) After parse_msg... 0(4084) preparing to run routing scripts... 0(4084) DEBUG:maxfwd:is_maxfwd_present: value = 20 0(4084) DEBUG: add_param: tag=30167a95 0(4084) DEBUG:parse_to:end of header reached, state=29 0(4084) radius_is_user_in(): Failure 0(4084) parse_headers: flags=200 0(4084) DEBUG: get_hdr_body : content_length=0 0(4084) found end of header 0(4084) find_first_route: No Route headers found 0(4084) loose_route: There is no Route HF 0(4084) grep_sock_info - checking if host==us: 9==10 && [mydomain.com] == [192.168.10.18] 0(4084) grep_sock_info - checking if port 5060 matches port 5060 0(4084) grep_sock_info - checking if host==us: 9==10 && [mydomain.com] == [192.168.10.18] 0(4084) grep_sock_info - checking if port 5060 matches port 5060 0(4084) check_nonce(): comparing [448fe8934593e3e7780fffee019ce58577231baf] and [448fe8934593e3e7780fffee019ce58577231baf] 0(4084) DEBUG:auth_radius:radius_authorize_sterman: Success 0(4084) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(4084) DEBUG:auth_radius:extract_avp: string is <rpid:101> 0(4084) DEBUG:auth_radius:extract_avp: AVP name is <rpid> 0(4084) DEBUG:auth_radius:extract_avp: AVP val is <101> 0(4084) DEBUG:auth_radius:generate_avps: AVP 'rpid'/0='101'/0 has been added 0(4084) DEBUG:auth_radius:extract_avp: string is <#2:192.168.10.17> 0(4084) DEBUG:auth_radius:extract_avp: AVP name is <2> 0(4084) DEBUG:auth_radius:extract_avp: AVP val is <192.168.10.17> 0(4084) DEBUG:auth_radius:generate_avps: AVP 'null'/2='192.168.10.17'/0 has been added 0(4084) DEBUG:avpops:check_avp: check <192.168.10.17> against <192.168.10.17> as str 0(4084) parse_headers: flags=ffffffffffffffff 0(4084) parse_headers: flags=4000000 0(4084) build_contact(): Created Contact HF: Contact: <sip:101@192.168.10.17;LINEID=6f5beb97d08670a1c8328ce1526ae928>;expires=300 0(4084) parse_headers: flags=ffffffffffffffff 0(4084) check_via_address(192.168.10.17, 192.168.10.17, 0) 0(4084) DEBUG:destroy_avp_list: destroying list 0xb60f0a50 0(4084) receive_msg: cleaning up 1(4086) build_recv_sock: <192.168.10.18_5060> ******************************* RADIUS Debug: ******************************* rad_recv: Access-Request packet from host 127.0.0.1:32900, id=196, length=185 User-Name = "101(a)shatel.ir" Digest-Attributes = 0x0a05313031 Digest-Attributes = 0x010b73686174656c2e6972 Digest-Attributes = 0x022a3434386666316666393332663830393337613034373266333837363163353036656166 3636613934 Digest-Attributes = 0x040f7369703a73686174656c2e6972 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "7a3f00f697286dd95c5aa654a9662dea" Service-Type = Sip-Session Sip-Uri-User = "101" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 100 modcall[authorize]: module "preprocess" returns ok for request 100 modcall[authorize]: module "chap" returns noop for request 100 modcall[authorize]: module "mschap" returns noop for request 100 rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 100 rlm_realm: Looking up realm "shatel.ir" for User-Name = "101(a)shatel.ir" rlm_realm: No such realm "shatel.ir" modcall[authorize]: module "suffix" returns noop for request 100 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 100 users: Matched entry DEFAULT at line 152 users: Matched entry 101(a)shatel.ir at line 255 modcall[authorize]: module "files" returns ok for request 100 modcall: leaving group authorize (returns ok) for request 100 rad_check_password: Found Auth-Type Digest auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 100 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "101" Digest-Realm = "shatel.ir" Digest-Nonce = "448ff1ff932f80937a0472f38761c506eaf66a94" Digest-URI = "sip:shatel.ir" Digest-Method = "REGISTER" A1 = 101:shatel.ir:101 A2 = REGISTER:sip:shatel.ir H(A1) = a5d8cf73217ea46c8fd8ca5da1c6a2e3 H(A2) = 27e9df2d1a7ed715c7204d1316039120 KD = a5d8cf73217ea46c8fd8ca5da1c6a2e3:448ff1ff932f80937a0472f38761c506eaf66a94:27 e9df2d1a7ed715c7204d1316039120 EXPECTED 7a3f00f697286dd95c5aa654a9662dea RECEIVED 7a3f00f697286dd95c5aa654a9662dea modcall[authenticate]: module "digest" returns ok for request 100 modcall: leaving group authenticate (returns ok) for request 100 radius_xlat: 'Authenticated' Sending Access-Accept of id 196 to 127.0.0.1 port 32900 Reply-Message = "Authenticated" SIP-AVP += "rpid:101" SIP-AVP += "#2:85.15.7.17" Finished request 100 Going to the next request Waking up in 2 seconds... rad_recv: Access-Request packet from host 127.0.0.1:32901, id=197, length=64 User-Name = "101(a)shatel.ir" Sip-Group = "suspended" Service-Type = Group-Check NAS-Port = 0 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 101 modcall[authorize]: module "preprocess" returns ok for request 101 modcall[authorize]: module "chap" returns noop for request 101 modcall[authorize]: module "mschap" returns noop for request 101 modcall[authorize]: module "digest" returns noop for request 101 rlm_realm: Looking up realm "shatel.ir" for User-Name = "101(a)shatel.ir" rlm_realm: No such realm "shatel.ir" modcall[authorize]: module "suffix" returns noop for request 101 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 101 users: Matched entry DEFAULT at line 152 users: Matched entry DEFAULT at line 252 modcall[authorize]: module "files" returns ok for request 101 modcall: leaving group authorize (returns ok) for request 101 rad_check_password: Found Auth-Type Reject rad_check_password: Auth-Type = Reject, rejecting user auth: Failed to validate the user. Delaying request 101 for 1 seconds Finished request 101 Any thoughts? Thanks, Hamid

3 5

[Serusers] Error message: Header V3 DSA signature:
by S.A.S. Lorensuhewa 19 Jun '06

19 Jun '06

Dear All, I got following error message while installing ser rpm. What is the meaning of this. arning: ser-0.9.6-6.fc5.i386.rpm: Header V3 DSA signature: NOKEY, key ID 1ac70ce6 Aruna ---------------------------------------------------------- S. A. S. Lorensuhewa Head, Department of Computer Science University of Ruhuna, Matara, Sri Lanka Telephone: 0094 (0)41 2227019 Fax: 0094 (0)41 2222683 Email: lorensuhewa(a)cc.ruh.ac.lk, aruna(a)cc.ruh.ac.lk

2 1

[Serusers] Unsupported document format
by Alois Schafferhofer 19 Jun '06

19 Jun '06

Hi, I use Presence Snapshot 4.1, and I want my Server to store x and y coordinate and send NOTIFY to the SUBSCRIBER whenever the position changes. I tried it with geopriv location but I get the answer: "Unsupported document format". Is there an error in my PUBLISH request or isn't it possible to store x and y? And in which table does it store? I added the collumns x, y in presentity, but SER ignores them. I hope, you can help me. Thanks. /usr/sbin/ser[9955]: handle publish request /usr/sbin/ser[9955]: ERROR: pidf.c:467: can't parse document /usr/sbin/ser[9955]: can't parse CPIM-PIDF document /usr/sbin/ser[9955]: handled publish U 10.0.0.3:5060 -> 10.0.0.11:5060 PUBLISH sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1845246585..Max-Forwards: 70..From: <sip:PDA_User@fhjoanneu m.at>;tag=1986441488..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: 1734692618- 1859786556-1651516169..CSeq: 6 PUBLISH..Contact: <sip:PDA_User@10.0.0.3>. .Content-Type: application/cpim-pidf+xml..Content-Length: 307..Expires: 7200..Event: presence..User-Agent: SIP .NET 1.0 evaluation version, www.independentsoft.com.... <?xml version="1.0 encoding="utf-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf"entity="sip:PDA_User@fh-joanneum.at"> <tuple id=9r28r59> <status> <basic>online</basic> <geopriv><locationinfo><civilAddress><x>01525.0125</x><y>04704.2002</y></civilAddress></locationinfo></geopriv> </status> </tuple> </presence> U 10.0.0.11:5060 -> 10.0.0.3:5060 SIP/2.0 415 Unsupported document format..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1845246585..From: <sip:PDA_User@fhjoanneum.at>;tag=1986441488.. To: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-a814..Call-ID: 1734692618-1859786556-1651516169..CSeq: 6 PUBLISH..Server: Sip E Xpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=9955 req_src_ip=10.0.0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fhjoanneum. at via_cnt==1" Regards, Alois

2 1

[Users] invalid port returned by mediaproxy
by Kenny Chua 19 Jun '06

19 Jun '06

Hello, I'm trying to implement mediaproxy with my openser. I'm getting this error: local /usr/local/sbin/openser[2781]: error: use_media_proxy(): invalid port returned by mediaproxy: # I've forwarded all ports that are required. I'm having problems with sip2sip (no audio). Also, for SIP to PSTN, I'm having problems too. The call goes through the PSTN gateway and actually connects. However, its still one way audio or sometimes no audio at all. I'm using FC5 with openser 1.0.1. Mediaproxy 1.7.2. Below are my configuration files for media proxy and openser.cfg: **************************************Openser.cfg********************************* debug=3 fork=yes log_stderror=no listen=192.168.1.5 # This is where OpenSER installed port=5060 children=4 alias=mydomain.com # This is where OpenSER installed dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transaction_flag", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("192.168.1.5:5060;nat=yes"); # This is where OpenSER installed } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); return; }; if (uri==myself) { if (method=="CANCEL") { route(3); return; } else if (method=="INVITE") { xlog("URI is myself, forward to route 3\n"); route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("mydomain.com","subscriber")) { www_challenge("mydomain.com","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- xlog("Call Invite\n"); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri=~"^sip:1[0-9]*@*") { xlog("PSTN Gateway\n"); use_media_proxy(); rewritehostport("pstn.gateway.com:5060"); # This is where PSTN route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; if (method=="CANCEL") { route(1); exit; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ***************************Mediaproxy.ini****************************************** [Dispatcher] ;start = yes socket = /var/run/proxydispatcher.sock ;group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = ser listen = None allow = None proxyIP = 192.168.1.5 # This is where OpenSER installed portRange = 21000:35000 # These values are set same in ATA's configs for RTP Range TOS = 0xb8 idleTimeout = 60 holdTimeout = 180 forceClose = 0 accounting = off [Accounting] user = root password = host = localhost database = ser table = acc Can anyone figure out my problem? Thank you in advance __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] serusers request
by rekik refka 18 Jun '06

18 Jun '06

Hi, I'm trying to use jabber module. I didn't know how to configure a DNS server with ser. I retreive some problem with a jabber server.'( file "jabber.xml") I would like to test tow clients one msn and another jabber to exhange an instant message. How can I do it. Can you help me! Thanks in advance.

1 0

[Serusers] Re: mediaproxy problem
by Andrey Kouprianov 18 Jun '06

18 Jun '06

Yup. Me too. I hope to get some comments on this... :) Anyone? On 6/17/06, Hakan YASTI <hakanyasti(a)gmail.com> wrote: > > Hi Andrey, > > I have the same problem with proxy dispatcher. Have got a solution for your > problem. I couldn't see any solution for your problem. > > Hope to hera soon. > > Best Regards, > Hakan. > > > > > Hi, I have tried to install mediaproxy 1.7.2. So, i did. Configuration > was about the same as 1.4.2, however, "mediaproxy start", just > started mediaproxy.py, but not proxydispatcher.py. Version 1.4.2 started > them both w/o any problem. Trying to start proxydispatcher.py > independently did not give any results. /var/run/proxydispatcher.sock is > simply not there anymore. There are no logs or whatsoever, that I could show > you, because I did not find any. Anyway, here's the config > (NOTE: mediaproxy's running on the same machine as SER): [Dispatcher] start > = yes socket = /var/run/proxydispatcher.sock listen = <my IP address> group > = wheel defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = > yes socket = /var/run/mediaproxy.sock group = wheel listen = <my IP > address> allow = <my IP address>, 127.0.0.1 proxyIP = <my IP > address> portRange = 60000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = > 3600 forceClose = 10800 [Accounting] dbaccounting = off //<my IP address> > is IP of SER/mediaproxy machine // the rest is irrelevant Please, also note > that I dont have a DNS SRV entry for media proxy. Could that cause the > problem? One way or another, I had to downgrade back to 1.4.2. Any > suggestions? Thanx in advance. Andrey. >

2 2

[Users] voice is not hearing...
by raviprakash sunkara 18 Jun '06

18 Jun '06

Hi , please help me on openser . I'm working on openser , Voice is clearly listening in softphone <----> softphones .but when comes to softphone <-------> hardphones and hardphones <------> hardphones.. I use nathelper module... also... voice is not clearly.... please hel ,What i need to do. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

1 0

[Serusers] sip library
by Felipe Louback 17 Jun '06

17 Jun '06

Hi! I have to write an application for a college project that, in some ways, will behavior like a SIP Proxy. Could any one poing me a library that already implements SIP? It could be C, C++, Perl.... Thanks in advance, Felipe Louback -- Mestrando - Departamento de Engenharia Elétrica Grupo de Pesquisa em Engenharia da Computação e Telecomunicações Universidade Federal de Minas Gerais - Brasil "Vinde a mim vós todos que estai cansados e aflitos, e eu vos aliviarei." Mateus 11:28

3 4

[Serusers] Unresolvable Destination
by Alois Schafferhofer 17 Jun '06

17 Jun '06

Hi, I'm using Presence Snapshot 4.1. If I want to REGISTER my client, I get the error: Unresolvable Destination. I have following lines in my log message: /usr/sbin/ser[8941]: Unknown domain to: sip:PDA_User@fh-joanneum.at from: sip:PDA_User@fh-joanneum.at /usr/sbin/ser[8941]: ERROR: mk_proxy: could not resolve hostname: "fhjoanneum.at" /usr/sbin/ser[8941]: ERROR: uri2proxy: bad host name in URI <sip:PDA_User@fhjoanneum.at> /usr/sbin/ser[8941]: ERROR: t_forward_nonack: failure to add branches /usr/sbin/ser[8941]: sl_reply error() /usr/sbin/ser[8941]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) U 10.0.0.3:5060 -> 10.0.0.11:5060 REGISTER sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK479720758..Max-Forwards: 70..From: <sip:PDA_User@fhjoanneu m.at>;tag=1505826169..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: 927318080- 406265317-1110482780..CSeq: 1 REGISTER..Contact: <sip:PDA_User@10.0.0.3>.. Content-Length: 0..User-Agent: SIP .NET 1.0 evaluation version, www.independentsoft.com.... # U 10.0.0.11:5060 -> 10.0.0.3:5060 SIP/2.0 478 Unresolvable destination (478/TM)..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK479720758..From: <sip:PDA_User@fhjoanneum. at>;tag=1505826 169..To: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28- 2449..Call-ID: 927318080-406265317-1110482780..CSeq: 1 REGISTER..Server: S ip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=8941 req_src_ip=10.0 .0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1".... # U 10.0.0.11:5060 -> 10.0.0.3:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK479720758..From: <sip:PDA_User@fh-joanneum. at>;tag=1505826169..To: <sip:PDA_User@fhjoanneum. at>;tag=56a704c9068c5b1b9228b761fe5fb2cc.2449..Call-ID: 927318080- 406265317-1110482780..CSeq: 1 REGI STER..Server: Sip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content- Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=8941 req_src_ip=10.0.0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1" I hope you can help me. Regards, Alois

2 1

[Serusers] CDRTool+Asterisk
by hgaillac-sip＠yahoo.fr 17 Jun '06

17 Jun '06

Hello, I wish to use CDRTool with both asterisk and ser however I get a lot of errors with cdrtool. I spend time to read scripts without success . Anybody is used to install CDRTool ? Regards Harry ___________________________________________________________________________ Yahoo! Mail réinvente le mail ! Découvrez le nouveau Yahoo! Mail et son interface révolutionnaire. http://fr.mail.yahoo.com

2 1

[Serusers] query about routing
by Aman Sachdeva 17 Jun '06

17 Jun '06

sir, I am aman sachdeva working in positive comsole on voip project. i am using ur server(sip express router) for trial. sir i need ur help to fulfill my requirement is that to add gateway, that means my client call using softphone and sip express router server , route that call to to x.x.x.x(ip). the server is install in other ip. such as clent---->dial a number using softphone(9891713625)----------->sip express royter server is install on y.y.y.y,add prefix and routing ip( xxxxx919891713625(a)x.x.x.x)-------->x.x.x.x(ip) server. i found a script file in ser0.9.6(scripts) folder serconf.sh and i run this file with the parameter dominname and routing ip. but it dosenot route to x.x.x.x(routing ip). could u please tell me where i can change in the source code or script file so that my call is routing to other server. thanks Regards Aman Sachdeva

2 1

[Serusers] Out-of-office-reply: Serusers Digest, Vol 38, Issue 18
by David.Monahan＠aepona.com 17 Jun '06

17 Jun '06

I will be out of the office until 28th June. During this time I will have no email access. In my absence please contact: Steven Egan (steven.egan(a)aepona.com) for project related issues Stephen Drew (stephen.drew(a)aepona.com) for technical issues Regards, David

1 0

[Users] Error while preloading domain 'location'
by zhan li 17 Jun '06

17 Jun '06

Hello klaus, I tried to start openser with acc modules, The error message like this: 0(0) fixing /usr/local/lib/openser/modules/registrar.so save 0(0) db_init: Connection 'mysql://openser:openserrw@localhost/openser' not found in pool 0(0) new_connection: Opening MySQL connection: mysql://openser:openserrw@localhost/openser 0(0) new_connection: Connection type is Localhost via UNIX socket 0(0) new_connection: Protocol version is 10 0(0) new_connection: Server version is 4.0.18-standard 0(0) usrloc:preload_udomain: Wrong version v136545352 for table <location>, expected v1001 0(0) register_udomain(): Error while preloading domain 'location' 0(0) pool_remove: Removing connection from the pool 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration I even tried to openser_mysql drop openser_mysql create after this, I tried again, still same error message. I am using the same configure on ser, no problem, but openser dose not allow me to use acc modules. Please give me help! Thanks John

2 2

[Serusers] Cisco 7960 don't REGISTER
by Ricardo Carvalho 16 Jun '06

16 Jun '06

Hi, I'm trying to get my Cisco 7960 working with my SIP server running SER-0.9.0, but it never registers in the server. When I run "ngrep sip" I always get "401 Unauthorized" responses sent from server to phone. I don't understand why is it happening because I think it's everything well configured. What could be causing this problem? Regards, Ricardo. ngrep log: U 192.168.6.16:50988 -> 192.168.10.90:5060 REGISTER sip:iric.up.pt SIP/2.0..Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK48104c63..From: <sip:105711@iric.up.pt>;tag=00036baad2f4 00d600629c68-0c16f516..To: sip:105711@iric.up.pt..Call-ID: 00036baa-d2f40009-20a38f54-53f31005@192.168.6.16..Max-Forwards: 70..Date: Fri, 16 Jun 2006 10:29:05 GMT..CSeq: 224 REGISTER..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:105711@192.168.6.16:5060;transport=udp>;+sip.in stance="<urn:uuid:00000000-0000-0000-0000-00036baad2f4>";+u.sip!model.ccm.cisco.com="7"..Content-Length: 0..Expires: 3600.... # U 192.168.10.90:5060 -> 192.168.6.16:50988 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK48104c63;rport=50988..From: <sip:105711@iric.up.pt>;tag=00036ba ad2f400d600629c68-0c16f516..To: sip:105711@iric.up.pt;tag=11df39ed0e444255bd65b6f6af831c6f.4db5..Call-ID: 00036baa-d2f40009-20a38f54-53f31 005@192.168.6.16..CSeq: 224 REGISTER..WWW-Authenticate: Digest realm="iric.up.pt", nonce="449288cfbfc18bc31d059bf8b935352cf56ec883"..Serve r: Sip EXpress router (0.9.0 (i386/linux))..Content-Length: 0....

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Re: [Users] Problem with TLS
by Javier Ramirez 16 Jun '06

16 Jun '06

I search in openser to user.conf and I go to search more, but somebody can send me an example to user.conf to openser with TLS support ? best regards ! ----- Original Message ----- From: matty91(a)gmail.com To: Javier Ramirez Sent: Thursday, June 15, 2006 3:34 PM Subject: Re: [Users] Problem with TLS I do not have one handy. You should be able to snag one from openser.org, or from the mailing list archives. - Ryan -- UNIX Administrator http://daemons.net/~matty On Thu, 15 Jun 2006, Javier Ramirez wrote: > > please, can you send me an example to user.conf ? > > > ----- Original Message ----- > From: matty91(a)gmail.com > To: Javier Ramirez > Sent: Thursday, June 15, 2006 12:35 PM > Subject: Re: [Users] Problem with TLS > > > > On Thu, 15 Jun 2006, Javier Ramirez wrote: > > > I don't understand is: > > > > commonName = somename.somewhere.com > > > > to administrator of server or to one for one all users ? > > The CommonName is the name embedded in a certificate to identify the > service hosting the secured service. Typically this is a FQDN, so > www.cnn.com for example would have a CommonName of www.cnn.com. You > can also use X.500 names, but that doesn't make much sense in most > environments. > > Hope this helps, > - Ryan > -- > UNIX Administrator > http://daemons.net/~matty > > > __________ NOD32 1.1600 (20060615) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > > __________ NOD32 1.1600 (20060615) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com

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[Users] ser doesn't anwer to that request...
by Olivier Taylor 16 Jun '06

16 Jun '06

Hi there, Strange beahvior or I am Blind... This phone try to register and ser doesn't answer with a wwwauthenticate, doesn't answer at all in fact... all ideas are welcome :) Olivier U 213.219.xxx.xxx:10002 -> 82.146.xxx.xxx:5060 REGISTER sip:82.146.xxx.xx SIP/2.0. Via: SIP/2.0/UDP 213.219.xxx.xxx:10002branch=z9hG4bK8fb09b20aace531cb21a1abd57c0957d;rport. From: Astel.be Greg <sip:995045099@82.146.xxx.xx>;tag=3528839207. To: Astel.be Greg <sip:995045099@82.146.xxx.xx>. Call-ID: 705145430@192_168_2_2. CSeq: 1 REGISTER. Contact: Astel.be Greg <sip:995045099@213.219.xxx.xxx:10002. Max-Forwards: 70. User-Agent: Siemens Gigaset C450 IP. Expires: 180. Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO. Content-Length: 0. .

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[Serusers] Help About Accounting
by Robert Zorop 16 Jun '06

16 Jun '06

Hi, Is there anybody who could help me with CDRTool configuration? . I can't see the CDRs normalized with price OK. Using lastest stable SER, mysql, freeradius, and lastes stable CDRtool. Thanks a lot.

1 0

[Users] Does OpenSER support more than one CPU, performance testing procedure
by Mark Kent 16 Jun '06

16 Jun '06

>> I have a small configuration (pstn gateway ser.cfg) running ... One of the things you have to be careful of with these tests is just how realistic is the test? We can come up with all these magic numbers about cps and #users, etc., but if the configs don't include sufficient security checks to make it safe to put such a server on the Internet then it could be a bit misleading. I like openser and I'm enjoying learning about it and experimenting. But I'm afraid that it's going to get a black-eye in the press due to some significant security scandal. We need to put together a "Security Considerations" section for the documentation, so that would at least increase awareness among users (I'll work with someone to do this, but I don't know enough yet to do it alone.) Now some of you are going to say: Hey, use a commercial package if you can't program the necessary security! But it's not me I'm worried about, it's everyone else (OK, I am worried about me too :-) Consider the path of sendmail. It took a long time for sendmail to "ship" with arbitrary email relaying turned OFF by default. This created big problems in the real world. Likewise look how long it took for the default inetd.conf to have virtually nothing defined in it, or for inetd itself to be turned off by default. Previous default configurations caused big problems in the real world. The only security note in openser I recall seeing in two months of reading and experimenting is about preventing someone from registering with the same IP as one of your gateways. Here's the thing: if you give people working configs then they will use them and some will use them on the open Internet. How does one person's deficient openser.cfg affect you? I don't know, but one thing I hope we've all learned from the past dozen of so years of Internet growth is that security at site A can be exploited to affect site B in ways that we hadn't previously thought of. Think of spam, distributed denial of service attacks, identity spoofing. Thanks, -mark

1 0

RE: [Serusers] Rtpproxy howto
by İlker Aktuna $Koç.net$ 16 Jun '06

16 Jun '06

Hi, In ser v10, I see a mediaproxy module and it points me to following address for a mediaproxy. http://mediaproxy.ag-projects.com/ Could it be that this is the recent version of rtpproxy ? should I use rtpproxy or mediaproxy ? Thanks, ________________________________ From: ram [mailto:talk2ram@gmail.com] Sent: Friday, June 16, 2006 5:12 PM To: İlker Aktuna (Koç.net) Subject: Re: [Serusers] Rtpproxy howto Hi In my i said get the Source of ser0.9.7 ( do not install ser of this version) inside you can see rtpproxy source directory install from that only rtpproxy. ram On 6/16/06, İlker Aktuna (Koç. net ) <ilkera(a)koc.net> wrote: Hi, I can't use ser0.9.7 because I need presence. I already have ser v10 installed and running. How can I add rtpproxy to it ? Thanks, ilker ________________________________ From: ram [mailto:talk2ram@gmail.com] Sent: Friday, June 16, 2006 5:01 PM To: İlker Aktuna (Koç.net <http://koç.net/> ) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Rtpproxy howto Hi just download from onsip site ser0.9.7 untar that, and make rtpproxy from that and run rtpproxy its very simple ram On 6/16/06, İlker Aktuna (Koç. net ) <ilkera(a)koc.net > wrote: Hi, I will try to install RTP proxy for my internal users. Is there a simple RTP proxy howto/handbook for this process ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/020C0F07/04004D09/01020F4E/038…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________ <http://387555.sigclick.mailinfo.com/sigclick/020D0506/05044E03/01030A4F/022…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

[Users] Does OpenSER support more than one CPU, performance testing procedure
by rod 16 Jun '06

16 Jun '06

Hello all, I'm guessing if OpenSER supports more than one CPU. I ask for this to know if I can achieve to double the call per second (cps) rate or registration rate using two processors instead of one. I suppose that many of us have read the excellent "getting started 5" guide and use the scripts provided in this guide to start their first SER/OpenSER proxy. I would like to know if other users using these scripts would like to share their experience on the amount of users their OpenSER platform is able to handle, and on which hardware. It could be interesting if some users can post their OpenSER performance (cps, registration rate, number of user) with a particular "getting started" script, for example : "Authenticating ser.cfg", "call forwarding ser.cfg". Moreover, it could be great if we could think of a standard procedure to stress test the proxy. I have heard of Sipp, so do you think this program could be the base of this procedure? I understand that OpenSER performance greatly depends of the ser.cfg, nat configuration, avpops, the database backend, but I think that it could be great to delimit some standards configurations and show what we can expect on different hardware specifications. I'm wondering about this, cause I'd like to know if one or two OpenSER will easily support 30 000 to 40 000 users, call forwarding, redirection to voicemail; or will I have to buy a commercial product to do this. I have a small configuration (pstn gateway ser.cfg) running with 50 users, registering every 60s and a peak of 20cps (limited by my pstn connectivity), running like a charm on a small Intel P3 700 Mhz with 256MB (it's a small test box running on my desk). If somebody could help to define a test, it will be a pleasure to share my experience on an IBM HS20 blade with a bi Xeon 3,06 Ghz and 4GB of RAM. Thanks to all for the daily support, rod.

3 3

[Users] UsrLoc version dose not match Location
by zhan li 16 Jun '06

16 Jun '06

Hi All, When I tried to start openser with accounting module, I got error: 0(0) db_init: Connection 'mysql://openser:openserrw@localhost/openser' not found in pool 0(0) new_connection: Opening MySQL connection: mysql://openser:openserrw@localhost/openser 0(0) new_connection: Connection type is Localhost via UNIX socket 0(0) new_connection: Protocol version is 10 0(0) new_connection: Server version is 4.0.18-standard 0(0) usrloc:preload_udomain: Wrong version v136545352 for table <location>, expected v1001 0(0) register_udomain(): Error while preloading domain 'location' 0(0) pool_remove: Removing connection from the pool 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration I tried to execute : openser_mysql drop openser_mysql create After this, I start openser, the same erroe message. I realy need help in this issue. Thanks John

2 1

RE: [Serusers] Rtpproxy howto
by İlker Aktuna $Koç.net$ 16 Jun '06

16 Jun '06

Hi, I can't use ser0.9.7 because I need presence. I already have ser v10 installed and running. How can I add rtpproxy to it ? Thanks, ilker ________________________________ From: ram [mailto:talk2ram@gmail.com] Sent: Friday, June 16, 2006 5:01 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Rtpproxy howto Hi just download from onsip site ser0.9.7 untar that, and make rtpproxy from that and run rtpproxy its very simple ram On 6/16/06, İlker Aktuna (Koç. net ) <ilkera(a)koc.net> wrote: Hi, I will try to install RTP proxy for my internal users. Is there a simple RTP proxy howto/handbook for this process ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/020C0F07/04004D09/01020F4E/038…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

[Serusers] Rtpproxy howto
by İlker Aktuna $Koç.net$ 16 Jun '06

16 Jun '06

Hi, I will try to install RTP proxy for my internal users. Is there a simple RTP proxy howto/handbook for this process ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/0A090F07/05044F00/09070C4E/021…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

3 2

[Serusers] replicate problems with openser.
by Shenen Shenen 16 Jun '06

16 Jun '06

This is my openser.cfg; I'm using it to replicate the SIP registration from one asterisk to another one (to have always the same registration status of SIP in all 2 asterisk,in case of falls, every 2 ,have the same configuration and softphones run without registration problem...(this is my purpose....) . # demo script showing how to set-up usrloc replication # SCRIPT PER COPIARE LO STATO DELLE REGISTRAZIONI DEI SIP DAL .12 all' .11 # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) #listen= 192.16.25.11 port=5060 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" #loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" #loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; #if (len_gt( max_len )) { if ( msg:len > max_len ){ sl_send_reply("513", "Message too big"); return; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # verify credentials if (!www_authorize("", "subscriber")) { www_challenge("", "0"); return; }; # if ok, update contacts and ... save("location"); # ... if this REGISTER is not a replica from our # peer server, replicate to the peer server if (!src_ip== 192.16.25.12) { t_replicate("192.16.25.12","5060"); }; return; }; # do whatever else appropriate for your domain log("non-REGISTER\n"); }; } When I run openser happens this: [root@asterisk11 ~]# openser 0(3294) WARNING: fix_socket_list: could not rev. resolve 192.16.25.11 0(3294) WARNING: fix_socket_list: could not rev. resolve 192.16.25.11 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.16.25.11 [ 192.16.25.11]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.16.25.11 [ 192.16.25.11]:5060 Aliases: tcp: asterisk11.local:5060 udp: asterisk11.local :5060 WARNING: no fork mode and more than one listen address found(will use only the the first one) stateless - initializing 0(0) Maxfwd module- initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) WARNING: using only the first listen address (no fork) and then the windows doesn't respond.....is all stop....where is the problem? 10000000 thanks!

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[Serusers] Again softphone not connected using port number other than 5060
by sumit dengri 16 Jun '06

16 Jun '06

sir Sorry to disturb you once again but my softphone (X lite) is still not connected to ser. I also checked its log and it is sending requests to port 5065. so PLZ tell all the possiblities and their corrections as well. so that I wont disturb you again. regards. __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

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RE: [Serusers] Presentity & Rls databases
by İlker Aktuna $Koç.net$ 16 Jun '06

16 Jun '06

Thank you :) -----Original Message----- From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Sent: Friday, June 16, 2006 10:31 AM To: İlker Aktuna (Koç.net) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Presentity & Rls databases See module parameters in presence handbook. Vaclav On Fri, Jun 16, 2006 at 10:27:40AM +0300, ?lker Aktuna (Koç.net) wrote: > Hi, > > Anyone who has information about this ? > > thanks > ilker > > ________________________________ > > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ?lker Aktuna > (Koç.net) > Sent: Thursday, June 15, 2006 9:33 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Presentity & Rls databases > > > > Hi, > > > In my Ser installation with mysql db, I see some tables for presentity > and RLS such as; > | presentity | > | presentity_contact | > | presentity_notes | > | presentity_persons | > | rls_subscription | > | rls_vs | > | rls_vs_names | > > But they are empty tables and seems that they are not used anyhow. > How can I make use of these tables ? > > I wonder how to save presence info on db. > > Thanks for any information. > > Regards, > ilker > <http://387555.sigclick.mailinfo.com/sigclick/020A0507/03014E02/0104014E/043…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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Re: [Serusers] replicate problems with openser.
by Andrey Kouprianov 16 Jun '06

16 Jun '06

1. Specify "listen=x.x.x.x" and add "children=N" directive as well (e.g. children=5) 2. Could not resolve blah-blah comes from the fact that you do not have a DNS reverse entry for this IP address. Try specifying the following directives. They might help. dns=no rev_dns=no Regards, Andrey. P.S. - This is just a warning, so dont worry too much. Nevertheless, seeing warnings is not that pleasant anyway. I understand your concern. ;-)

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[Users] Not Hearing voice..problem with audio
by raviprakash sunkara 16 Jun '06

16 Jun '06

Hi User, I just to say Hello ! to all.... I using openser ....When I making the call from one A place to other B place.. THe place B having the openser server When making the calls from A to B calls are invited but,,,Not hearing the voices. But Local network its working.. I'm used the Nathelper mdules and installed the rtpproxy. And below my openser file.. plz check it.. solve the issues. modparam("acc","log_level",1) modparam("acc","log_flag",1) modparam("acc","log_missed_flag",2) modparam("acc", "log_fmt", "cdfimorstup") #modparam("acc", "failed_transaction_flag",3) modparam("acc", "report_cancels", 1) modparam("acc","report_ack",0) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc","radius_flag",1) #modparam("acc","radius_level",1) #modparam("acc","radius_missed_flag",2) modparam("acc","service_type",15) #modparam("acc", "detect_direction", 1) modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") ##########33 modparam("nathelper","natping_interval",30) modparam("nathelper","ping_nated_only",1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #################33 modparam("rr", "enable_full_lr", 1) ############################################### route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("13")) { sl_send_reply("483","Too Many Hops........................!"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if(!method== "REGISTER") { record_route(); }; # loose-route processing if (loose_route()) { # add now 0n 9 setflag(1); acc_rad_request("200 ok "); t_relay(); exit; }; # account completed transactions via syslog #setflag(1); # setflag(2); if(uri==myself) { if(method=="REGISTER") { if(!radius_www_authorize("192.168.2.55")) { www_challenge("192.168.2.55","0"); exit; }; #consume_credentials(); save("location"); if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; if(method=="INVITE" && method=="ACK") { record_route(); acc_rad_request("200"); force_rtp_proxy(); #setflag(1); t_on_reply("1"); }; /* if (method=="BYE") { record_route(); }; */ if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { #log (1, "BYE or CANCEL\n"); #setflag(1); acc_rad_request("200 "); #acc_rad_request("Stop"); #unforce_rtp_proxy(); #setflag(1); }; if(!lookup("location")) { sl_send_reply("404","Woo......... NOt found"); }; }; if(!t_relay()) { sl_reply_error(); }; lookup("aliases"); #setflag(2); exit; } onreply_route[1] { if(status=~"[0-9][0-9][0-9]") { force_rtp_proxy(); }; } -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

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[Serusers] Still Not able to connect softphone on 5065
by sumit dengri 16 Jun '06

16 Jun '06

sir, I am using XLite softphone and also make necessary changes in its settings but its not working. Is it sufficient to change port number only in global parameters section of ser.cfg? plz help once again. regards. --------------------------------- Yahoo! India Answers: Share what you know. Learn something new Click here Send free SMS to your Friends on Mobile from your Yahoo! Messenger Download now

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[Serusers] Problem in changing port of SER
by sumit dengri 16 Jun '06

16 Jun '06

sir I have changed default port 5060 to 5065 in ser.cfg in Global configuration parameters section. Server started beautifully. But I am not able to connect soft phones to the server. Softphone showed 'Login Failed' So PLZ tell me if I have to make any other changes in configuration file or any other file. regards --------------------------------- Yahoo! India Answers: Share what you know. Learn something new Click here Send free SMS to your Friends on Mobile from your Yahoo! Messenger Download now

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[Serusers] openser IP listen: but only itself or all IPs...
by Shenen Shenen 16 Jun '06

16 Jun '06

Hi I've a question; I've added some SIP softphones in openser and they're ok in the database;(I've seen that they're registered) I've this softphone(x-lite and phoner) in a Xp pc with IP 192.16.25.10, so I've registered in my openser on an asterisk at home box in the following way; openserctl add 650 1234 650(a)192.16.25.10 openserctl add 651 1234 651(a)192.16.25.10 My openser in on an asterisk at home with 192.16.25.12 like IP; If I run the openser command, I've this; Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.16.25.12 [192.16.25.12]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.16.25.12 [192.168.251.12]:5060 Aliases: tcp: localhost:5060 udp: localhost:5060 Is it right or I've to put in openser.cfg the following lines? listen=192.16.25.12 listen=192.16.25.10 but If I put this 2 lines, I've the following when I run openser; Listening on udp: 192.168.251.200 [192.168.251.200]:5060 udp: 192.168.251.12 [192.168.251.12]:5060 udp: 192.168.251.11 [192.168.251.11]:5060 tcp: 192.168.251.200 [192.168.251.200]:5060 tcp: 192.168.251.12 [192.168.251.12]:5060 tcp: 192.168.251.11 [192.168.251.11]:5060 Aliases: I think isn't right...... My question is: In the first case,without editing the openser.cfg, openser listen also on the Xp pc?Or only on 192.16.25.12 and 127.0.0.1? How can I make that openser can listen on all the IP?(I want that it listens on my Xp and then on others...)

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RE: [Serusers] Presentity & Rls databases
by İlker Aktuna $Koç.net$ 16 Jun '06

16 Jun '06

Hi, Anyone who has information about this ? thanks ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Thursday, June 15, 2006 9:33 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Presentity & Rls databases Hi, In my Ser installation with mysql db, I see some tables for presentity and RLS such as; | presentity | | presentity_contact | | presentity_notes | | presentity_persons | | rls_subscription | | rls_vs | | rls_vs_names | But they are empty tables and seems that they are not used anyhow. How can I make use of these tables ? I wonder how to save presence info on db. Thanks for any information. Regards, ilker _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________ <http://387555.sigclick.mailinfo.com/sigclick/01090500/0C044F07/02040749/0B6…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Serusers] problem in adding exec module in ser.cfg
by Shashwat Gupta 16 Jun '06

16 Jun '06

I am trying to mail shashwat user for any message that passes through my ser and destined for shashwat. I found that it could be done through existing exec module... so I added following lines to ser.cfg loadmodule "/usr/local/lib/ser/modules/exec.so" along with other module of ser.cfg in route {} I added following lines if (uri=~"^sip:shashwat@") { exec_msg("echo 'body: call arrived'|mail -s 'call for you' shashwat"); } but when I restart server I found that even then shashwat is not getting any mail from ser server.... I am not able to find out the possible reason ..any idea how can make I makei it work.. It will be great help for me thanks in advance -- regards shashwat gupta

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[Serusers] which version of eyebeam to work with presence server
by Yan Tian 16 Jun '06

16 Jun '06

I am wondering which version of eyebeam client works with the new presence server (ie, resource list support + xcap ). The one I have (v1.1.3014w) doesn't use the correct Accpet header for SUBSCRIBE to a resource list and hence is rejected by SER. thanks, - ming

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[Serusers] SER+PostgreSQL module for Accouting
by Emerson Oatis 16 Jun '06

16 Jun '06

hi list! we tried to log accounting of ser on postgres, but it seems that logging was not successful. here are the simulations we did : - we convert all the schema of mysql to postgresql databse (ser schema), just minor revision on datatypes since we don't find any posgresql.sql for ser on the net - followed this steps http://www.voip-info.org/wiki/view/SER+module+postgres - recompiled ser to support postgresql database - tried to configure/run ser.cfg to log accounting but no luck - we did some search and tried it first on mysql database - edit the src of acc module and edit the Makefile to support SQL logging, recompile the acc.so - copy the new acc.so to /path/of/ser/modules/ - add db_flag on ser.cfg - now acc mysql was logging - tried to replace the connector from mysql to postgresql on ser.cfg with the same config - ngrep port 5432 on localhost but it seems no request came from ser - ser was running ok and we see logs on /var/log/messages that acc was working * do we missed something? or probably tweak some configuration? * below are snippet of my ser.cfg thanks and regards emer ------------------------------------------------------------ fifo="/tmp/ser_fifo" fifo_db_url="postgres://ser:heslo-757@localhost:5432/ser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/postgres.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db","db_url", "postgres://ser:heslo-757@localhost:5432/ser") # -- acc params -- modparam("acc", "log_level", 1) # that is the flag for which we will account -- don't forget to # set the same one :-) modparam("acc", "log_flag", 1 ) modparam("acc", "db_flag", 1) /* ********* RR ********************************** */ /* grant Route routing if route headers present */ if (loose_route()) { # mark the BYEs -- they are requests within a dialog if(method=="BYE") setflag(1); t_relay(); break; }; /* record-route INVITEs -- all subsequent requests must visit us */ if (method=="INVITE") { setflag(1); record_route(); }; ---------------------------------------------------------------------------------------------------- # postges/ser process 10199 ? S 0:00 ser -f pstn.ser02 10201 ? S 0:00 ser -f pstn.ser02 10205 ? S 0:00 postgres: ser ser 127.0.0.1(56629) idle 10203 ? S 0:00 ser -f pstn.ser02 10209 ? S 0:00 postgres: ser ser 127.0.0.1(56630) idle 10211 ? S 0:00 postgres: ser ser 127.0.0.1(56631) idle 10206 ? S 0:00 ser -f pstn.ser02

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[Serusers] Confusion about ser.cfg file
by sumit dengri 16 Jun '06

16 Jun '06

sir >I have put ser-0.9.6_src.tar.gz in /usr/local and compiled it from here. >Now there are two ser.cfg files created. Are two configuration files created for ser? >One in /usr/local/etc/ser and another one in /usr/local/ser-0.9.6/etc >The problem is when I make changes in /usr/local/etc/ser/ser.cfg according to the docs it shows an error in loading mysql.so module. >And similar changes made in /usr/local/ser-0.9.6/ser.cfg showed no error and server is running very well. >Could anyone tell me the reasons behind this. thanks. __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

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[Serusers] LCR Compilation error
by Ryan Pagquil 16 Jun '06

16 Jun '06

Hi, I'm compiling LCR module from CVS head with ser-0.9.4 and I got these errors: In file included from fifo.c:34: lcr_mod.h:50: parse error before "uri_type" lcr_mod.h:50: warning: no semicolon at end of struct or union lcr_mod.h:54: parse error before '}' token make[1]: *** [fifo.o] Error 1 make[1]: Leaving directory `/usr/local/src/ser-0.9.4/modules/lcr' Please help. Thanks, Ryan

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RE: [Devel] RE: [Users] OpenSER - one year old
by Douglas Garstang 15 Jun '06

15 Jun '06

> -----Original Message----- > From: Tavis P [mailto:tavis.lists@galaxytelecom.net] > Sent: Thursday, June 15, 2006 3:33 PM > To: devel(a)openser.org > Cc: users(a)openser.org > Subject: Re: [Devel] RE: [Users] OpenSER - one year old > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: Tavis P [mailto:tavis.lists@galaxytelecom.net] > >> Sent: Thursday, June 15, 2006 3:04 PM > >> Cc: devel(a)openser.org; users(a)openser.org > >> Subject: Re: [Devel] RE: [Users] OpenSER - one year old > >> > >> > >> Juha Heinanen wrote: > >> > >>> Douglas Garstang writes: > >>> > >>> > My comments are constructive. I've constructively > stated several > >>> > times that the documentation is very bad. Each time I've > >>> > >> made this > >> > >>> > constructive comment, I've been attacked. In my previous > >>> > >> post, I made > >> > >>> > another constructive comment. I said that maybe the reason the > >>> > documentation is so bad is that the project isn't that old. > >>> > >>> rather that complaining, learn to read the relevant pieces > >>> > >> of code and > >> > >>> write better documentation. > >>> > >>> -- juha > >>> > >>> _______________________________________________ > >>> Devel mailing list > >>> Devel(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/devel > >>> > >>> > >>> > >>> > >> I find it interesting that you would characterize your > statements as > >> "constructive" and the responses you've received as "attacks" > >> > >> I believe this may be one of the central issues > >> > > > > Why do you consider my opinion that the documentation is > bad, and that it should be maintained by the developers interesting? > > > > > > > Its not your opinion on the documentation that i find interesting, its > how you characterize your dislike for the documentation (and repeated > expression of it) > > > In my opinion, if you are able to use an application you are able to > document how you use it. > I am not a C developer however i have contributed to the > documentation, > and i believe it is with this type of action that Open Source > communities become strong (contribution). > Having someone constantly express what they don't like about > the project > yet do little else to effect change in it serves no positive or > productive purpose > > OpenSER is an excellent application and i am proud to be a part of its > community. A community that responds to impatience and contempt with > patience and wisdom > Please help us keep this community healthy because in the end it > benefits us all (in more ways that i can count) > > Demanding that others solve perceived issues is generally not a good > approach (unless you are dealing with a commercial entity and has a > professional responsibility to you) Sorry, but I disagree with you. I'm fed up with relying on documentation that's badly written or wrong, because the person who wrote it made assumptions about what they thought they knew... because they in turn relied upon someone else who had made the same mistakes. The reason the chain of events got started in the first place was because people played around, experimated, tinkered, etc, and came to incorrect conclusions.

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RE: [Devel] RE: [Users] OpenSER - one year old
by Douglas Garstang 15 Jun '06

15 Jun '06

> -----Original Message----- > From: Tavis P [mailto:tavis.lists@galaxytelecom.net] > Sent: Thursday, June 15, 2006 3:04 PM > Cc: devel(a)openser.org; users(a)openser.org > Subject: Re: [Devel] RE: [Users] OpenSER - one year old > > > Juha Heinanen wrote: > > Douglas Garstang writes: > > > > > My comments are constructive. I've constructively stated several > > > times that the documentation is very bad. Each time I've > made this > > > constructive comment, I've been attacked. In my previous > post, I made > > > another constructive comment. I said that maybe the reason the > > > documentation is so bad is that the project isn't that old. > > > > rather that complaining, learn to read the relevant pieces > of code and > > write better documentation. > > > > -- juha > > > > _______________________________________________ > > Devel mailing list > > Devel(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/devel > > > > > > > I find it interesting that you would characterize your statements as > "constructive" and the responses you've received as "attacks" > > I believe this may be one of the central issues Why do you consider my opinion that the documentation is bad, and that it should be maintained by the developers interesting?

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RE: [Users] OpenSER - one year old
by Douglas Garstang 15 Jun '06

15 Jun '06

> -----Original Message----- > From: Juha Heinanen [mailto:jh@tutpro.com] > Sent: Thursday, June 15, 2006 12:59 PM > To: Douglas Garstang > Cc: ram; users(a)openser.org; devel(a)openser.org > Subject: RE: [Users] OpenSER - one year old > > > Douglas Garstang writes: > > > It's only a year old? Oh... I thought it was older. That might help > > to explain the really bad documention then. > > if you don't have anythign constructive to say about openser > open source > project, could you please switch to other sip proxy software and stop > insulting hard working people on this list. My comments are constructive. I've constructively stated several times that the documentation is very bad. Each time I've made this constructive comment, I've been attacked. In my previous post, I made another constructive comment. I said that maybe the reason the documentation is so bad is that the project isn't that old. Doug.

4 3

[Serusers] Voicemail
by ram 15 Jun '06

15 Jun '06

Hi iam working with Ser + ASterisk how can i send the call to voice mail after X rings ( any way of the user is not there iam already sending to voice mail) example : if some one calls registered user if the user is not respond for 6 rings its go to Voice mail if the user not there it should go directly to voice mail help will be appriciated ram

4 3

[Serusers] asterisk+cdrtool
by hgaillac-sip＠yahoo.fr 15 Jun '06

15 Jun '06

hello, i try to install the latest cdrtool but i get this messsage: #!/usr/bin/php4 Fatal error: Cannot redeclare class db_sql in /var/www/CDRTool/phplib/db_mysql.inc on line 12 Thanks for help Harry __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail

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RE: [Users] OpenSER - one year old
by Douglas Garstang 15 Jun '06

15 Jun '06

It's only a year old? Oh... I thought it was older. That might help to explain the really bad documention then. -----Original Message--- -- From: ram [mailto:talk2ram@gmail.com] Sent: Wednesday, June 14, 2006 9:14 AM To: users(a)openser.org; devel(a)openser.org Subject: Re: [Users] OpenSER - one year old Hi happy to hear that I wish all who are involve to make this project so success and helpfull happy <http://openser.org/> OpenSER project celebrates one year ram

3 2

[Users] Voicemail
by ram 15 Jun '06

15 Jun '06

Hi iam working with OpenSer + ASterisk how can i send the call to voice mail after X rings ( any way of the user is not there iam already sending to voice mail) example : if some one calls registered user if the user is not respond for 6 rings its go to Voice mail if the user not there it should go directly to voice mail help will be appriciated ram

1 0

[Users] Setting up voicemail
by Edgar Barbosa 15 Jun '06

15 Jun '06

Hi all, I'm trying to setup an openser using an asterisk as a pstn gw and voicemail server. Everything seems to be working fine except when a call is redirected to voicemail by timeout (using failure_route). In this situation, asterisk is asking for proxy auth to openser that redirects the request to the calling ua. But when the ua responds to openser, this latest sends the invitation to the destination ua and not to asterisk, as he should. I can't figure out a way of accomplishing this... any idea? The sip trace diagram and the full openser.cfg are in attachment. The wrong invite is represented as "F19 INVITE" in the diagram... Thanks, Edgar

2 1

[Users] setdsturi
by Simon Morvan 15 Jun '06

15 Jun '06

Hello there, Why does the following configuration statement : setdsturi("sip:hostname:5061"); is not accepted and results in the following in my logs : set_dst_uri: Invalid parameter value It's probably straightforward but I don't understand... Thanks in advance. -- Simon Morvan.

2 1

[Users] renaming locations table
by Fatih Cerit 15 Jun '06

15 Jun '06

Hi all I want to rename locations table for entegrating to my database structure. is there any way??

2 3

[Serusers] SIP registration with openser
by Shenen Shenen 15 Jun '06

15 Jun '06

I've set the SIP_DOMAIN environment variable, (the IP of the pc where I'm running the openser and then the same command but with the IP where I've a sopftphone on a XP pc) and then I've run this: openserctl add 650 1234 650(a)192.16.25.12 (650 is a softphone registered on 192.16.25.10...a XP pc)(I've made also with 650@192.16.25.10..but doesn't change...) Mysql password: (empty I think) MySql password for user 'openser@localhost': (empty) Enter password: (root password ) ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES) introducing the new user to the database failed Where is the problem? Which are the right passwords?....I'm using openser on an asterisk at home box..

2 1

[Serusers] About Change in Listening port of SER
by sumit dengri 15 Jun '06

15 Jun '06

sir Can we change default listening port (5060) of SER to any other port by changing the port number in ser.cfg file. Or by any other means? regards Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

2 1

[Serusers] Database in openser and SIP registration
by Shenen Shenen 15 Jun '06

15 Jun '06

Hi!I've run the command to create the database in this way; openser_mysql.sh create I've entered a blank password for root, I entered ser as the domain: I got a creating database ... message, then got another "Enter password:" request, entered blank and then returned to the command line Domain (realm) for the default user 'admin' : ser I've put; If I run in mysql with mysql -p and entered in it,I run connect,I've this; mysql> connect Connection id: 115 Current database: *** NONE *** If I use mysql>connect ser ERROR 1049 (42000): Unknown database 'ser' Where is the problem? I use the following command to register my openser; openserctl add 650 1234 650(a)192.168.251.12 [mailto:650@192.168.251.12<650(a)192.168.251.12> ] /usr/local/sbin/openserctl 1.1 - $Revision: 1.6.2.2 $ parameter usage: * subscribers * add <username> <password> <email> .. add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) mail <username> .................... send an email to a user alias show [<alias>] ............... show aliases alias rm <alias> ................... remove an alias alias add <alias> <uri> ............ add an aliases rpid add <username> <rpid> ......... add rpid for a user (*) rpid rm <username> ................. set rpid to NULL for a user (*) rpid show <username> ............... show rpid of a user alias_db show <alias> .............. show alias details ............. etc etc.... Where is the problem?

2 1

[Serusers] DB Connectivity Error in SerWeb
by sumit dengri 15 Jun '06

15 Jun '06

sir When I try to login in serweb it showed the following error message: DB Error: Connect Failed Bad User Name/Password PLZ tell me the possible reasons for this error message. regards. Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

2 1

[Serusers] sip to h323 gateway ...
by Cesc 15 Jun '06

15 Jun '06

Hi, I know, a bit off topic ... but i am sure people here have the expertise, so :) I am facing a request to bridge a sip network and an h323 network. I would like to operate the sip with ser as the proxy and some gatekeeper on the h323 side (not required though). The question is ... what should i use to bridge the two networks? - asterisk - yate - vovida -> vocal Any others? comments? Ideally, the bridge would be only signalling-wise (rtp to be direct end-to-end). But, if someone had bad experience with this and would recommend to use a B2BUA approach, please, tell me. I don't know if it makes a difference, but most of the calls would go from the H323 side to the SIP side ... but i don't really want to restrict SIP->H323. Cesc

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[Serusers] ser.cfg for VRRP protocol => using t_replicate for 2 proxies
by Shenen Shenen 15 Jun '06

15 Jun '06

Hi!I've 2 asterisk at home and I use openser becouse I want use more proxies for my softphones;my problem is this: I'm using vrrp protocol; 192.168.25.1 is the master (asterisk1) 192.168.25.2 is the slave (asterisk2) 192.168.25.200 is the virtual IP. then I have my softphones on a Xp pc with IP 192.168.25.10 I set my softphones with 192.168.25.200 and they run, only that, the problem with asterisk is that softphones and IP phones(some like wi-fi) can register only one proxy, and asterisk can't make a option for more proxies than one;so the proxy is the 192.168.25.200, but at real it is 192.168.25.1..andsoftphones or wi-fi are registered only in the first asterisk with the 192.168.25.1..if the master falls down the slave hasn't the authentication SIP registration...only the master..So I must use openser or ser to have 2 = box my problem is resolved only if I use the redirect option to have 2 = asterisk box so the database can be opened and read from one asterisk to the other;if the master falls down the slave has all the configuration and the SIP registration, so it can work without problems. A) I've set this file on asterisk 192.168.25.1 the master # # $Id: replicate.cfg,v 1.2 2005/07/25 16:27:33 miconda Exp $ # # demo script showing how to set-up usrloc replication # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) # ------------------ module loading ---------------------------------- loadmodule "modules/mysql/mysql.so" loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/usrloc/usrloc.so" loadmodule "modules/registrar/registrar.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); return; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # verify credentials if (!www_authorize("", "subscriber")) { www_challenge("", "0"); return; }; # if ok, update contacts and ... save("location"); # ... if this REGISTER is not a replica from our # peer server, replicate to the peer server if (!src_ip== 192.168.25.2) { t_replicate("192.168.25.2", "5060"); }; return; }; # do whatever else appropriate for your domain log("non-REGISTER\n"); }; } B) I've set this file on asterisk 192.168.25.2 the slave # # $Id: replicate.cfg,v 1.2 2005/07/25 16:27:33 miconda Exp $ # # demo script showing how to set-up usrloc replication # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) # ------------------ module loading ---------------------------------- loadmodule "modules/mysql/mysql.so" loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/usrloc/usrloc.so" loadmodule "modules/registrar/registrar.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); return; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # verify credentials if (!www_authorize("", "subscriber")) { www_challenge("", "0"); return; }; # if ok, update contacts and ... save("location"); # ... if this REGISTER is not a replica from our # peer server, replicate to the peer server if (!src_ip== 192.168.25.1) { t_replicate("192.168.25.1", "5060"); }; return; }; # do whatever else appropriate for your domain log("non-REGISTER\n"); }; } Must I call this file ser.cfg?Right?Must I set the on the master listen=192.168.25.1 #on the master port=5060 on the slave listen=192.168.25.2 #on the slave port=5060 Then I use on all the 2 linux; #chkconfig -add ser #chkconfig -add mediaproxy And ser and mediaproxy works like deamon every time I start-up the 2 linux. Is it right to add this 2 sip number in openser or ser in this way?(they're registered in asterisk and 650 is a number from a X-lite, 651 from Phoner) serctl add 650 1234 650(a)192.168.25.200 [mailto: 650(a)192.168.25.200] serctl add 651 1234 651(a)192.168.25.200 [mailto: 651(a)192.168.25.200] Is it right this configuration?If the master falls down the slave has all the Sip registration authentificated and run without any problem? 100000 thanks!

2 1

[Serusers] serctl errors;adding new users
by Shenen Shenen 15 Jun '06

15 Jun '06

Hi!I've a lot of softphones to one pc with IP 192.1.168.10 (XP pc) and ser in a asterik_at_home box with IP 192.1.168.12, if I make : serctl add 650(a)192.1.168.10 1234 mymail(a)mymail.com Mysql password: Password?is helso or pass0rd ...no one of this,it returns me always this error ERROR 1045 (28000): Access denied for user 'serro'@'localhost' (using password: YES) ERROR 1045 (28000): Access denied for user 'serro'@'localhost' (using password: YES) ERROR 1045 (28000): Access denied for user 'ser'@'localhost' (using password: YES) introducing the new user to the database failed which is the password? With serctl ps I'vent the IP of the XP... I must add it in ser.cfg?And like?If I have other softphones or IP phones on other machine,must I set in ser.cfg the IP of every on pc or IP phone? And how?

1 0

[Serusers] alias table
by olivier.taylor 15 Jun '06

15 Jun '06

Hi all, Is there a way to reload alias table without restarting Ser? When I add an alias, I need to restart Ser, otherwise the alias will not be found. Via serct? Olivier

2 1

[Serusers] SER with nathelper & rtprpoxy - how to disable rtpproxy when two phones are behind the SAME NAT
by Arek Bekiersz 15 Jun '06

15 Jun '06

Hi, Not sure if such issue was discussed before. Is this possible to detect that two phones are behind the same NAT, to avoid forcing rtpproxy in SER? This way we let the phones to send media directly to each other in private network behind NAT. Imagine two SIP phones connected to WAN using router with NAT. Both phones are in the same private network. Both phones have accounts on SER proxy, that in turn runs nathelper and rtpproxy. Phone_1 <--> NAT <--> SER,nathelper,rtpproxy Phone_2 <--> Normally when we make call between Phone_1 and Phone_2 using SER, RTP media will flow from Phone_1 to rtpproxy and back to Phone_2. Well, this is basically what we expect from rtpproxy & nathelper :-) Now somehow we could be able to detect in SER that both phones are behind the SAME NAT. When such situation is confirmed, we do not use nathelper and rtpproxy. As a result the phones send RTP media directly between them in private network, not using (and probably saturating) the poor internet link. I was thinking about: 1) Standard detection of NAT-ed situation, and then 2) comparing source IP of INVITE from Phone_1 with Contact IP stored for Phone_2 in "location" table. If they match, both phones MAY be in the same private network and we do not use nathelper functions. Of course it would work only if there is no other NAT before the primary NAT (i.e. we do not have the double NAT situation or similar) -- Regards, Arek Bekiersz

2 1

[Serusers] Nedd Help Badly
by Nhadie 15 Jun '06

15 Jun '06

Hi All, I'm having problem that the rtpproxy is using 54% of my CPU. Unfortunately i don't have enough money to buy a newer server. Can I make it such a way that I will force only rtpproxy when the call is from a private top another private, but when a private goes to a public IP (like my PSTN gateway) I won't use rtpproxy. another prob of mine is that some phones don't have STUN configured on it, but I have a STUN server. The big clients Ihave pu STUN already, but still they're going to the rtpproxy, I tried this config if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@202.202.202.203")) { if (is_user_in("From", "world-pstn")) { if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the Worldn");}; if ((is_user_in("From", "onstun"))) { # Strip first to digits (00) strip(2); prefix("25050#"); rewritehostport("202.202.202.202:5060"); consume_credentials(); t_relay_to_udp("202.202.202.202", "5060"); break; } else { if (isflagset(6)) route(1); strip(2); prefix("25050#"); rewritehostport("202.202.202.202:5060"); #rewritehostport("202.202.202.202:5060"); consume_credentials(); t_relay_to_udp("202.202.202.202", "5060"); #t_relay_to_udp("202.202.202.202", "5060"); break; }; } else { if (isflagset(8)) {xlog("L_INFO", "Your not allowed to call PSTNn");}; sl_send_reply("404", "DENIED: USER NOT IN WORLD-PSTN"); break; }; which would detect if an extension is a member of the onstun it will not use rtpproxy, but the caller (using IP phone) cannot hear the called party but the call is connecting, hope anyone can help me, badly need it. Thank You Regards, Nhadie ________________________________________________ Message sent using UebiMiau 2.7

2 1

[Serusers] Multiple SERs sharing same database
by Nick Hoffman 15 Jun '06

15 Jun '06

If I have multiple SER servers, should they each have their own database, or can they all share the same database? They're all going to be for the same purpose and will have the exact same subscriber, alias, etc information. Cheers, -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

2 1

[Serusers] Presentity & Rls databases
by İlker Aktuna $Koç.net$ 15 Jun '06

15 Jun '06

Hi, In my Ser installation with mysql db, I see some tables for presentity and RLS such as; | presentity | | presentity_contact | | presentity_notes | | presentity_persons | | rls_subscription | | rls_vs | | rls_vs_names | But they are empty tables and seems that they are not used anyhow. How can I make use of these tables ? I wonder how to save presence info on db. Thanks for any information. Regards, ilker <http://387555.sigclick.mailinfo.com/sigclick/020B0600/040D4E03/01070149/032…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

RE: [Serusers] logging missed calls and informing when online
by İlker Aktuna $Koç.net$ 15 Jun '06

15 Jun '06

Is there a voicemail mechanism ready for Ser ? Thanks, ilker -----Original Message----- From: Edson [mailto:4lists@gmail.com] Sent: Wednesday, June 14, 2006 11:41 PM To: İlker Aktuna (Koç.net); serusers(a)lists.iptel.org Subject: RE: [Serusers] logging missed calls and informing when online How about some sort of Voicemail? You can think on a predefined message to only register the caller number, or the hole Voicemail mechanism for voice messages... than a predefined message (or e-mail) to tell the 'off-line' user that someone has called. Edson. ________________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ilker Aktuna (Koç.net) Sent: quarta-feira, 14 de junho de 2006 13:39 To: serusers(a)lists.iptel.org Subject: [Serusers] logging missed calls and informing when online Hi, Is there a way to log missed calls for offline users and then inform the user when the user is online ? For offline messages it can be done with msilo module, but what about offline missed calls ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/060E0603/040D4E00/0505084A/031…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

[Serusers] LCR Download
by Ryan Pagquil 15 Jun '06

15 Jun '06

Hi, How can I download the latest LCR module from CVS? I'm not familiar in using CVS. Thanks, Ryan

1 0

[Users] Building on Solaris10?
by Bret Schuhmacher 15 Jun '06

15 Jun '06

Hi all, I'm having problems loading the mysql module. Seems to be a linker issue - the debug says function floor isn't exported. Anything different required to build or load this module under Solaris10? Do I have to force a 64 bit link or something? The Makefile has a bunch of 64 bit stuff in it: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.1"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.3"' -D__CPU_sparc64 -D__OS_solaris ..... All the other modules load just fine... Here's my openser stdout: 0(12978) loading module /usr/local/lib/openser/modules/mysql.so 0(12978) ERROR: load_module: could not open module </usr/local/lib/openser/modules/mysql.so>: ld.so.1: openser: fatal: relocation error: file /usr/local/lib/openser/modules/mysql.so: symbol floor: referenced symbol not found ldd shows this, but it's not very helpful: bash-3.00# ldd /usr/local/lib/openser/modules/mysql.so libz.so => /usr/lib/libz.so libc.so.1 => /lib/libc.so.1 libm.so.2 => /lib/libm.so.2 /platform/SUNW,UltraAX-i2/lib/libc_psr.so.1 I've played with LDFLAGS, LD_LIBRARY_PATH, etc. Any tips or pointers in the right direction would be welcomed if this rings a bell with anyone. Thanks in advance! Bret

3 2

[Serusers] logging missed calls and informing when online
by İlker Aktuna $Koç.net$ 14 Jun '06

14 Jun '06

Hi, Is there a way to log missed calls for offline users and then inform the user when the user is online ? For offline messages it can be done with msilo module, but what about offline missed calls ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/02090F04/04004D00/01020D4D/031…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

[Users] Is this configuration possible?
by Kenny Chua 14 Jun '06

14 Jun '06

I'm trying to build Openser with these requirements. Sorry, for the newbie questions. I have 10 users and a PSTN gateway. Let's say each users makes a call for 10 minutes. Is it possible to just show 100 minutes being charged on the PSTN gateway instead of 10 seperate users? If that is possible. Would I need the B2BUA in asterisk to accomplish this? I read Asterisk at large. I was wondering besides using Asterisk for voicemail, what other functions are used with integrating asterisk and openser? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] TLS problem
by Tao 14 Jun '06

14 Jun '06

hi guys, happy OpenSER's 1st Anniversary and happy world cup! I configure the TLS on OpenSER-1.0.1 release, but it doesn't work well. i searched on the web and found the discussion (attach it below) which posted monthes ago, my problem is very similar to it. but i can't find any conclusion about this discussion. Does anyone has resolved the similar problem, can you share the experiences? thanks in advance. my openssl's version is 0.9.8a when used snom360 to connect openser via tls, it blocked and freezed after receive ServerHelloDone. windows messenger 5.1 can go further, but still popup the "There was a problem verifying the certificate..." msg. and openser print the error are SSL_ERROR_WANT_READ and SSL_ERROR_SYSCALL... my certificate should be right, i have checked and regenerated it heaps of times... ---------------------------------------- [prev in list] [next in list] [prev in thread] [next in thread] List: voipsec Subject: Re: [VOIPSEC] Snom Softphone with TLS and Openser From: dennis <m8939605 () yahoo ! com ! tw> Date: 2006-02-24 13:44:01 Message-ID: 20060224134401.62975.qmail () web17506 ! mail ! tpe ! yahoo ! com [Download message RAW] Hi Martin, I folllow your method, but I still have somme problem. 1.After receive ClientHello, openser will be terminated. my openser is 1.0.0 1 1 0.0023 (0.0023) C>S Handshake ClientHello Version 3.1 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_NULL_MD5 TLS_RSA_WITH_NULL_SHA TLS_DH_anon_WITH_3DES_EDE_CBC_SHA TLS_DH_anon_WITH_RC4_128_MD5 TLS_RSA_WITH_DES_CBC_SHA TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_DH_anon_WITH_DES_CBC_SHA compression methods NULL 1 0.2734 (0.2710) S>C TCP FIN /////////////////////////////////// 2. Add the tls_ciphers_list="NULL-SHA:NULL-MD5", openser was ok, but snom soft phone was stuck immediately after starting and did not accept any input via the user interface. 1 1 0.0894 (0.0894) C>S Handshake ClientHello Version 3.1 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_NULL_MD5 TLS_RSA_WITH_NULL_SHA TLS_DH_anon_WITH_3DES_EDE_CBC_SHA TLS_DH_anon_WITH_RC4_128_MD5 TLS_RSA_WITH_DES_CBC_SHA TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_DH_anon_WITH_DES_CBC_SHA compression methods NULL 1 2 0.0913 (0.0018) S>C Handshake ServerHello Version 3.1 session_id[32]= 86 63 02 13 cd 51 12 d8 02 61 aa cc 66 63 84 d8 21 42 01 8e c1 d6 8e b0 c3 b6 d1 26 68 73 0d 02 cipherSuite TLS_RSA_WITH_NULL_MD5 compressionMethod NULL 1 3 0.0913 (0.0000) S>C Handshake Certificate 1 4 0.0913 (0.0000) S>C Handshake ServerHelloDone 1 131.0737 (130.9823) S>C TCP FIN When you re-executed the program, the ceritificate will be clean away. I thought that the soft phone lost it's certificate, so it hang on. Another root causer may be openssl (0.97f), I will try to upgrade or reinstall it. /////////////////////////////////////// In my environment, Windows Messenger always has some problems with Openser, when openser sent certificate, WM always pop up a error messange. 3 1 0.8193 (0.8193) C>S Handshake ClientHello Version 3.1 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_3DES_EDE_CBC_SHA TLS_RSA_WITH_DES_CBC_SHA TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_RSA_EXPORT_WITH_RC4_40_MD5 TLS_RSA_EXPORT_WITH_RC2_CBC_40_MD5 TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA TLS_DHE_DSS_WITH_DES_CBC_SHA TLS_DHE_DSS_EXPORT1024_WITH_DES_CBC_SHA compression methods NULL 3 2 0.8199 (0.0006) S>C Handshake ServerHello Version 3.1 session_id[32]= c3 b3 f1 16 de e4 76 d6 97 e3 ae ba 68 06 31 92 1a 5c 62 c7 f5 8c 7d 2c 2e 2b 87 47 32 a6 04 32 cipherSuite TLS_RSA_WITH_3DES_EDE_CBC_SHA compressionMethod NULL 3 3 0.8199 (0.0000) S>C Handshake Certificate 3 4 0.8199 (0.0000) S>C Handshake ServerHelloDone //////////////////////////////////// But after replaced key size from 2048 to 1024, there was improvement in Windows Messenger, although it still pop up the same error. 3 1 0.8193 (0.8193) C>S Handshake ClientHello Version 3.1 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_3DES_EDE_CBC_SHA TLS_RSA_WITH_DES_CBC_SHA TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_RSA_EXPORT_WITH_RC4_40_MD5 TLS_RSA_EXPORT_WITH_RC2_CBC_40_MD5 TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA TLS_DHE_DSS_WITH_DES_CBC_SHA TLS_DHE_DSS_EXPORT1024_WITH_DES_CBC_SHA compression methods NULL 3 2 0.8199 (0.0006) S>C Handshake ServerHello Version 3.1 session_id[32]= c3 b3 f1 16 de e4 76 d6 97 e3 ae ba 68 06 31 92 1a 5c 62 c7 f5 8c 7d 2c 2e 2b 87 47 32 a6 04 32 cipherSuite TLS_RSA_WITH_3DES_EDE_CBC_SHA compressionMethod NULL 3 3 0.8199 (0.0000) S>C Handshake Certificate 3 4 0.8199 (0.0000) S>C Handshake ServerHelloDone 3 5 0.8701 (0.0501) C>S Handshake ClientKeyExchange 3 6 0.8701 (0.0000) C>S ChangeCipherSpec 3 7 0.8701 (0.0000) C>S Handshake 3 8 0.8736 (0.0035) S>C ChangeCipherSpec 3 9 0.8738 (0.0001) S>C Handshake 3 1.6979 (0.8241) C>S TCP FIN 3 10 1.6985 (0.0006) S>C Alert 3 1.6986 (0.0000) S>C TCP FIN The Alert was not a standard TLS alert description, so I can't analyze it. The Alter messange is below: 15 03 01 00 18 fe ef bc 84 a3 c7 8c 8c a5 91 e7 da e1 7c ^^^^^^^^ (there are some problems.....) 06 ee 35 9d 32 21 ec ef 8c 79 --- Christian Stredicke <Christian.Stredicke(a)snom.de> ��G > Instead of using DNS SRV you can also use a > transport parameter in the > outbound proxy. E.g. > > server.example.at:5061;transport=tls > > Christian > > > -----Original Message----- > > From: Voipsec-bounces(a)voipsa.org > > [mailto:Voipsec-bounces@voipsa.org] On Behalf Of > Martin Petraschek > > Sent: Thursday, February 23, 2006 5:01 AM > > To: Voipsec(a)voipsa.org > > Subject: [VOIPSEC] Snom Softphone with TLS and > Openser > > > > Hi all, > > > > I just wanted to share the experiences I made when > trying to > > get the Snom 360 Softphone to work with TLS > support together > > with Openser. Maybe my findings can be of use for > other > > people having similar problems. > > > > The Snom Softphone is one of the few Softphones I > am aware of > > that support TLS as well as RTP encryption. > Unfortunately it > > is not Open Source, but the binary is freely > available at > > http://www.snom.com/download/snom360-5.3.exe > > > > When trying to use TLS, one might be disappointed > that the > > configuration menus do not offer any setting like > "enable > > TLS". This is because the Snom phone uses DNS SRV > queries in > > order to find out which connection method to use. > The first > > task is therefore to configure SRV records of the > DNS server. > > For bind, the following lines did the trick: > > > > example.at. IN NAPTR 10 50 "s" "SIPS+D2T" "" > _sips._tcp.example.at. > > example.at. IN NAPTR 20 50 "s" "SIP+D2U" "" > _sip._udp.example.at. > > example.at. IN NAPTR 30 50 "s" "SIP+D2T" "" > _sip._tcp.example.at. > > > > ; ----- SRV records ----- > > _sip._udp IN SRV 0 0 5060 > server.example.at. > > _sip._tcp IN SRV 0 0 5060 > server.example.at. > > _sips._tcp IN SRV 0 0 5061 > server.example.at. > > > > > > After that, the Snom phone tried to contact the > SIP server via TLS. > > However, the program was stuck immediately after > starting and > > did not accept any input via the user interface. I > inspected > > the network traffic it generated with the help of > the tool > > ssldump, which showed the following: > > > > server:/etc/openser/tools# ssldump -i eth0 port > 5061 New TCP > > connection #1: user.example.at(3695) <-> > server.example.at(5061) > > 1 1 0.0124 (0.0124) C>S Handshake > > ClientHello > > Version 3.1 > > cipher suites > > TLS_RSA_WITH_RC4_128_MD5 > > TLS_RSA_WITH_RC4_128_SHA > > TLS_RSA_WITH_NULL_MD5 > > TLS_RSA_WITH_NULL_SHA > > TLS_DH_anon_WITH_3DES_EDE_CBC_SHA > > TLS_DH_anon_WITH_RC4_128_MD5 > > TLS_RSA_WITH_DES_CBC_SHA > > TLS_RSA_EXPORT1024_WITH_RC4_56_SHA > > TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA > > TLS_DH_anon_WITH_DES_CBC_SHA > > compression methods > > NULL > > 1 2 0.0145 (0.0021) S>C Handshake > > ServerHello > > Version 3.1 > > session_id[32]= > > 5d a6 8d 61 58 ed c6 08 ae 76 d1 eb 24 > 82 6a c3 > > 2e 12 4c 29 17 7b 80 bf 1d 98 82 2c 67 > 53 ab f0 > > cipherSuite > TLS_RSA_EXPORT1024_WITH_RC4_56_SHA > > compressionMethod NULL > > 1 3 0.0146 (0.0000) S>C Handshake > > Certificate > > 1 4 0.0146 (0.0000) S>C Handshake > > CertificateRequest > > certificate_types > rsa_sign > > certificate_types > dss_sign > > ServerHelloDone > > 1 9.5153 (9.5006) C>S TCP RST > > > > > > I noticed that the chosen ciphersuite was 1024 bit > RSA. > > Checking the certificate file > > /etc/openser/tls/user/user-cert.pem, I found that > the > > certificate configured for openser is 2048 bit! To > overcome > > this problem, I changed the configuration files > ca.conf and > > user.conf as well as gen_rootCA.sh (just replaced > 2048 with > > 1024 at every occurence). > > After re-generating the certificates and restaring > openser, > > the TLS connection finally worked like a charm. > > > > Cheers, > > > > Martin > > > > _______________________________________________ > > Voipsec mailing list > > Voipsec(a)voipsa.org > > > http://voipsa.org/mailman/listinfo/voipsec_voipsa.org > > > > > > > > _______________________________________________ > Voipsec mailing list > Voipsec(a)voipsa.org > http://voipsa.org/mailman/listinfo/voipsec_voipsa.org >

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RE: [Users] OpenSER - one year old
by Ricardo Martinez 14 Jun '06

14 Jun '06

Happy Birthday! Thanks to all the people involved in this project, thanks for your support and dedication. We really appreciate all your efforts. Thanks again and happy birthday. Cheers!! Ricardo Javier Martinez Ogalde Ingeniero de Desarrollo VOISS NET S.A. Cisco Certified Network Associate ,CCNA (CSCO 10643101) * : (56 2) 240 81 96 * : (56 2) 245 74 95 * : rmartinez(a)redvoiss.net > -----Mensaje original----- > De: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Enviado el: Miércoles, 14 de Junio de 2006 10:31 > Para: users(a)openser.org; devel(a)openser.org > Asunto: [Users] OpenSER - one year old > > > Hello everybody, > > today OpenSER is one year old. In the hope that its first > year brought > many features needed by you, helped you in your business, > study or just > in private use, we will like to thank to all developers and community > members for their contributions, support, usage and testing > of OpenSER. > > The news and summary of this year of activity is posted at: > > http://openser.org/openser-news-20060614.php > > For the next year, we look forward to many new features, to make the > functionality fit in most of business cases, from residential > services > to carrier grade. We are welcoming ideas and contributions > as well as > patches and testing reports. > > Cheers, > Daniel > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] OpenSER - one year old
by Daniel-Constantin Mierla 14 Jun '06

14 Jun '06

Hello everybody, today OpenSER is one year old. In the hope that its first year brought many features needed by you, helped you in your business, study or just in private use, we will like to thank to all developers and community members for their contributions, support, usage and testing of OpenSER. The news and summary of this year of activity is posted at: http://openser.org/openser-news-20060614.php For the next year, we look forward to many new features, to make the functionality fit in most of business cases, from residential services to carrier grade. We are welcoming ideas and contributions as well as patches and testing reports. Cheers, Daniel

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[Users] called failed 408 Request timeout
by raviprakash sunkara 14 Jun '06

14 Jun '06

Hi Users, Today OPENSER's I st Anniversary , hi dedues please solve my issues In Local network openser with radius (AAA) is working fine. by using xten softphones But when i'm deploy in some other places with different network . Its successfully logined , but when making the calls its gives log as ' called failed : 408 request time out..." here is my openser,cfg below ........................... modparam("usrloc", "db_mode", 2) modparam("auth_db", "password_column", "password") modparam("auth_db", "calculate_ha1", yes) modparam("usrloc|acc|auth_db|group|msilo", "db_url", " mysql://openser:openserrw@localhost/openser") ########333333 modparam("acc","log_level",1) modparam("acc","log_flag",1) modparam("acc","log_missed_flag",2) modparam("acc", "log_fmt", "cdfimorstup") #modparam("acc", "failed_transaction_flag",3) modparam("acc", "report_cancels", 1) modparam("acc","report_ack",0) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc","radius_flag",1) #modparam("acc","radius_level",1) #modparam("acc","radius_missed_flag",2) modparam("acc","service_type",15) #modparam("acc", "detect_direction", 1) modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") ##########33 modparam("nathelper","natping_interval",30) modparam("nathelper","ping_nated_only",1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #################33 modparam("rr", "enable_full_lr", 1) ############################################### route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("13")) { sl_send_reply("483","Too Many Hops........................!"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if(!method== "REGISTER") { record_route(); }; # loose-route processing if (loose_route()) { # add now 0n 9 setflag(1); acc_rad_request("200 ok "); t_relay(); exit; }; # account completed transactions via syslog #setflag(1); # setflag(2); if(uri==myself) { if(method=="REGISTER") { if(!radius_www_authorize("192.168.2.55")) { www_challenge("192.168.2.55","0"); exit; }; #consume_credentials(); save("location"); if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; if(method=="INVITE" && method=="ACK") { record_route(); acc_rad_request("200"); force_rtp_proxy(); #setflag(1); t_on_reply("1"); }; /* if (method=="BYE") { record_route(); }; */ if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { #log (1, "BYE or CANCEL\n"); #setflag(1); acc_rad_request("200 "); #acc_rad_request("Stop"); #unforce_rtp_proxy(); #setflag(1); }; if(!lookup("location")) { sl_send_reply("404","Woo......... NOt found"); }; }; if(!t_relay()) { sl_reply_error(); }; lookup("aliases"); #setflag(2); exit; } onreply_route[1] { if(status=~"[0-9][0-9][0-9]") { force_rtp_proxy(); }; } ----------------------------------------------------------------------------------- please help me........ -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

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[Users] Re: Openser as front end to Asterisk
by Kenny Chua 14 Jun '06

14 Jun '06

I'm trying to use Asterisk's B2BUA with Openser. However I read somewhere that Asterisk can't handle as much concurrent calls as Openser can with SIP. So that's why I am trying to use both of them together. This is gonna be somewhat I'm trying to do: UA ----->Openser ---> Asterisk ------> PSTN Gateway or SIP All UA should be regisetered on Openser using mysql database. Question is, if I use this implementation, wouldn't all the call load be on Asterisk instead of Openser? Maybe I'm getting the wrong picture, as I'm fairly new to this. So I appreciate as much help as possible. I'm trying to do this for my senior design. Thank you. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Serusers] Content-Type: multipart/mixed; boundary=uniqueBoundary problems?
by Jiri Kuthan 14 Jun '06

14 Jun '06

Hi Greg, looks like you will have to start debugging :-) Normally, SER doesn't care of payload (you could do H.323 tunnelling in it if you wished ;-)) I can only imagine SER can complain about the message if it uses some SDP-mangling module like nathelper but even then i would not expect SER dropping silently. Let me know what you have found out in any case -- it looks like a perfectly legitimate SIP request. -jiri At 07:44 14/06/2006, Greg Fausak wrote: >Before I start debugging I thought I'd ask. >I have a sip message from a gateway (Cisco IOS) that >is being sliently dropped. Should ser be able to handle >a message that looks like this? I haven't done a multipart >message before. > >Thanks, >---greg > >INVITE sip:+19194724170@sn-sip-in.ca-sn1.cisco.com:5060 SIP/2.0 >Via: SIP/2.0/UDP 172.18.109.91:5060;branch=z9hG4bK4F0109C >Remote-Party-ID: <sip: >+19199915651(a)172.18.109.91>;party=calling;screen=yes;privacy=off >From: <sip:+19199915651@172.18.109.91>;tag=249E4980-FFC >To: <sip:+19194724170@sn-sip-in.ca-sn1.cisco.com> >Date: Tue, 13 Jun 2006 19:37:55 GMT >Call-ID: F241878C-FA4A11DA-81EFC5C8-2F1D7951(a)172.18.109.91 >Supported: 100rel,timer,resource-priority,replaces >Min-SE: 1800 >Cisco-Guid: 4064260884-4199158234-2157445126-1397050494 >User-Agent: Cisco-SIPGateway/IOS-12.x >Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, >SUBSCRIBE, NOTIFY, INFO, REGISTER >CSeq: 101 INVITE >Max-Forwards: 10 >Timestamp: 1150227475 >Contact: <sip:+19199915651@172.18.109.91:5060> >Expires: 180 >Allow-Events: telephone-event >Content-Type: multipart/mixed;boundary=uniqueBoundary >Mime-Version: 1.0 >Content-Length: 797 > >--uniqueBoundary >Content-Type: application/sdp >Content-Disposition: session;handling=required > >v=0 >o=CiscoSystemsSIP-GW-UserAgent 8340 2382 IN IP4 172.18.109.91 >s=SIP Call >c=IN IP4 172.18.109.91 >t=0 0 >m=audio 19122 RTP/AVP 18 3 0 4 100 101 >c=IN IP4 172.18.109.91 >a=rtpmap:18 G729/8000 >a=fmtp:18 annexb=yes >a=rtpmap:3 GSM/8000 >a=rtpmap:0 PCMU/8000 >a=rtpmap:4 G723/8000 >a=fmtp:4 annexa=yes >a=rtpmap:100 X-NSE/8000 >a=fmtp:100 192-194 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-16 > >--uniqueBoundary >Content-Type: application/gtd >Content-Disposition: signal;handling=optional > >IAM, >PRN,isdn*,,NI***, >USI,rate,c,s,c,1 >USI,lay1,ulaw >TMR,00 >CPN,02,,1,4724170 >CGN,04,,1,y,2,9199915651 >CPC,09 >FCI,,,,,,,y, >GCI,f23fb314fa4a11da8098000653454c7e > > >--uniqueBoundary-- > > > >-- >Greg Fausak >greg(a)thursday.com >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] openser forking
by Padmaja RV 14 Jun '06

14 Jun '06

Hi all, i need to implement openser for forking an incoming sip call. can anyone pls guide me how to do this?

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Re: [Users] openser forking
by Padmaja RV 14 Jun '06

14 Jun '06

Hi, sorry, i could not understand what u meant by that.... please elaborate a bit, Thanks, Padmaja ----- Original Message ----- From: ram To: Padmaja RV Sent: Wednesday, June 14, 2006 4:30 PM Subject: Re: [Users] openser forking Ngrep should help you for the same ram On 6/14/06, Padmaja RV <padmaja.rv(a)vodcalabs.com> wrote: Hi all, i need to implement openser for forking an incoming sip call. can anyone pls guide me how to do this? _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] Openser as front end to Asterisk
by Kenny Chua 14 Jun '06

14 Jun '06

Hello everyone, I've finally got openser working with mysql. Now I'm trying to integrate Openser with Asterisk. I read Asterisk at large. However I have a few questions: 1) It is said that openser as a front end to asterisk will enable more concurrent calls comparing to asterisk alone. How is that so? 2) If openser acts as a front end connecting all SIP calls, what is the role of asterisks? 3) Will the load of SIP calls be handled by openser alone? Or is it both asterisk and openser? I'm confuse with this, please help a newbie out here. Thank you in advance. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] rewriting the outgoing URI to its alias
by Andy Thomas 14 Jun '06

14 Jun '06

Greger, Im already doing this for calls coming into my SER installation. However I want to do it the other way around, i.e. from SER/UA to PSTN Therefore I want to rewrite the username if the outgoing INVITE message to be the alias, not the location, when sending to the PSTN. Andy _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: 12 June 2006 09:33 To: Andy Thomas Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] rewriting the outgoing URI to its alias I see a lot of confusion on the different user related ids in SIP. Let me try to explain (and answer your question, Andy, at the end of the post...) Let's start at the user agent(UA) side: -------------------- AOR: Address of record is the sip:myuser@domain.com address that you are known as, just like your email address. Many user agents don't ask for AOR, but builds it from other parameters you need to set Username: The username is normally the user (before @) of AOR) Realm: Often the domain portion of the AOR, thus AOR = sip:username@realm Friendly name/displayname: "My name" Out of this info, the From header is constructed: From: "My name" <sip:username@realm> --- Entirely separate (but sometimes confused in user agents): Authentication username: The user used for authentication Authentication realm: The realm to authenticate within Very often the authentication realm is implicitly assumed to be the same as the AOR realm --- Then, the user agent will create the Contact header. The Contact header should be the public contact address of your current location. Thus: Contact: sip:username@mypublicip:5060 If the user agent is behind a NAT, the mypublicip will be a private address. --- A sidenote: Unless registration server is explicitly specified, the realm in the AOR will be used for looking up the SIP registration server using DNS SRV or A lookups. You should avoid putting the FQDN of your SIP server in the realm. Also, you may in some user agents specify outbound proxy. This is the proxy where the user agent will send INVITEs (and other outbound messages). ------------------- And on the SER side: ------------------- Authentication user/realm are used to do Digest authentication, but are then forgotten (i.e. not stored). The AOR is registered in the location table, together with the Contact header, as well as the source ip and port (if different from Contact). The fix_nated_register() function handles this setting of the so-called received parameter. So, to the routing: - Messages that need to be routed (i.e. do not have Route headers) will have a Request URI; the first line and the part after the message type: INVITE sip:username@domain.com - It is by changing this request URI, you do routing. The t_relay() command uses the URI to forward the message correctly - Before forwarding to a user agent, you want the request uri to be the same as the stored Contact header for the AOR you are looking up. If not, the user agent may reply with a 404 User not found - The From header is NOT used for routing, and for backwards RFC compatibility, you should not change the From header as some UAs will use the content of the From header to match a dialog (however, if your UAs from experience still work, there should not be a problem doing it, it's just not RFC-compliant and may pop up and kick your butt later ;-) - The AOR in location table is used for looking up incoming messages if you can find a direct match between the Request-URI in the incoming message and the stored AOR. If not, you can use the aliases table (and lookup("aliases")) to match the Request-URI with something in the aliases table, that again will map to the AOR in the location table --- So keeping this (fairly simple) concept in focus: Routing (regardless of LCR, avps or whatever) should focus on finding the correct Request-URI before you call t_relay(). BUT, there are ways of "messing" up this... --- - There are several commands in SER you can use to override the Request-URI (forward_*). They should be avoided, unless you have a valid reason for having a Request-URI in the message you are forwarding that is NOT resolvable (either IP address or DNS name or DNS SRV/A resolvable) to the party you are forwarding to. The reason can be if you want the R-URI to contain the AOR and then forward the message to a server handling voicemail - The dst_uri parameter (implicitly set by lookup) will tell t_relay() to send the message to dst_uri instead of the Request-URI. lookup() will set this when it finds that the Contact stored for the AOR also has an associated received ip:port (because the user agent was NATed) In general trying all sorts of tricks o fix-up things the way you need it may not be so smart. Stick to the basics and question yourself: Is this something I really want to do? ---------------------- To Andy, you want the aliases table to map to the AOR (as registered by the UA in the REGISTER command). You do lookup("aliases") to resolve your DID into an AOR. Then, later you can do lookup("location") to map the AOR to the location of the UA (i.e. Contact/dst_uri). Then your Request-URI will be correct. g-) Andy Thomas wrote: Does anyone know how I would do this- For all users who have PSTN access, they are assigned a number in the MySQL alias table which matches a PSTN DDI number. e.g. user 8000 has an alias of 2071231234, so on an incoming call the lookup("aliases") function correctly matches the DDI to the user. I want my ser.cfg to rewrite the user on an outgoing call, if a number exists in the alias table for that user Obviously, the rewriteuser function will be used, but what do I put in after that? I have tried rewriteuser (lookup ("aliases")) but that doesn't work Can anyone help? _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 3

[Users] OpenSER as a remote side Outbound Proxy?
by Enky 14 Jun '06

14 Jun '06

Is it possible to configure the OpenSER with nathelper/mediaproxy to act as a remote-side outbound proxy? I have tried to make it, but unfortunately still can not do that. It will be best if someone shares a working openser.cfg file. The best I got is that when the UA makes call it works OK, but the UA can not be dialed from other UA in the network. Maybe it is not even registered.

1 0

[Serusers] Serctl commands don't work
by Alois Schafferhofer 14 Jun '06

14 Jun '06

Hi, I have installed the Presence Version of SER on a SUSE Linux Server. If I want to add a user with the command "serctl add ...", I get following message: "error: 500 Command 'ul_show_contact' not found" The same with serctl moni: 500 Command 'version' not found 500 Command 'uptime' not found ... I hope, someone can help me. Thanks, Alois

1 0

[Users] redirect
by Andrew And 14 Jun '06

14 Jun '06

Does someone has some example with redirect in OpenSER? And could send it?

2 1

[Users] Realtime MYSQL
by Shane Burrell 14 Jun '06

14 Jun '06

Is the mysql auth supposed to be realtime? I seem to have to restart openser when I make a change to the alias or subscriber in openser. Thanks, shane ***************************************************************

2 1

[Serusers] Content-Type: multipart/mixed; boundary=uniqueBoundary problems?
by Greg Fausak 14 Jun '06

14 Jun '06

Before I start debugging I thought I'd ask. I have a sip message from a gateway (Cisco IOS) that is being sliently dropped. Should ser be able to handle a message that looks like this? I haven't done a multipart message before. Thanks, ---greg INVITE sip:+19194724170@sn-sip-in.ca-sn1.cisco.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.109.91:5060;branch=z9hG4bK4F0109C Remote-Party-ID: <sip: +19199915651(a)172.18.109.91>;party=calling;screen=yes;privacy=off From: <sip:+19199915651@172.18.109.91>;tag=249E4980-FFC To: <sip:+19194724170@sn-sip-in.ca-sn1.cisco.com> Date: Tue, 13 Jun 2006 19:37:55 GMT Call-ID: F241878C-FA4A11DA-81EFC5C8-2F1D7951(a)172.18.109.91 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 4064260884-4199158234-2157445126-1397050494 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 10 Timestamp: 1150227475 Contact: <sip:+19199915651@172.18.109.91:5060> Expires: 180 Allow-Events: telephone-event Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 797 --uniqueBoundary Content-Type: application/sdp Content-Disposition: session;handling=required v=0 o=CiscoSystemsSIP-GW-UserAgent 8340 2382 IN IP4 172.18.109.91 s=SIP Call c=IN IP4 172.18.109.91 t=0 0 m=audio 19122 RTP/AVP 18 3 0 4 100 101 c=IN IP4 172.18.109.91 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,NI***, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,02,,1,4724170 CGN,04,,1,y,2,9199915651 CPC,09 FCI,,,,,,,y, GCI,f23fb314fa4a11da8098000653454c7e --uniqueBoundary-- -- Greg Fausak greg(a)thursday.com

2 1

[Serusers] Presence's libs
by Rosalvina Romero 14 Jun '06

14 Jun '06

Hello, I have Ser 0.9.6, and I'm trying to install the Presence model to it. I have recompiled SER but it doesn't generate all the libraries that are needed. In particular I can't find libxcap, libcds nor libpresence. Could you tell me where I could find them, I have been looking for them in several websites but I couldn't find them. Thanks in advance Ragards, Rose __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] Redirect
by Anderson Alves Albuquerque 14 Jun '06

14 Jun '06

Does someone has some example with redirect in SER and could send it? -- [], Anderson

2 3

RE: [Serusers] Sip soft phones for XP with tcp support
by Benjamin.George＠t-systems.com 14 Jun '06

14 Jun '06

Hi, There are a lot of softphones available for free/trial with audio/conf support. But softphones with video, IM, WB, AS, etc support are all paid ones. You can try SJPhone from SJ Labs (www.sjlabs.com <http://www.sjlabs.com/> ). It has got support for TCP too. Regards, Benjamin. _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of sandeep kamath Sent: Wednesday, June 14, 2006 2:52 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Sip soft phones for XP with tcp support Hi , Are there any good softphones that have support for tcp other than the one by counterpath/xten . regards, sandeep,

1 0

[Serusers] Append prefix after load_gw()?
by Ryan Pagquil 14 Jun '06

14 Jun '06

Hi, Is it possible to append prefix after load_gw() in lcr? I need to prefix the RURI so that I comply with our provider requirements. Is this possible? I'm using "prefix" before "rewritehost" and this is working but how about "load_gw" first then "prefix" based on the gateway obtained from LCR? Thanks in advance, Ryan

3 7

[Users] OpenSER performance
by Helge Waastad 13 Jun '06

13 Jun '06

Hi, just want to tell you that if you want so see what performance you'll get out of your OpenSER proxy, I found out by using a software called super-smack I easily can generate a typical OpenSER sql query and see how many qps I can get. ...nice to have :-) br hw -- Helge Waastad Senior Konsulent Smartnet

4 3

[Serusers] Sip soft phones for XP with tcp support
by sandeep kamath 13 Jun '06

13 Jun '06

Hi , Are there any good softphones that have support for tcp other than the one by counterpath/xten . regards, sandeep,

1 0

[Serusers] Problems with starting SER
by Alois Schafferhofer 13 Jun '06

13 Jun '06

Hi, I have following problem with SER. I have downloaded the Presence Snapshot version ser-0.10.99-dev-pa-4.1, but I can't start it. My config file looks like that: debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=2 alias="fh-joanneum.at" #user=ser #group=ser #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #fifo="/tmp/ser_fifo" sip_warning=yes If I want start SER, it generates a PID, but with the command ps -aux, I can't find this process. With the command serctl ps, I get the message: Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config. OEFEG:~ # serctl start Starting SER : started pid(7431) OEFEG:~ # ser 0(7435) WARNING: fix_socket_list: could not rev. resolve 10.15.200.19 0(7435) WARNING: fix_socket_list: could not rev. resolve 10.21.200.50 0(7435) WARNING: fix_socket_list: could not rev. resolve 10.15.200.19 0(7435) WARNING: fix_socket_list: could not rev. resolve 10.21.200.50 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.15.200.19 [10.15.200.19]:5060 udp: 10.21.200.50 [10.21.200.50]:5060 udp: 10.0.0.11 [10.0.0.11]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.15.200.19 [10.15.200.19]:5060 tcp: 10.21.200.50 [10.21.200.50]:5060 tcp: 10.0.0.11 [10.0.0.11]:5060 Aliases: tcp: OEFEG:5060 tcp: OEFEG.fh-joanneum.at:5060 tcp: localhost:5060 udp: OEFEG:5060 udp: OEFEG.fh-joanneum.at:5060 udp: localhost:5060 *: fh-joanneum.at:* 0(7435) init_tcp: using epoll_lt as the io watch method (auto detected) 0(7437) AVPops - initializing OEFEG:~ # 0(7437) Maxfwd module- initializing 0(7437) using 'none' rls-subscription authorization! 0(7437) using 'none' subscription authorization for watcher info! If I try to uncomment the line fifo="/tmp/ser_fifo", I get the following message: OEFEG:~ # serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed OEFEG:~ # /base/ser/directory/sbin/ser -f /base/ser/directory/etc/ser/ser.cfg 0(7383) parse error (19,5-6): syntax error 0(7383) parse error (19,5-6): unknown config variable 0(7383) parse error (19,7-8): ERROR: bad config file (3 errors) I hope, someone can help me. Thanks, Alois

3 2

[Users] mysql re-install??
by Mark Kent 13 Jun '06

13 Jun '06

Hello, Using openser 1.0.1, I'm trying to add multi-domain support using the module "domain".... so I created a database in mysql, stuffed a few entries in it, pointed at it in openser.cfg, and started openser. I get this: 0(0) ERROR: domain:mod_init(): invalid table version (use ser_mysql.sh reinstall) 0(0) init_mod(): Error while initializing module domain Now, I looked into openser_mysql.h and it has a table "version" that lists versions for various databases: INSERT INTO version VALUES ( 'subscriber', '5'); ... INSERT INTO version VALUES ( 'location', '1001'); INSERT INTO version VALUES ( 'aliases', '1001'); ... INSERT INTO version VALUES ( 'gw_grp', '1'); INSERT INTO version VALUES ( 'lcr', '1'); So, what exactly are these "versions"?? How is it possible that a new database that I create/name/point-at should match something in this static list? Thanks, -mark

2 4

[Users] bad avp flags error
by Raymond Chen 13 Jun '06

13 Jun '06

Dear all, trying to upgrade the 1.0.1 configuration to cvs head to test the path module. I keep getting this error no matter what I do. avp_db_load("$from","s:areacode/areacodes") 0(0) ERROR:avops:parse_avp_db:: error - bad avp flags 0(0) ERROR:avpops:fixup_db_avp: parse failed 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 251 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration thanks Ray

2 1

[Users] Openser+serweb
by ram 13 Jun '06

13 Jun '06

Hi i have download servweb 0.9.4 and installed in FC5, configured as per the installtion documented when i try to access from web, i get error to install classkit when i install classkit, its not available on that site at all how can i rectify this problem Another problem, how can i make accounting system for the user I dont see rate table in my openser databases,when can i export my rate table and integrate with user for billing ram

1 0

[Users] More debugging info?
by Bret Schuhmacher 13 Jun '06

13 Jun '06

Hello all, I'm trying to debug my OpenSER installation. I've edited the config file, told it not to fork and set debug to various numbers (10,30,100,300). All the debugging is to stdout. The problem is that it seems to stop after initialization - I was hoping to see messages for everything - REGISTRATION, INVITE, etc. How can I get a LOT of debug info? Thanks, Bret

2 1

[Users] Jabber & PA modules
by Rodney G. McDuff 13 Jun '06

13 Jun '06

Hi All What is the current state of play with the openser jabber and pa modules? -- Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group| Ex luce ad tenebras Information Technology Services | The University of Queensland | EMAIL: mcduff(a)its.uq.edu.au | TELEPHONE: +61 7 3365 8220 |

2 1

[Users] called failed 408 Request timeout
by raviprakash sunkara 13 Jun '06

13 Jun '06

hi users, I need to say just "hello to all" I'm working on openser + radius.. Actually in LAN network its working fine... When I deploy the openser + radius in My main server which is in some other place . its successfully login and When I trying to make call its show the log messages as " Called Failed :- 408 Request TimeOut.... " ========================================================================= modparam('registrar","nat_flag',6) modparam("nathelper","natping_interval",30) modparam("nathelper","ping_nated_only",1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #################33 modparam("rr", "enable_full_lr", 1) ############################################### route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("13")) { sl_send_reply("483","Too Many Hops........................!"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if(!method== "REGISTER") { record_route(); }; # loose-route processing if (loose_route()) { # add now 0n 9 setflag(1); acc_rad_request("200 ok "); t_relay(); exit; }; # account completed transactions via syslog if(uri==myself) { if(method=="REGISTER") { if(!radius_www_authorize("192.168.2.55")) { www_challenge("192.168.2.55","0"); exit; }; #consume_credentials(); save("location"); if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; if(method=="INVITE"|| method=="ACK") { record_route(); acc_rad_request("200"); if(isflagset(6)) { force_rtp_proxy("e"); }; #setflag(1); t_on_reply("1"); }; /* if (method=="BYE") { record_route(); }; */ if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; fix_nat_contact(); if (method=="BYE" || method=="CANCEL") { #log (1, "BYE or CANCEL\n"); #setflag(1); acc_rad_request("200"); #acc_rad_request("Stop"); unforce_rtp_proxy("e"); #setflag(1); }; if(!lookup("location")) { sl_send_reply("404","Woo......... NOt found"); }; }; if(!t_relay()) { sl_reply_error(); }; lookup("aliases"); #setflag(2); exit; } onreply_route[1] { if( isflagset96) && status=~"[0-9][0-9][0-9]") { fix_nat_contact(); force_rtp_proxy("e"); }; } ========================================================================= -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

2 1

[Users] Usrloc failure on startup
by Johannes Lehninger 13 Jun '06

13 Jun '06

Hi, I use openser 1.0.1 and I get the following error on starting it up 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing vm10:~ # 0(0) usrloc:preload_udomain: Wrong version v136427728 for table <location>, expected v1001 0(0) register_udomain(): Error while preloading domain 'location' 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration My config file is the default one except of the additional modules: mysql,mediaproxy, domain. loadmodule "/usr/local/openser10/lib/openser/modules/mysql.so" loadmodule "/usr/local/openser10/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/openser10/lib/openser/modules/domain.so" loadmodule "/usr/local/openser10/lib/openser/modules/xlog.so" .. # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/openser10/lib/openser/modules/auth.so" loadmodule "/usr/local/openser10/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("auth_db|usrloc|domain", "db_url", "mysql://openser:openserrw@localhost/openser10") modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) Regards Hannes

3 2

[Users] kphone and xlite with TLS
by D'Addelfio Davide 13 Jun '06

13 Jun '06

Hi to all, does anyone know if user agent Kphone and X-lite supports TLS in Openser? Thanks in advance for reply. Davide

2 1

[Users] Authentication using auth_diameter
by Dave Waiting 13 Jun '06

13 Jun '06

I am able to authenticate users using Radius without problem, but I am struggling to do the same using Diameter. I have installed the Diameter Server Client (DISC) that is mentioned in the auth_diameter readme, and have it listening on port 3000. When I start openSER there are no errors (i.e. the auth_diameter module loads), but the DISC client complains that the peer is unknown. Both SER and the DISC client are running on the same machine, so I added aaa://localhost as a peer, but that does not seem to help. Looking at a tcpdump output it shows that SER is trying to contact the DISC client from a range of ports but is never successful. Are there any further installation manuals, sample configurations or tutorials on this topic? Regards Dave

2 1

Re: [Users] DID Origination
by Bogdan-Andrei Iancu 13 Jun '06

13 Jun '06

Hi Kevin, the proper mailing list for this kind of topics is business(a)openser.org. regards, bogdan Kelvin Williams wrote: > This post is definitely going to be off-topic and I ask for your > forgiveness in advance. > > However, we are a cable ISP and are looking to provide voice services > to our cable modem subscribers. We’ve been in talks with several > companies which provide DID Origination and Termination but want to > make sure we’ve covered them all before making a decision on which to > pick, or would like to get suggestions from anyone in our same (or > like) position of companies to look at or stay away from. > > You can reply to me directly to prevent the list from being populated > with off-topic subjects. > > Thanks, > > Kw > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

Re: [Users] how to get sdp data
by Bogdan-Andrei Iancu 13 Jun '06

13 Jun '06

Hi Stefan, you may use pseudo-variables to access the body and regexp to get the fields you need into AVPs. Then you have the option to either write them via avp_db_store(), either to log them into accounting via the extra accounting feature. regards, bogdan Franz Edler wrote: >>I need to get the sdp data transmitted in SIP-Three-Way handshake. >>Especially the IP-Adresses and Portnumbers of the RTP stream but didn't >>find a solution for this yet. >> >> > >Look at functions offered by >http://openser.org/docs/modules/1.1.x/textops.html > > > >>Additionaly Information about what i'm doing in my master thesis: >>Take SDP Data mentioned above and send it from a PC running OpenSER to >>another PC working as IP-Router. On the IP-Router a traffic control >>filter will be added so that this router marks the DS-Field and >>priorizes the RTP-stream. >> >>So the PC with OpenSER works as a SIP-Proxy and as Policy-Server that >>configures the routers in a network. >> >>COPS Protocol is used for Communication between SIP-Proxy and IP- >>Router. >> >> > >Interesting. Keep us informed when finalized. > >Regards >Franz > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

[Users] RTPProxy problem
by Edgar Barbosa 13 Jun '06

13 Jun '06

Hi all, I'm having some problems with rtpproxy and openser... Although having rtpproxy running on the same machine, and using /var/run/rtpproxy.sock, openser seems not to be able to connect to it: 10(7357) ERROR: send_rtpp_command: can't connect to RTP proxy 10(7357) send_rtpp_command(): proxy <unix:/var/run/rtpproxy.sock> does not responding, disable it 10(7357) WARNING: rtpp_test: can't get version of the RTP proxy 10(7357) WARNING: rtpp_test: support for RTP proxy <unix:/var/run/rtpproxy.sock>has been disabled temporarily I'm using recent versions: openser-devel-cvs-20060608-141301.tgz rtpproxy-cvs-20060420-194006.tgz Any idea of what can be wrong? Thanks, Edgar

2 1

[Users] Allow only perticular country
by ram 13 Jun '06

13 Jun '06

Hi all Just working around with openser just looking config, and had one requirement I have users 3 i want 1 and 2 can access calls where ever they can but 3 need to call only USA, and he should not get access to rest of the place can some one give me idea, how can i achive this or any sample config ram

3 3

Re: [Serusers] DB Unknown error
by Zenone 13 Jun '06

13 Jun '06

Thank you Karel, now it works! But after subscribing i don't receive any mail to confirm my registration. I saw apache logs and serweb logs, and they are normal...no error. Thank you for your help! Michel --------- Message d'origine -------- De: Karel Kozlik <karel(a)iptel.org> A: Zenone <zenone(a)telip.com> Copies: serusers(a)lists.iptel.org Objet: Re: [Serusers] DB Unknown error Date: 13/06/06 11:43 > Hi, > look at these config options (config_data_layer.php) > Karel > > > /** > * Needs to be set when you are useing MySQL >= 4.1 > * see mysql manual for more info > * > * $config->data_sql->collation = "utf8_general_ci"; > */ > > $config->data_sql->collation = ""; > > /** > * Set to true when you are useing MySQL >= 4.1 > * This option set mysql system variables character_set_client, > * character_set_connection, and character_set_results to charset > * used in serweb > * > * $config->data_sql->set_charset = true; > */ > > $config->data_sql->set_charset = false; > > > > Zenone napsal(a): > > Hi, > > I still have troubles with my serveb... (thanks karel for answering my > > previous question) I saw the serweb log and it can be read : > > > > "" 'Jun 13 10:13:52 serweb [error] file: > > /usr/local/serweb-0.9.4/data_layer/method.get_aliases.php:27: DB Error: > > unknown error - select username, domain from aliases where > > lower(contact)=lower('sip:zenone@192.168.0.120') order by username > > [nativecode=1267 ** Illegal mix of collations (latin1_swedish_ci,IMPLICIT) > > and (utf8_general_ci,COERCIBLE) for operation '='] "" > > I followed karel advice of july 2005 and it doesn't work anyway. > > I work on a fedora with ser 0.9.6 and serweb 0.9.4. > > Thanks for your help! > > Michel > > > > ________________________________________________ > > Message sent using UebiMiau 2.7.8 > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > ________________________________________________ Message sent using UebiMiau 2.7.8

1 0

[Serusers] Problem in establishing communication between sip phones using Windows Messenger
by sumit dengri 13 Jun '06

13 Jun '06

sir, > I have WM version-5 and successfully connected two sip users created using serctl. >But when both users login to WM on different machines status of both users is "Offline". >Is there any problem in installation of SER. > I am using default configuration file for testing purpose as written in the docs. So kindly put some suggestions to help me. Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

1 0

[Serusers] DB Unknown error
by Zenone 13 Jun '06

13 Jun '06

Hi, I still have troubles with my serveb... (thanks karel for answering my previous question) I saw the serweb log and it can be read : "" 'Jun 13 10:13:52 serweb [error] file: /usr/local/serweb-0.9.4/data_layer/method.get_aliases.php:27: DB Error: unknown error - select username, domain from aliases where lower(contact)=lower('sip:zenone@192.168.0.120') order by username [nativecode=1267 ** Illegal mix of collations (latin1_swedish_ci,IMPLICIT) and (utf8_general_ci,COERCIBLE) for operation '='] "" I followed karel advice of july 2005 and it doesn't work anyway. I work on a fedora with ser 0.9.6 and serweb 0.9.4. Thanks for your help! Michel ________________________________________________ Message sent using UebiMiau 2.7.8

2 1

RE: [Serusers] How to connect sip phones into SER for tetsing Pur pose
by Benjamin.George＠t-systems.com 13 Jun '06

13 Jun '06

Hi, You can get more information on 'serctl' command and its options on the ser administration guide. Here goes the link: http://www.iptel.org/ser/doc/seruser/seruser.html#SERCTL It is worth downloading PDF version of that as it will help for further SER configurations, if any. Thanks, Benjamin. -----Original Message----- From: Andrey Kouprianov [mailto:andrey.kouprianov@gmail.com] Sent: Tuesday, June 13, 2006 2:04 PM To: serusers(a)iptel.org Subject: Re: [Serusers] How to connect sip phones into SER for tetsing Purpose There's not much about serctl to write about :). Just type serctl in the command line and press enter. You'll see the options it supports. On 6/13/06, Tony Weber <sanchosandybanks(a)hotmail.com> wrote: > Hi, > > In which document can I find documentation about serctl? > Thanx a lot!!! > > Tony > > > >From: "Andrey Kouprianov" <andrey.kouprianov(a)gmail.com> > >To: serusers(a)iptel.org > >Subject: Re: [Serusers] How to connect sip phones into SER for tetsing > >Purpose > >Date: Tue, 13 Jun 2006 13:07:16 +0700 > > > >Hi, > > > >You have to use "serctl add" command to add user to the database first. For > >ex, > > > >serctl add johndoe password johndoe(a)neverland.net > > > >Then configure your client with whatever username and password you just > >added. > > > >Regards. > > > >P.S. - you gotta READ. > > > >On 6/13/06, sumit dengri <sumitdengri(a)yahoo.com> wrote: > >>sir, > >>PLZ tell me step by step procedure to connect sip > >>phones for testing SER. > >> > >> >I would be grateful to all of you. > >>regards > >> > >>Send instant messages to your online friends http://in.messenger.yahoo.com > >> Stay connected with your friends even when away from PC. Link: > >>http://in.mobile.yahoo.com/new/messenger/ > >>_______________________________________________ > >>Serusers mailing list > >>Serusers(a)lists.iptel.org > >>http://lists.iptel.org/mailman/listinfo/serusers > >> > >_______________________________________________ > >Serusers mailing list > >Serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > _________________________________________________________________ > Was halten Sie von einer Seite, die all Ihre Lieblingsthemen beinhaltet? > http://at.msn.com/ > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] How to connect sip phones into SER for tetsing Purpose
by Andrey Kouprianov 13 Jun '06

13 Jun '06

There's not much about serctl to write about :). Just type serctl in the command line and press enter. You'll see the options it supports. On 6/13/06, Tony Weber <sanchosandybanks(a)hotmail.com> wrote: > Hi, > > In which document can I find documentation about serctl? > Thanx a lot!!! > > Tony > > > >From: "Andrey Kouprianov" <andrey.kouprianov(a)gmail.com> > >To: serusers(a)iptel.org > >Subject: Re: [Serusers] How to connect sip phones into SER for tetsing > >Purpose > >Date: Tue, 13 Jun 2006 13:07:16 +0700 > > > >Hi, > > > >You have to use "serctl add" command to add user to the database first. For > >ex, > > > >serctl add johndoe password johndoe(a)neverland.net > > > >Then configure your client with whatever username and password you just > >added. > > > >Regards. > > > >P.S. - you gotta READ. > > > >On 6/13/06, sumit dengri <sumitdengri(a)yahoo.com> wrote: > >>sir, > >>PLZ tell me step by step procedure to connect sip > >>phones for testing SER. > >> > >> >I would be grateful to all of you. > >>regards > >> > >>Send instant messages to your online friends http://in.messenger.yahoo.com > >> Stay connected with your friends even when away from PC. Link: > >>http://in.mobile.yahoo.com/new/messenger/ > >>_______________________________________________ > >>Serusers mailing list > >>Serusers(a)lists.iptel.org > >>http://lists.iptel.org/mailman/listinfo/serusers > >> > >_______________________________________________ > >Serusers mailing list > >Serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > _________________________________________________________________ > Was halten Sie von einer Seite, die all Ihre Lieblingsthemen beinhaltet? > http://at.msn.com/ > >

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[Serusers] How to connect sip phones into SER for tetsing Purpose
by sumit dengri 13 Jun '06

13 Jun '06

sir, PLZ tell me step by step procedure to connect sip phones for testing SER. >I would be grateful to all of you. regards Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

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Re: [Serusers] reregistration problems
by David 13 Jun '06

13 Jun '06

I am seeing the same thing at a few of our sites so if anyone has any insight, I would be interested as well. Thanks, Dave --------- Original Message -------- From: Giuseppe Parlato <gparlato(a)tnet.it> To: serusers(a)iptel.org <serusers(a)iptel.org> Subject: [Serusers] reregistration problems Date: 12/06/06 13:05 > > Hi > Sometimes my IP phones and ATAs.. they don't REregister and I have to reboot > them. I wonder if it can be a ser configuration problem. > > Thanks > Giuseppe > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

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[Users] multi-domain, non-unqiue userID
by Mark Kent 13 Jun '06

13 Jun '06

A couple of general questions about users and domains: a) The module 'domain' supports multiple domains, but does not appear to support non-unique user IDs spread across the domains. That is, if your openser box supports baz.net and bif.com, then frankie(a)baz.net and frankie(a)bif.com are "resolved" to the same user, and are both available as frankie(a)your-sip-proxy.com Is there anything readily available to support non-unique user names in a multi-domain scenario? b) On Sipura SPA equipment, and I'ld guess other equipment, one can specify a userID and also an authID. The authID is used, not surprisingly, for authentication with the proxy to register the userID. Is there any readily available module that puts restrictions on userIDs? That is, if I enable {foo, foopw} and {bar, barpw} then, right now, user foo can register to get calls to bar@ and user bar can register to get calls to foo@... that's not good. The way I'ld like this to work is very similar to an email server: there are sets of unique authIDs/passwords, and a relatively static set of aliases or virtual users. It's not possible for a POP mail user to route inbound email to arbitrary userIDs to himself (although outbound spoofing is easy). Any hints? Thanks, -mark

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[Serusers] DB Connectivity Error in SERWEB
by sumit dengri 13 Jun '06

13 Jun '06

sir >when I login with default or self created username and password.SERWEB shows the error: DB ERROR: Connect Failed Bad Username/Password >PLZ help me out, I got stuck there regards Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

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[Serusers] reregistration problems
by Giuseppe Parlato 13 Jun '06

13 Jun '06

Hi Sometimes my IP phones and ATAs.. they don't REregister and I have to reboot them. I wonder if it can be a ser configuration problem. Thanks Giuseppe

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[Users] DID Origination
by Kelvin Williams 12 Jun '06

12 Jun '06

This post is definitely going to be off-topic and I ask for your forgiveness in advance. However, we are a cable ISP and are looking to provide voice services to our cable modem subscribers. We've been in talks with several companies which provide DID Origination and Termination but want to make sure we've covered them all before making a decision on which to pick, or would like to get suggestions from anyone in our same (or like) position of companies to look at or stay away from. You can reply to me directly to prevent the list from being populated with off-topic subjects. Thanks, Kw

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[Serusers] Registering with serweb doesn't work
by Zenone 12 Jun '06

12 Jun '06

Hi, As you can see in the title, I have problems to create new account with serweb: -First, when I click on register (after following the subscribing steps), it doesn't work. An error message appears on the top of the page : ''DB Error: unknown error'' -Secondly, when I log in admin and I look at the users table, at the top of the table there are many ''DB Error: unknown error'' ( four ) and several user doesn't appear. I think that the number of error message is linked to the number of absent user in the table. All the users have been added thanks to the serctl command. I work on a fedora 4 and my ser and serweb are the latest (resp. 0.9.6 and 0.9.4). Thanks for your help, Michel ________________________________________________ Message sent using UebiMiau 2.7.8

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[Serusers] rewriting the outgoing URI to its alias
by Andy Thomas 12 Jun '06

12 Jun '06

Does anyone know how I would do this- For all users who have PSTN access, they are assigned a number in the MySQL alias table which matches a PSTN DDI number. e.g. user 8000 has an alias of 2071231234, so on an incoming call the lookup("aliases") function correctly matches the DDI to the user. I want my ser.cfg to rewrite the user on an outgoing call, if a number exists in the alias table for that user Obviously, the rewriteuser function will be used, but what do I put in after that? I have tried rewriteuser (lookup ("aliases")) but that doesn't work Can anyone help?

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[Serusers] DB Connection Error in Serweb
by sumit dengri 12 Jun '06

12 Jun '06

sir, when I try to login in user form using my username and password, following error comes: DB Error: Connect Failed Bad Username or password. Same error msg comes when I subscribe new user. PLZ help me to sort out this error or the cause of this error. Regards. Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

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[Users] how to get sdp data
by Stefan Roemer 12 Jun '06

12 Jun '06

hi openser users! i'm new in using OpenSER and new at this mailinglist, because I began working with OpenSER as part of my master thesis just a few weeks ago. So lets say "Hello" at you all! I need to get the sdp data transmitted in SIP-Three-Way handshake. Especially the IP-Adresses and Portnumbers of the RTP stream but didn't find a solution for this yet. Does anybody know a solution? Additionaly Information about what i'm doing in my master thesis: Take SDP Data mentioned above and send it from a PC running OpenSER to another PC working as IP-Router. On the IP-Router a traffic control filter will be added so that this router marks the DS-Field and priorizes the RTP-stream. So the PC with OpenSER works as a SIP-Proxy and as Policy-Server that configures the routers in a network. COPS Protocol is used for Communication between SIP-Proxy and IP-Router. Thank you very much and hope for useful answers. Stefan

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[Serusers] acc replaced by acc_db ?
by olivier.taylor 11 Jun '06

11 Jun '06

hello, just downloaded las ser version from cvs. It seems that the acc module doesn't exist in the tree anymore. Does the new acc modules (acc_db, ...) replaces acc? Do I miss? Olivier

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RE: [Serusers] rewriting the From URI
by Bjorn Asmul 10 Jun '06

10 Jun '06

Does anyone know the regular expression to replace the callerID name? from="Jane Doe" <sip:4445556666@10.0.1.1> -> from="John Doe" <sip:4445556666@10.0.1.1> Obviously using an AVP with the subst-function. ( subst('/^(From|f):(.*)sip:............); ) Thanks, Bjorn ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of sip Sent: Saturday, June 10, 2006 4:32 PM To: Andy Thomas; serusers(a)lists.iptel.org Cc: serusers-bounces(a)lists.iptel.org Subject: Re: [Serusers] rewriting the From URI You still wouldn't really want to rewrite the from URI.... there's no need to. Just rewrite the RURI on its way out to the PSTN. N. On Sat, 10 Jun 2006 17:10:34 +0100, Andy Thomas wrote > Does anyone know how to rewrite the From: URI for an incoming call? > All the rewrite functions are for outgoing URI's - i.e. local users, but I want to rewrite a URI that is incoming from the PSTN to one of those users. > > This is so I can add a prefix to it, so that if you dial a missed call, the prefix is automatically there so it will go out to the PSTN > >

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[Serusers] rewriting the From URI
by Andy Thomas 10 Jun '06

10 Jun '06

Does anyone know how to rewrite the From: URI for an incoming call? All the rewrite functions are for outgoing URI's - i.e. local users, but I want to rewrite a URI that is incoming from the PSTN to one of those users. This is so I can add a prefix to it, so that if you dial a missed call, the prefix is automatically there so it will go out to the PSTN

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[Users] Need OpenSER Experts
by John Peters 10 Jun '06

10 Jun '06

Hi All, We are an established company seeking a SIP and OpenSER expert to assist in the development and roll-out of a new SIP-based service. If you're interested in working on an innovative new service, and have an excellent understand of VoIP and OpenSER, please contact us! We look forward to any responses. Thanks! John Peters petersprc(a)gmail.com

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[Users] troubles with usrloc module: db_mode=3 (DB only)
by Christian Schlatter 09 Jun '06

09 Jun '06

Hi all, Running openser (CVS HEAD) with usrloc:db_mode=3 causes it to crash repeatedly after 15 secs or so. The core dump does not say much except (gdb) bt #0 0x00b2b392 in ?? () #1 0x08131468 in mem_pool () #2 0xbfff8b54 in ?? () #3 0xbfff8b54 in ?? () #4 0x00000008 in ?? () #5 0x00b3adbf in ?? () #6 0x00000007 in ?? () #7 0x00b3a88c in ?? () #8 0x00000006 in ?? () #9 0x00b3adcd in ?? () #10 0x00000005 in ?? () #11 0x00b3adae in ?? () #12 0xb576a560 in ?? () #13 0x00000007 in ?? () #14 0x00b39c59 in ?? () #15 0x00000005 in ?? () #16 0x00b3adae in ?? () #17 0x00000001 in ?? () #18 0x00000001 in ?? () #19 0x00000000 in ?? () The openser log looks like: ERROR: receive_fd: EOF on 16 child process 18101 exited by a signal 11 core was generated I also don't see any mysql errors. The last mysql query issued by openser before dying is always Query delete from location where expires<'2006-06-05 16:03:17' (only the timestamp changes) Deleting all location table entries before starting up openser does not help either. Is there something special I have to consider when running openser in DB-only mode? Thanks for any help, Christian

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[Serusers] how to catch an asterisk *
by bails 09 Jun '06

09 Jun '06

Hi all how do i catch a dtmf * in ser.cfg thanks Bails

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Re: [Users] openserctl error.
by Daniel-Constantin Mierla 09 Jun '06

09 Jun '06

I have changed openserctl to do the computation of GW_IP=$(($1+$2*256+$3*65536+$4*16777216)) within mysql query. The version for dbtext uses "expr" (more portable) if available. It is available only on development (frozen) branch. Cheers, Daniel On 06/09/06 17:55, Kelvin Williams wrote: > Changed the /bin/sh to /bin/bash at the beginning of the script and it > works. > > Kw > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Friday, June 09, 2006 10:31 AM > To: Ovidiu Sas > Cc: Kelvin Williams; users(a)openser.org > Subject: Re: [Users] openserctl error. > > It seems that only bash understands expressions like > $(($1+$2*256+$3*65536+$4*16777216)). Maybe using expr is better. > > Kevin can you confirm that 'expr' is present in Solaris 10 box? On > debian it is part of coreutils. > > Cheers, > Daniel > > > On 06/09/06 17:28, Ovidiu Sas wrote: > >> What shell are you using? >> It looks like your shell doesn't understand the setting of GW_IP syntax. >> >> >> -ovi >> >> On 6/8/06, Kelvin Williams <kelvin-lists(a)williamschadwell.com> wrote: >> >>> >>> >>> This is a new OpenSER installation on a Solaris 10 box. >>> >>> >>> >>> Any time we call openserctl we receive the following error: >>> >>> >>> >>> /usr/local/sbin/openserctl: syntax error at line 1026: `GW_IP=$' >>> unexpected >>> >>> >>> >>> Line 26 of openserctl reads: >>> >>> >>> >>> GW_IP=$(($1+$2*256+$3*65536+$4*16777216)) >>> >>> >>> >>> Any advice would be greatly appreciated! >>> >>> >>> >>> Thanks, >>> >>> kw >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Users] rlm_sql is not updating the record...
by raviprakash sunkara 09 Jun '06

09 Jun '06

Hi .... I'm working openser + mysql + Freeradius.... when i'm making a call for accounting, first it say " Acct-Status-Type = Failed " and then its " Acct-Status-Type = start " After that when the User hang up the call , it didn't update the Acct Stop time and inserting the t2 records ... And also what means this...... rlm_sql (sql): Reserving sql socket id: 2 radius_xlat: 'stop packet with zero session length. [user '666(a)192.168.2.55', nas '192.168.2.55']' rlm_sql (sql) in sql_accounting: stop packet with zero session length. [user '666(a)192.168.2.55', nas '192.168.2.55'] rlm_sql (sql): Released sql socket id: 2 This query is not updating ---> " 'UPDATE radacct SET AcctStopTime = '2006-06-09 19:57:27', AcctSessionTime = '', AcctInputOctets = '', AcctOutputOctets = '', AcctTerminateCause = '', AcctStopDelay = '0', ConnectInfo_stop = '' WHERE AcctSessionId = ' 76788233-9611-4A96-9FAA-E829F175C69C(a)192.168.2.20' AND NASIPAddress = ' 192.168.2.55'' " why ? and this query is inserting -----> " INSERT into radacct (AcctSessionId, AcctUniqueId, UserName, Realm, NASIPAddress, NASPortId, NASPortType, AcctStartTime, AcctStopTime, AcctSessionTime, AcctAuthentic, ConnectInfo_start, ConnectInfo_stop, AcctInputOctets, AcctOutputOctets, CalledStationId, CallingStationId, AcctTerminateCause, ServiceType, FramedProtocol, FramedIPAddress, AcctStartDelay, AcctStopDelay) values(' 76788233-9611-4A96-9FAA-E829F175C69C(a)192.168.2.20', '63170499b83f412d', ' 666(a)192.168.2.55', '', '192.168.2.55', '5060', '', DATE_SUB('2006-06-09 19:57:27', INTERVAL (0 + 0) SECOND), '2006-06-09 19:57:27', '', '', '', '', '', '', 'sip:555@192.168.2.55', 'sip:666@192.168.2.55', '', 'Sip-Session', '', '', '0', '0')' " Acct-Session-time is zero showing ? why ? Can U see the below one .... I need to change . please guys help me , this very important to complete the task .... please . rad_recv: Accounting-Request packet from host 192.168.2.55:33879, id=219, length=233 Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = BYE User-Name = "666(a)192.168.2.55" Calling-Station-Id = "sip:666@192.168.2.55" Called-Station-Id = "sip:555@192.168.2.55" Sip-Translated-Request-URI = "sip:555@192.168.2.21:5060" Acct-Session-Id = "76788233-9611-4A96-9FAA-E829F175C69C(a)192.168.2.20 " Sip-To-Tag = "2044753626" Sip-From-Tag = "4110988557" Sip-Cseq = "57538" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 192.168.2.55 Processing the preacct section of radiusd.conf modcall: entering group preacct for request 2 modcall[preacct]: module "preprocess" returns noop for request 2 rlm_acct_unique: Hashing 'NAS-Port = 5060,Client-IP-Address = 192.168.2.55,NAS-IP-Address = 192.168.2.55,Acct-Session-Id = " 76788233-9611-4A96-9FAA-E829F175C69C(a)192.168.2.20",User-Name = " 666(a)192.168.2.55"' rlm_acct_unique: Acct-Unique-Session-ID = "63170499b83f412d". modcall[preacct]: module "acct_unique" returns ok for request 2 rlm_realm: Looking up realm "192.168.2.55" for User-Name = " 666(a)192.168.2.55" rlm_realm: No such realm "192.168.2.55" modcall[preacct]: module "suffix" returns noop for request 2 modcall[preacct]: module "files" returns noop for request 2 modcall: leaving group preacct (returns ok) for request 2 Processing the accounting section of radiusd.conf modcall: entering group accounting for request 2 radius_xlat: '/usr/local/var/log/radius/radacct/192.168.2.55/detail-20060609' rlm_detail: /usr/local/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /usr/local/var/log/radius/radacct/192.168.2.55/detail-20060609 modcall[accounting]: module "detail" returns ok for request 2 radius_xlat: '/usr/local/var/log/radius/radutmp' radius_xlat: '666(a)192.168.2.55' modcall[accounting]: module "radutmp" returns ok for request 2 radius_xlat: '666(a)192.168.2.55' rlm_sql (sql): sql_set_user escaped user --> '666(a)192.168.2.55' radius_xlat: 'UPDATE radacct SET AcctStopTime = '2006-06-09 19:57:27', AcctSessionTime = '', AcctInputOctets = '', AcctOutputOctets = '', AcctTerminateCause = '', AcctStopDelay = '0', ConnectInfo_stop = '' WHERE AcctSessionId = '76788233-9611-4A96-9FAA-E829F175C69C(a)192.168.2.20' AND NASIPAddress = '192.168.2.55'' rlm_sql (sql): Reserving sql socket id: 2 radius_xlat: 'stop packet with zero session length. [user '666(a)192.168.2.55', nas '192.168.2.55']' rlm_sql (sql) in sql_accounting: stop packet with zero session length. [user '666(a)192.168.2.55', nas '192.168.2.55'] rlm_sql (sql): Released sql socket id: 2 radius_xlat: 'INSERT into radacct (AcctSessionId, AcctUniqueId, UserName, Realm, NASIPAddress, NASPortId, NASPortType, AcctStartTime, AcctStopTime, AcctSessionTime, AcctAuthentic, ConnectInfo_start, ConnectInfo_stop, AcctInputOctets, AcctOutputOctets, CalledStationId, CallingStationId, AcctTerminateCause, ServiceType, FramedProtocol, FramedIPAddress, AcctStartDelay, AcctStopDelay) values(' 76788233-9611-4A96-9FAA-E829F175C69C(a)192.168.2.20', '63170499b83f412d', ' 666(a)192.168.2.55', '', '192.168.2.55', '5060', '', DATE_SUB('2006-06-09 19:57:27', INTERVAL (0 + 0) SECOND), '2006-06-09 19:57:27', '', '', '', '', '', '', 'sip:555@192.168.2.55', 'sip:666@192.168.2.55', '', 'Sip-Session', '', '', '0', '0')' rlm_sql (sql): Released sql socket id: 2 modcall[accounting]: module "sql" returns noop for request 2 modcall: leaving group accounting (returns ok) for request 2 Sending Accounting-Response of id 219 to 192.168.2.55 port 33879 -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

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Re: [Users] openserctl error.
by Ovidiu Sas 09 Jun '06

09 Jun '06

What shell are you using? It looks like your shell doesn't understand the setting of GW_IP syntax. -ovi On 6/8/06, Kelvin Williams <kelvin-lists(a)williamschadwell.com> wrote: > > > > > This is a new OpenSER installation on a Solaris 10 box. > > > > Any time we call openserctl we receive the following error: > > > > /usr/local/sbin/openserctl: syntax error at line 1026: `GW_IP=$' unexpected > > > > Line 26 of openserctl reads: > > > > GW_IP=$(($1+$2*256+$3*65536+$4*16777216)) > > > > Any advice would be greatly appreciated! > > > > Thanks, > > kw > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Users] insering the 3 record in radius db my using openser+radius+mysql
by raviprakash sunkara 09 Jun '06

09 Jun '06

Hi I'm using openser and freeradius server . Accounting part in openser and radius by using the mysql. help me ... Below one is radius server in debug mode..... in that is didn't understand what is " unsupported acct-status-type =15 " and " send packet with zero session length " please help me..... rad_recv: Accounting-Request packet from host 192.168.2.55:35094, id=151, length=220 Acct-Status-Type = Failed Service-Type = Sip-Session Sip-Response-Code = 0 Sip-Method = INVITE User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Calling-Station-Id = "sip:666@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Called-Station-Id = "sip:555@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Sip-Translated-Request-URI = "sip:555@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Acct-Session-Id = "893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Sip-To-Tag = "n/a" Sip-From-Tag = "757680932" Sip-Cseq = "18235" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 192.168.2.55 Processing the preacct section of radiusd.conf modcall: entering group preacct for request 2 modcall[preacct]: module "preprocess" returns noop for request 2 rlm_realm: Looking up realm "192.168.2.55" for User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " rlm_realm: No such realm "192.168.2.55" modcall[preacct]: module "suffix" returns noop for request 2 modcall: leaving group preacct (returns noop) for request 2 Processing the accounting section of radiusd.conf modcall: entering group accounting for request 2 rlm_acct_unique: Hashing 'NAS-Port = 5060,Client-IP-Address = 192.168.2.55,NAS-IP-Address = 192.168.2.55,Acct-Session-Id = "893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>",User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> "' rlm_acct_unique: Acct-Unique-Session-ID = "5488f64bd8d99c57". modcall[accounting]: module "acct_unique" returns ok for request 2 radius_xlat: '/usr/local/var/log/radius/radacct/192.168.2.55/detail-20060606' rlm_detail: /usr/local/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /usr/local/var/log/radius/radacct/192.168.2.55/detail-20060606 modcall[accounting]: module "detail" returns ok for request 2 modcall[accounting]: module "unix" returns noop for request 2 radius_xlat: '/usr/local/var/log/radius/radutmp' rlm_radutmp: NAS radiusdtest port 5060 unknown packet type 15) modcall[accounting]: module "radutmp" returns noop for request 2 rlm_sql (sql): Unsupported Acct-Status-Type = 15 modcall[accounting]: module "sql" returns noop for request 2 modcall: leaving group accounting (returns ok) for request 2 Sending Accounting-Response of id 151 to 192.168.2.55 port 35094 Finished request 2 Going to the next request --- Walking the entire request list --- Waking up in 6 seconds... --- Walking the entire request list --- Cleaning up request 2 ID 151 with timestamp 44857cc6 Nothing to do. Sleeping until we see a request. rad_recv: Accounting-Request packet from host 192.168.2.55:35096, id=152, length=232 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = ACK User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Calling-Station-Id = "sip:666@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Called-Station-Id = "sip:555@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Sip-Translated-Request-URI = "sip:555@192.168.2.21<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> :5060" Acct-Session-Id = "893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Sip-To-Tag = "2710541959" Sip-From-Tag = "757680932" Sip-Cseq = "18235" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 192.168.2.55 Processing the preacct section of radiusd.conf modcall: entering group preacct for request 3 modcall[preacct]: module "preprocess" returns noop for request 3 rlm_realm: Looking up realm "192.168.2.55" for User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " rlm_realm: No such realm "192.168.2.55" modcall[preacct]: module "suffix" returns noop for request 3 modcall: leaving group preacct (returns noop) for request 3 Processing the accounting section of radiusd.conf modcall: entering group accounting for request 3 rlm_acct_unique: Hashing 'NAS-Port = 5060,Client-IP-Address = 192.168.2.55,NAS-IP-Address = 192.168.2.55,Acct-Session-Id = "893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>",User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> "' rlm_acct_unique: Acct-Unique-Session-ID = "5488f64bd8d99c57". modcall[accounting]: module "acct_unique" returns ok for request 3 radius_xlat: '/usr/local/var/log/radius/radacct/192.168.2.55/detail-20060606' rlm_detail: /usr/local/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /usr/local/var/log/radius/radacct/192.168.2.55/detail-20060606 modcall[accounting]: module "detail" returns ok for request 3 modcall[accounting]: module "unix" returns ok for request 3 radius_xlat: '/usr/local/var/log/radius/radutmp' radius_xlat: '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' modcall[accounting]: module "radutmp" returns ok for request 3 radius_xlat: '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' rlm_sql (sql): sql_set_user escaped user --> '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' radius_xlat: 'INSERT into radacct (AcctSessionId, AcctUniqueId, UserName, Realm, NASIPAddress, NASPortId, NASPortType, AcctStartTime, AcctStopTime, AcctSessionTime, AcctAuthentic, ConnectInfo_start, ConnectInfo_stop, AcctInputOctets, AcctOutputOctets, CalledStationId, CallingStationId, AcctTerminateCause, ServiceType, FramedProtocol, FramedIPAddress, AcctStartDelay, AcctStopDelay) values('893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ', '5488f64bd8d99c57', '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', '', '192.168.2.55', '5060', '', '2006-06-06 18:32:05', '0', '0', '', '', '', '0', '0', 'sip:555@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', 'sip:666@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', '', 'Sip-Session', '', '', '0', '0')' rlm_sql (sql): Reserving sql socket id: 2 rlm_sql (sql): Released sql socket id: 2 modcall[accounting]: module "sql" returns ok for request 3 modcall: leaving group accounting (returns ok) for request 3 Sending Accounting-Response of id 152 to 192.168.2.55 port 35096 Finished request 3 Going to the next request --- Walking the entire request list --- Waking up in 6 seconds... --- Walking the entire request list --- Cleaning up request 3 ID 152 with timestamp 44857ccd Nothing to do. Sleeping until we see a request. rad_recv: Accounting-Request packet from host 192.168.2.55:35097, id=153, length=232 Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = BYE User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Calling-Station-Id = "sip:666@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Called-Station-Id = "sip:555@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Sip-Translated-Request-URI = "sip:555@192.168.2.21<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> :5060" Acct-Session-Id = "893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Sip-To-Tag = "2710541959" Sip-From-Tag = "757680932" Sip-Cseq = "18236" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 192.168.2.55 Processing the preacct section of radiusd.conf modcall: entering group preacct for request 4 modcall[preacct]: module "preprocess" returns noop for request 4 rlm_realm: Looking up realm "192.168.2.55" for User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " rlm_realm: No such realm "192.168.2.55" modcall[preacct]: module "suffix" returns noop for request 4 modcall: leaving group preacct (returns noop) for request 4 Processing the accounting section of radiusd.conf modcall: entering group accounting for request 4 rlm_acct_unique: Hashing 'NAS-Port = 5060,Client-IP-Address = 192.168.2.55,NAS-IP-Address = 192.168.2.55,Acct-Session-Id = "893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>",User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> "' rlm_acct_unique: Acct-Unique-Session-ID = "5488f64bd8d99c57". modcall[accounting]: module "acct_unique" returns ok for request 4 radius_xlat: '/usr/local/var/log/radius/radacct/192.168.2.55/detail-20060606' rlm_detail: /usr/local/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /usr/local/var/log/radius/radacct/192.168.2.55/detail-20060606 modcall[accounting]: module "detail" returns ok for request 4 modcall[accounting]: module "unix" returns ok for request 4 radius_xlat: '/usr/local/var/log/radius/radutmp' radius_xlat: '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' modcall[accounting]: module "radutmp" returns ok for request 4 radius_xlat: '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' rlm_sql (sql): sql_set_user escaped user --> '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' radius_xlat: 'UPDATE radacct SET AcctStopTime = '2006-06-06 18:34:11', AcctSessionTime = '', AcctInputOctets = '', AcctOutputOctets = '', AcctTerminateCause = '', AcctStopDelay = '0', ConnectInfo_stop = '' WHERE AcctSessionId = '893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>' AND UserName = '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>' AND NASIPAddress = '192.168.2.55'' rlm_sql (sql): Reserving sql socket id: 1 radius_xlat: 'stop packet with zero session length. [user '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', nas '192.168.2.55']' rlm_sql (sql) in sql_accounting: stop packet with zero session length. [user '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', nas '192.168.2.55'] rlm_sql (sql): Released sql socket id: 1 radius_xlat: 'INSERT into radacct (AcctSessionId, AcctUniqueId, UserName, Realm, NASIPAddress, NASPortId, NASPortType, AcctStartTime, AcctStopTime, AcctSessionTime, AcctAuthentic, ConnectInfo_start, ConnectInfo_stop, AcctInputOctets, AcctOutputOctets, CalledStationId, CallingStationId, AcctTerminateCause, ServiceType, FramedProtocol, FramedIPAddress, AcctStartDelay, AcctStopDelay) values('893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ', '5488f64bd8d99c57', '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', '', '192.168.2.55', '5060', '', DATE_SUB('2006-06-06 18:34:11', INTERVAL (0 + 0) SECOND), '2006-06-06 18:34:11', '', '', '', '', '', '', 'sip:555@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ', 'sip:666@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', '', 'Sip-Session', '', '', '0', '0')' rlm_sql (sql): Released sql socket id: 1 modcall[accounting]: module "sql" returns noop for request 4 modcall: leaving group accounting (returns ok) for request 4 Sending Accounting-Response of id 153 to 192.168.2.55 port 35097 Finished request 4 Going to the next request --- Walking the entire request list --- Waking up in 6 seconds... rad_recv: Accounting-Request packet from host 192.168.2.55:35098, id=154, length=232 Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = BYE User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Calling-Station-Id = "sip:666@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Called-Station-Id = "sip:555@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Sip-Translated-Request-URI = "sip:555@192.168.2.21<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> :5060" Acct-Session-Id = "893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " Sip-To-Tag = "2710541959" Sip-From-Tag = "757680932" Sip-Cseq = "18236" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 192.168.2.55 Processing the preacct section of radiusd.conf modcall: entering group preacct for request 5 modcall[preacct]: module "preprocess" returns noop for request 5 rlm_realm: Looking up realm "192.168.2.55" for User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> " rlm_realm: No such realm "192.168.2.55" modcall[preacct]: module "suffix" returns noop for request 5 modcall: leaving group preacct (returns noop) for request 5 Processing the accounting section of radiusd.conf modcall: entering group accounting for request 5 rlm_acct_unique: Hashing 'NAS-Port = 5060,Client-IP-Address = 192.168.2.55,NAS-IP-Address = 192.168.2.55,Acct-Session-Id = "893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>",User-Name = "666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> "' rlm_acct_unique: Acct-Unique-Session-ID = "5488f64bd8d99c57". modcall[accounting]: module "acct_unique" returns ok for request 5 radius_xlat: '/usr/local/var/log/radius/radacct/192.168.2.55/detail-20060606' rlm_detail: /usr/local/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands to /usr/local/var/log/radius/radacct/192.168.2.55/detail-20060606 modcall[accounting]: module "detail" returns ok for request 5 modcall[accounting]: module "unix" returns ok for request 5 radius_xlat: '/usr/local/var/log/radius/radutmp' radius_xlat: '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' rlm_radutmp: Logout for NAS radiusdtest port 5060, but no Login record modcall[accounting]: module "radutmp" returns ok for request 5 radius_xlat: '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' rlm_sql (sql): sql_set_user escaped user --> '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ' radius_xlat: 'UPDATE radacct SET AcctStopTime = '2006-06-06 18:34:11', AcctSessionTime = '', AcctInputOctets = '', AcctOutputOctets = '', AcctTerminateCause = '', AcctStopDelay = '0', ConnectInfo_stop = '' WHERE AcctSessionId = '893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>' AND UserName = '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>' AND NASIPAddress = '192.168.2.55'' rlm_sql (sql): Reserving sql socket id: 0 radius_xlat: 'stop packet with zero session length. [user '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', nas '192.168.2.55']' rlm_sql (sql) in sql_accounting: stop packet with zero session length. [user '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', nas '192.168.2.55'] rlm_sql (sql): Released sql socket id: 0 radius_xlat: 'INSERT into radacct (AcctSessionId, AcctUniqueId, UserName, Realm, NASIPAddress, NASPortId, NASPortType, AcctStartTime, AcctStopTime, AcctSessionTime, AcctAuthentic, ConnectInfo_start, ConnectInfo_stop, AcctInputOctets, AcctOutputOctets, CalledStationId, CallingStationId, AcctTerminateCause, ServiceType, FramedProtocol, FramedIPAddress, AcctStartDelay, AcctStopDelay) values('893D5817-07A7-40D2-B0A8-987237749015(a)192.168.2.20<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ', '5488f64bd8d99c57', '666(a)192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', '', '192.168.2.55', '5060', '', DATE_SUB('2006-06-06 18:34:11', INTERVAL (0 + 0) SECOND), '2006-06-06 18:34:11', '', '', '', '', '', '', 'sip:555@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…> ', 'sip:666@192.168.2.55<https://email.secureserver.net/view.php?folder=INBOX.Sent_Items&uid=70#Comp…>', '', 'Sip-Session', '', '', '0', '0')' rlm_sql (sql): Released sql socket id: 0 -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

2 1

RE: [Users] openserctl error.
by Edson 09 Jun '06

09 Jun '06

First rule on scripting debug: use 'echo'.. ;) Try to include, just before line 26, a command like 'echo "[$1+$2*256+$3*65536+$4*16777216]" > ~/DEBUG' Run it again and take a look on the contends of '~/DEBUG' I hope that You would be able to find Your way out.. ;) Edson. _____ From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Kelvin Williams Sent: sexta-feira, 9 de junho de 2006 08:32 To: 'ram'; users(a)openser.org Subject: RE: [Users] openserctl error. # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } _____ From: ram [mailto:talk2ram@gmail.com] Sent: Friday, June 09, 2006 1:33 AM To: Kelvin Williams Subject: Re: [Users] openserctl error. post ur config file to assist ram On 6/9/06, Kelvin Williams <kelvin-lists(a)williamschadwell.com> wrote: This is a new OpenSER installation on a Solaris 10 box. Any time we call openserctl we receive the following error: /usr/local/sbin/openserctl: syntax error at line 1026: `GW_IP=$' unexpected Line 26 of openserctl reads: GW_IP=$(($1+$2*256+$3*65536+$4*16777216)) Any advice would be greatly appreciated! Thanks, kw _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Serusers] Problem in starting SER
by sumit dengri 09 Jun '06

09 Jun '06

sir when I start SER using "serctl start ser" command it showed following error message: Starting SER : cat: /var/run/ser.pid: No such file or directory started pid() Also there is no entry of ser in /etc/init.d PLZ help me out. Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

4 3

[Serusers] RE: novice issue
by Shyamsundar, Purkayastha (Purkayastha)** CTR ** 09 Jun '06

09 Jun '06

I fixed the problem which was basically a path issue but now it is cribbing for mysql.so The path is correct but when I run "./ser -c" I get the following output ./ser -c 0(4472) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory How do I sort it out?? thanks Shyam > -----Original Message----- > From: Shyamsundar, Purkayastha (Purkayastha)** CTR ** > Sent: Thursday, June 08, 2006 4:48 PM > To: 'serusers(a)lists.iptel.org' > Subject: novice issue > > Hi all > > Just started using ser but i am stuck at the first step itself > > i downloaded ser-0.9.6_linux_i386.tar > > untared it , set the cfg file and modules in correct path as per cfg file > and ran ./ser from the sbin directory > > and it says > > ERROR in cfg file ( 24 errors) > > if i comment out the loadmodule statements on the cfg files then the > errors reduce but inspite of verifying the correct path several times its > not getting through. > That means its reading the cfg file correctly but not loading the modules > or not able to interpret loadmodule command maybe > > Please help me > > thanks in advance > Shyam > >

4 6

RE: [Users] openserctl error.
by Kelvin Williams 09 Jun '06

09 Jun '06

# # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } _____ From: ram [mailto:talk2ram@gmail.com] Sent: Friday, June 09, 2006 1:33 AM To: Kelvin Williams Subject: Re: [Users] openserctl error. post ur config file to assist ram On 6/9/06, Kelvin Williams <kelvin-lists(a)williamschadwell.com> wrote: This is a new OpenSER installation on a Solaris 10 box. Any time we call openserctl we receive the following error: /usr/local/sbin/openserctl: syntax error at line 1026: `GW_IP=$' unexpected Line 26 of openserctl reads: GW_IP=$(($1+$2*256+$3*65536+$4*16777216)) Any advice would be greatly appreciated! Thanks, kw _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Serusers] ser.cfg Operators and Operands
by Szabolcs Szasz 09 Jun '06

09 Jun '06

Hi All! What is the correct operand in a condtional statement for negation? I have a problem in a ser.cfg line: if (uri=~"sip:0[1-9][0-9]*@.*" && uri!="sip:01234[6-9][0-9]@.*") The SER doesn't take in consideration the 2nd part of the conditional statement after the && operand. What is wrong in this line? Thanks! Szasz Szabolcs -- This message was scanned for spam and viruses by BitDefender. For more information please visit http://linux.bitdefender.com/

2 1

RE: [Serusers] mysql configuration issue
by İlker Aktuna (Koç.net) 09 Jun '06

09 Jun '06

Hi, No, it's not leftover file or anything else. Actually I think new CVS version has some problems with usrloc and domain database support. Because I have the same problem when I use usrloc with db support on new source. As a solution I used the old binary usrloc module with new ser. Regards, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Greger V. Teigre Sent: Friday, June 09, 2006 8:36 AM To: Benjamin.George(a)t-systems.com Cc: serusers(a)iptel.org Subject: Re: [Serusers] mysql configuration issue Good to hear. You probably had a leftover so file or something, making the db inconsistent with the binaries... g-) Benjamin.George(a)t-systems.com wrote: Hi, We tried in that way also ...but didn't work! So we reinstalled the SER binary once again (debian packages including mysql module) after that it starts working properly. Thanks! Regards, Benjamin. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, June 09, 2006 1:52 AM To: George, Benjamin Cc: serusers(a)iptel.org Subject: Re: [Serusers] mysql configuration issue :-) I just assumed you had run ser_mysql.sh and that maybe the error was a result of a connection error. But you're probably right. The value for register in the version needs to match what this version of SER register module requires. If you rename you ser DB and let ser_mysql.sh reinitialize it, you will have the reference. g-) Benjamin.George(a)t-systems.com wrote: Hi, I feel it is not a linking issue. This is something related to table version of table 'subscriber'/'domain'/'location', I think so. So it will be great if some body can throw some light in to this issue. Here go the error messages once again: [ stateless - initializing root@ubuntu:/usr/local/etc/ser # 0(0) Maxfwd module- initializing textops - initializing 0(0) register_udomain(): Invalid table version (use ser_mysql.sh reinstall) 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration ] My SER version is 0.9.x And MySql version is 4.1.x OS: Ubuntu linux Thanks and regards, Benjamin. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Thursday, June 01, 2006 11:40 AM To: George, Benjamin Cc: serusers(a)iptel.org Subject: Re: [Serusers] mysql configuration issue Have you linked against the right libraries? This is for 0.9.x on CentOS 4: [greger@siprouter modules]$ ldd mysql.so libmysqlclient.so.14 => /usr/lib/mysql/libmysqlclient.so.14 (0x00111000) libz.so.1 => /usr/lib/libz.so.1 (0x002aa000) libc.so.6 => /lib/tls/libc.so.6 (0x00c94000) libcrypt.so.1 => /lib/libcrypt.so.1 (0x006a6000) libnsl.so.1 => /lib/libnsl.so.1 (0x00376000) libm.so.6 => /lib/tls/libm.so.6 (0x00530000) libssl.so.4 => /lib/libssl.so.4 (0x00c0a000) libcrypto.so.4 => /lib/libcrypto.so.4 (0x00e29000) /lib/ld-linux.so.2 (0x00987000) libgssapi_krb5.so.2 => /usr/lib/libgssapi_krb5.so.2 (0x00baa000) libkrb5.so.3 => /usr/lib/libkrb5.so.3 (0x0022a000) libcom_err.so.2 => /lib/libcom_err.so.2 (0x00faa000) libk5crypto.so.3 => /usr/lib/libk5crypto.so.3 (0x003fc000) libresolv.so.2 => /lib/libresolv.so.2 (0x00a5e000) libdl.so.2 => /lib/libdl.so.2 (0x0028f000) [greger@siprouter modules]$ rpm -q mysql mysql-4.1.12-3.RHEL4.1 Benjamin.George(a)t-systems.com wrote: Hi All, Thanks Adam for providing the link. But as mentioned by zhangshuai, I am also getting the same error after doing those procedures. Error messages are pasted below: stateless - initializing 0(0) Maxfwd module- initializing textops - initializing root@ubuntu:/usr/local/sbin # 0(0) register_udomain(): Invalid table version (use ser_mysql.sh reinstall) 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration It will be great somebody gives some pointer to resolve this issue. Regards, Benjamin. -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of zhangshuai Sent: Thursday, June 01, 2006 6:57 AM To: serusers Subject: Re: Re: [Serusers] mysql configuration issue Dear All, I use mysql-4.1.15 and encountered the same error. After I did as the webpage said, I got another error: "invalid table version". I compiled the source code from cvs: cvs -d :pserver:anonymous@cvs.berlios.de:/cvsroot/ser checkout sip_router Is it the latest stable version? Thanks for your answers. Benjamin.George(a)t-systems.com wrote: *root@ubuntu:/usr/local/sbin # 0(0) new_connection(): Client does not support authentication protocol requested by server; consider upgrading MySQL client* http://dev.mysql.com/doc/refman/5.0/en/old-client.html Regards Adam _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers = = = = = = = = = = = = = = = = = = = = _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers <http://387555.sigclick.mailinfo.com/sigclick/010B0604/04044901/04054D03/823…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

1 0

[Serusers] rtpproxy only for calling local extension
by Nhadie 09 Jun '06

09 Jun '06

Hi How can I do it in such a way maultiple phones behind a single NAT gateway can call each other using rtpproxy and if these phone calls the PSTN it will not use rtpproxy, i tried removing the if (isflagset(6)) route(1); on the these part if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@1.2.3.4")) { but can't hear anything on the phones, but if i put if (isflagset(6)) route(1) back it will work. How can I do this? Thank You Regards, Nhadie ________________________________________________ Message sent using UebiMiau 2.7

2 1

RE: [Serusers] mysql configuration issue
by Benjamin.George＠t-systems.com 09 Jun '06

09 Jun '06

Hi, We tried in that way also ...but didn't work! So we reinstalled the SER binary once again (debian packages including mysql module) after that it starts working properly. Thanks! Regards, Benjamin. _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, June 09, 2006 1:52 AM To: George, Benjamin Cc: serusers(a)iptel.org Subject: Re: [Serusers] mysql configuration issue :-) I just assumed you had run ser_mysql.sh and that maybe the error was a result of a connection error. But you're probably right. The value for register in the version needs to match what this version of SER register module requires. If you rename you ser DB and let ser_mysql.sh reinitialize it, you will have the reference. g-) Benjamin.George(a)t-systems.com <mailto:Benjamin.George@t-systems.com> wrote: Hi, I feel it is not a linking issue. This is something related to table version of table 'subscriber'/'domain'/'location', I think so. So it will be great if some body can throw some light in to this issue. Here go the error messages once again: [ stateless - initializing root@ubuntu:/usr/local/etc/ser # 0(0) Maxfwd module- initializing textops - initializing 0(0) register_udomain(): Invalid table version (use ser_mysql.sh reinstall) 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration ] My SER version is 0.9.x And MySql version is 4.1.x OS: Ubuntu linux Thanks and regards, Benjamin. _____ From: Greger V. Teigre [mailto:greger@teigre.com <mailto:greger@teigre.com> ] Sent: Thursday, June 01, 2006 11:40 AM To: George, Benjamin Cc: serusers(a)iptel.org <mailto:serusers@iptel.org> Subject: Re: [Serusers] mysql configuration issue Have you linked against the right libraries? This is for 0.9.x on CentOS 4: [greger@siprouter modules]$ ldd mysql.so libmysqlclient.so.14 => /usr/lib/mysql/libmysqlclient.so.14 (0x00111000) libz.so.1 => /usr/lib/libz.so.1 (0x002aa000) libc.so.6 => /lib/tls/libc.so.6 (0x00c94000) libcrypt.so.1 => /lib/libcrypt.so.1 (0x006a6000) libnsl.so.1 => /lib/libnsl.so.1 (0x00376000) libm.so.6 => /lib/tls/libm.so.6 (0x00530000) libssl.so.4 => /lib/libssl.so.4 (0x00c0a000) libcrypto.so.4 => /lib/libcrypto.so.4 (0x00e29000) /lib/ld-linux.so.2 (0x00987000) libgssapi_krb5.so.2 => /usr/lib/libgssapi_krb5.so.2 (0x00baa000) libkrb5.so.3 => /usr/lib/libkrb5.so.3 (0x0022a000) libcom_err.so.2 => /lib/libcom_err.so.2 (0x00faa000) libk5crypto.so.3 => /usr/lib/libk5crypto.so.3 (0x003fc000) libresolv.so.2 => /lib/libresolv.so.2 (0x00a5e000) libdl.so.2 => /lib/libdl.so.2 (0x0028f000) [greger@siprouter modules]$ rpm -q mysql mysql-4.1.12-3.RHEL4.1 Benjamin.George(a)t-systems.com <mailto:Benjamin.George@t-systems.com> wrote: Hi All, Thanks Adam for providing the link. But as mentioned by zhangshuai, I am also getting the same error after doing those procedures. Error messages are pasted below: stateless - initializing 0(0) Maxfwd module- initializing textops - initializing root@ubuntu:/usr/local/sbin # 0(0) register_udomain(): Invalid table version (use ser_mysql.sh reinstall) 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration It will be great somebody gives some pointer to resolve this issue. Regards, Benjamin. -----Original Message----- From: serusers-bounces(a)lists.iptel.org <mailto:serusers-bounces@lists.iptel.org> [mailto:serusers-bounces@lists.iptel.org <mailto:serusers-bounces@lists.iptel.org> ] On Behalf Of zhangshuai Sent: Thursday, June 01, 2006 6:57 AM To: serusers Subject: Re: Re: [Serusers] mysql configuration issue Dear All, I use mysql-4.1.15 and encountered the same error. After I did as the webpage said, I got another error: "invalid table version". I compiled the source code from cvs: cvs -d :pserver:anonymous@cvs.berlios.de:/cvsroot/ser checkout sip_router Is it the latest stable version? Thanks for your answers. Benjamin.George(a)t-systems.com <mailto:Benjamin.George@t-systems.com> wrote: *root@ubuntu:/usr/local/sbin # 0(0) new_connection(): Client does not support authentication protocol requested by server; consider upgrading MySQL client* http://dev.mysql.com/doc/refman/5.0/en/old-client.html <http://dev.mysql.com/doc/refman/5.0/en/old-client.html> Regards Adam _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers> = = = = = = = = = = = = = = = = = = = = _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers>

2 1

[Serusers] failure_route and onreply_route question
by Ryan Pagquil 09 Jun '06

09 Jun '06

Hi list, I have a question about failure_route and onreply_route. Is there a rule which comes first appear on ser.cfg? I have this experience that my onreply_route is located before my failure_route and whenever I call t_on_failure nothing happens. I decided to move failure_route before onreply_route and it falls to the failure_route when I call t_on_failure. Anybody could clarify this? Thanks, Ryan

1 0

Re: [Users] ACC module detect-direction
by Glenn Dalgliesh 09 Jun '06

09 Jun '06

thanks this seems to have fixed it.... --------------------- Sent with ChatterEmail True push email for the Treo Smartphone www.chatteremail.com -----Original Message----- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Date: Wednesday, Jun 7, 2006 9:39 pm Subject: Re: [Users] ACC module detect-direction Hi Glenn, there was a bug which is now fixed on the CVS. Please update and let me know if it properly works now. regards, bogdan Glenn Dalgliesh wrote: >Ok, checked record-router and it is present with ftag but from to revesed . What debug output should I look at next. --------------------- Sent with ChatterEmail True push email for the Treo Smartphone www.chatteremail.com > -----Original Message----- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Date: Monday, Jun 5, 2006 8:19 pm Subject: Re: [Users] ACC module detect-direction >Hi Glenn, >For this feature to work, you need to be sure that you do record_route() on the INVITE and that "append_fromtag" RR module param is not disable. >regards, bogdan >Glenn Dalgliesh wrote: > >>modparam("acc", "detect_direction", 1) in CVS headI tried this option > > thinking that it would cause radius records to maintain the same values for username,to,from as the initial INVITE. ie. if the number you called hangup and not you the would still reflect your username and the called partyIs this correct? and if so this might be a bug this it seems to have no effect...Thanks > >>_______________________________________________ > > Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Users] openserctl error.
by Kelvin Williams 08 Jun '06

08 Jun '06

This is a new OpenSER installation on a Solaris 10 box. Any time we call openserctl we receive the following error: /usr/local/sbin/openserctl: syntax error at line 1026: `GW_IP=$' unexpected Line 26 of openserctl reads: GW_IP=$(($1+$2*256+$3*65536+$4*16777216)) Any advice would be greatly appreciated! Thanks, kw

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[Serusers] Caller-ID & Remote-Party-ID example
by Kostas Marneris 08 Jun '06

08 Jun '06

Hello all, I'm trying to get 'caller-id with Remote-Party-ID' work, but I think I'm missing something.. What is the "saved value of the SIP URI received from radius" ? (according to auth/README file). What is supposed to be the RADIUS attribute/value in RADIUS user profile ? Can someone send an example ? thank you.

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RE: [Serusers] mysql configuration issue
by Benjamin.George＠t-systems.com 08 Jun '06

08 Jun '06

Hi, I feel it is not a linking issue. This is something related to table version of table 'subscriber'/'domain'/'location', I think so. So it will be great if some body can throw some light in to this issue. Here go the error messages once again: [ stateless - initializing root@ubuntu:/usr/local/etc/ser # 0(0) Maxfwd module- initializing textops - initializing 0(0) register_udomain(): Invalid table version (use ser_mysql.sh reinstall) 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration ] My SER version is 0.9.x And MySql version is 4.1.x OS: Ubuntu linux Thanks and regards, Benjamin. _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Thursday, June 01, 2006 11:40 AM To: George, Benjamin Cc: serusers(a)iptel.org Subject: Re: [Serusers] mysql configuration issue Have you linked against the right libraries? This is for 0.9.x on CentOS 4: [greger@siprouter modules]$ ldd mysql.so libmysqlclient.so.14 => /usr/lib/mysql/libmysqlclient.so.14 (0x00111000) libz.so.1 => /usr/lib/libz.so.1 (0x002aa000) libc.so.6 => /lib/tls/libc.so.6 (0x00c94000) libcrypt.so.1 => /lib/libcrypt.so.1 (0x006a6000) libnsl.so.1 => /lib/libnsl.so.1 (0x00376000) libm.so.6 => /lib/tls/libm.so.6 (0x00530000) libssl.so.4 => /lib/libssl.so.4 (0x00c0a000) libcrypto.so.4 => /lib/libcrypto.so.4 (0x00e29000) /lib/ld-linux.so.2 (0x00987000) libgssapi_krb5.so.2 => /usr/lib/libgssapi_krb5.so.2 (0x00baa000) libkrb5.so.3 => /usr/lib/libkrb5.so.3 (0x0022a000) libcom_err.so.2 => /lib/libcom_err.so.2 (0x00faa000) libk5crypto.so.3 => /usr/lib/libk5crypto.so.3 (0x003fc000) libresolv.so.2 => /lib/libresolv.so.2 (0x00a5e000) libdl.so.2 => /lib/libdl.so.2 (0x0028f000) [greger@siprouter modules]$ rpm -q mysql mysql-4.1.12-3.RHEL4.1 Benjamin.George(a)t-systems.com <mailto:Benjamin.George@t-systems.com> wrote: Hi All, Thanks Adam for providing the link. But as mentioned by zhangshuai, I am also getting the same error after doing those procedures. Error messages are pasted below: stateless - initializing 0(0) Maxfwd module- initializing textops - initializing root@ubuntu:/usr/local/sbin # 0(0) register_udomain(): Invalid table version (use ser_mysql.sh reinstall) 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration It will be great somebody gives some pointer to resolve this issue. Regards, Benjamin. -----Original Message----- From: serusers-bounces(a)lists.iptel.org <mailto:serusers-bounces@lists.iptel.org> [mailto:serusers-bounces@lists.iptel.org <mailto:serusers-bounces@lists.iptel.org> ] On Behalf Of zhangshuai Sent: Thursday, June 01, 2006 6:57 AM To: serusers Subject: Re: Re: [Serusers] mysql configuration issue Dear All, I use mysql-4.1.15 and encountered the same error. After I did as the webpage said, I got another error: "invalid table version". I compiled the source code from cvs: cvs -d :pserver:anonymous@cvs.berlios.de:/cvsroot/ser checkout sip_router Is it the latest stable version? Thanks for your answers. Benjamin.George(a)t-systems.com <mailto:Benjamin.George@t-systems.com> wrote: *root@ubuntu:/usr/local/sbin # 0(0) new_connection(): Client does not support authentication protocol requested by server; consider upgrading MySQL client* http://dev.mysql.com/doc/refman/5.0/en/old-client.html <http://dev.mysql.com/doc/refman/5.0/en/old-client.html> Regards Adam _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers> = = = = = = = = = = = = = = = = = = = = _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers>

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[Users] RE: [Devel] Memory Leak in latest cvs version?
by Douglas Garstang 08 Jun '06

08 Jun '06

Bogdan, Below is the first part of openser.cfg. As you can see, I am not loading a lot of modules. This system has 4GB of RAM. The errors are still occurring. I did have more modules, and removed them, and the problem still persists. How can I get that memory dump? Sending the process a USR1 has no effect. # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/dispatcher.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" #loadmodule "/usr/local/lib/openser/modules/uri_db.so" > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Thursday, June 08, 2006 3:20 AM > To: Douglas Garstang > Cc: Users(a)openser.org; devel(a)openser.org > Subject: Re: [Devel] Memory Leak in latest cvs version? > > > Douglas, > > in this case, it seams you have a module that loads a lot of stuff at > startup in shared memory. Like USRLOC for location and > aliases, LCR for > rule...... > > is it your case? > > regards, > bogdan > > > Douglas Garstang wrote: > > >Thanks Bogdan. > > > >A few things of note. Firstly, I'm using the default memory > size, and secondly, the system has processed hardly any > calls, IF ANY. Given these two confiditions, it would seem > that problem is abnormal, not normal. > > > >Douglas. > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Wednesday, June 07, 2006 9:23 AM > >>To: Douglas Garstang > >>Cc: Users(a)openser.org; devel(a)openser.org > >>Subject: Re: [Devel] Memory Leak in latest cvs version? > >> > >> > >>Hi Douglas, > >> > >>I just put together a wiki page about debugging memory > >>problems. Please see: > >> http://openser.org/dokuwiki/doku.php?id=memory > >> > >>and follow the instructions. > >> > >>regards, > >>bogdan > >> > >> > >>Douglas Garstang wrote: > >> > >> > >> > >>>I'm wondering if there's a memory leak or something in the > >>> > >>> > >>latest cvs version of openser. > >> > >> > >>>Only been running it for a while, processing very few calls, > >>> > >>> > >>and this was appearing... > >> > >> > >

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RE: [Users] Sucess Stories of openser+asterisk+voicemail
by Ricardo Martinez 08 Jun '06

08 Jun '06

Hello. Yes, i made some test a few weeks ago, and works great. Take a look at : http:// www.onsip.org <http://www.onsip.org> . BTW, is there any news about the issue 5a for the onsip manual? Thanks! Ricardo Martinez.- -----Mensaje original----- De: ram [mailto:talk2ram@gmail.com] Enviado el: Jueves, 08 de Junio de 2006 11:10 Para: users(a)openser.org Asunto: [Users] Sucess Stories of openser+asterisk+voicemail Hi does any one have success stories of above said suject i been trying from past 7days, but not able to success could some one succeeded the same ram

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[Users] Bug in branch handling?
by Federico Giannici 08 Jun '06

08 Jun '06

I'm using OpenSER 1.0.0 under OpenBSD 3.9-stable amd64. An user of our SIP server have a VoIP PBX called "voispeed". Their calls always hung up after 20 seconds because they say that our SIP server doesn't correctly handle the branch parameter. They say that RFC3261 specifies that ACK messages for a 2xx reply MUST use a different "branch" parameter (contrary to ACK messages to non 2xx replies). But it seems that OpenSER doesn't recognize these ACKs with different branch parameter and keeps sending the 200 reply. Can somebody confirm this bug in OpenSER? Ant solution? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Users] Sucess Stories of openser+asterisk+voicemail
by ram 08 Jun '06

08 Jun '06

Hi does any one have success stories of above said suject i been trying from past 7days, but not able to success could some one succeeded the same ram

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RE: [Serusers] SER with Heartbeat in two node cluster
by Barta János 08 Jun '06

08 Jun '06

Hello! You have to make some change in the conf.file of Node1 & Node2: 1, In the ser.cfg: ------------------------- alias="clusterIP" 2, In the /etc/heartbeat/haresources: --------------------------------------------------- cluster1 clusterIP/mask/eth1/broadcast IPsrcaddr::clusterIP ser By Jani -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Stephen Helfen Sent: Friday, June 02, 2006 1:31 PM To: serusers(a)iptel.org Subject: [Serusers] SER with Heartbeat in two node cluster All, I am trying to implement a two-node cluster using heartbeat on two SIP SER proxies running on Linux. I am having a bit of trouble as it appears the two proxies are sharing messages rather than acting as an active and standby. Has anyone succesfully implemented this type of configuration and/or does anyone know a good reference for enabling this configuration? Thanks in advance. Steve _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________________________________________ Hogy el ne tévedjen! A navigáció még soha nem volt ennyire egyszerű! PDA-k széles választékban - DigitalMarket http://ad.adverticum.net/b/cl,1,6022,99786,162268/click.prm

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RE: [Serusers] Modifying "From" field in INVITE message
by İlker Aktuna (Koç.net) 08 Jun '06

08 Jun '06

Hi, Ok; I found it. Thanks for your quick reply. Regards, ilker -----Original Message----- From: Alexandr Dubovikov [mailto:shurik@start4.info] Sent: Thursday, June 08, 2006 5:22 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Modifying "From" field in INVITE message On Thu, Jun 08, 2006 at 05:05:01PM +0300, ?lker Aktuna (Ko?.net) wrote: > Thanks for your reply. How can I learn more on the subject ? take the README file from modules/textops. It'll help you. > > For example I want to replace the user part of uri with last 8 digits. Sorry, I don't understand what you mean in "last 8 digits" ? > > Thanks, > ilker > Wbr, -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info <http://387555.sigclick.mailinfo.com/sigclick/070C0004/03074C08/010C4B03/144…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] OpenSER 1.0.x Main Index
by Edson 08 Jun '06

08 Jun '06

Hi, everybody... Just to let the list know that there are 4 new links on the main page of the OpenSER Dokuwiki (http://openser.org/dokuwiki/doku.php?id=start). Each one points to Index Pages: - Parameters Index (http://openser.org/dokuwiki/doku.php?id=index_of_openser_1.0.x_parameters); - Functions Index (http://openser.org/dokuwiki/doku.php?id=index_of_openser_1.0.x_functions); - Examples Index (http://openser.org/dokuwiki/doku.php?id=index_of_openser_1.0.x_examples); and - FIFO Commands Index (http://openser.org/dokuwiki/doku.php?id=index_of_openser_1.0.x_fifo_command s). All entries point to the original Module's README (where the parm/function, etc was mentioned/defined). So You will find many repeated entries, but this is intentional, since many modules use/export the same parameter/function names. As suggested by Gerry, on the side of each link/entry is another link that drives You to a page where user comments can be written. Have fun, and help complement/correct it. Edson.

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RE: [Serusers] Modifying "From" field in INVITE message
by İlker Aktuna (Koç.net) 08 Jun '06

08 Jun '06

Thanks for your reply. How can I learn more on the subject ? For example I want to replace the user part of uri with last 8 digits. Thanks, ilker -----Original Message----- From: Alexandr Dubovikov [mailto:shurik@start4.info] Sent: Thursday, June 08, 2006 4:08 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Modifying "From" field in INVITE message On Thu, Jun 08, 2006 at 03:51:02PM +0300, ?lker Aktuna (Ko?.net) wrote: > Hi, Hi, > > I need to modify From field of INVITE messages. > Is that possible ? How ? > sample: if(method=="INVITE") { subst('/^(From|f):(.*)sip:.*@/\1:\2sip:2222/g'); } -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info <http://387555.sigclick.mailinfo.com/sigclick/02090501/0C044E00/04074E06/911…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] openser with nathelper/rtpproxy as SBC
by Ovidiu Sas 08 Jun '06

08 Jun '06

Hi all, I'm trying to get openser to act as an SBC using the mediaproxy module and the rtpproxy (built from the cvs). openser is configured to run in a multihomed mode. I'm starting rtpproxy in debug mode: rtpproxy -f -l 10.110.1.174/10.111.5.15 The rtpproxy is binding to both IPs. When an INVITE is received on the 10.110 interface, I call force_rtp_proxy(). Regrdless of the "i" or "e" parameter that I pass to the force_rtp_proxy(), the rtpproxy is replying with the same IP. output from rtpproxy if force_rtp_proxy("i"): received command "UI 4a2fabe1-24c7fe03(a)10.110.50.4 10.110.48.7 10046 60e4ff4a-soma;1" new session 4a2fabe1-24c7fe03(a)10.110.50.4, tag 60e4ff4a-soma;1 requested, type strong new session on a port 35000 created, tag 60e4ff4a-soma;1 pre-filling caller's address with 10.110.48.7:10046 sending reply "35000 10.110.1.174 " output from rtpproxy if force_rtp_proxy("e"): received command "UE 44709fe7-423307f6(a)10.110.49.134 10.110.48.7 10012 1f713156-soma;1" new session 44709fe7-423307f6(a)10.110.49.134, tag 1f713156-soma;1 requested, type strong new session on a port 35002 created, tag 1f713156-soma;1 pre-filling caller's address with 10.110.48.7:10012 sending reply "35002 10.110.1.174 " The IP address sent out in the outgoing INVITE is 10.110.1.174 instead of 10.111.5.15. Is this the expected behaviour from the rtpproxy? Do I need to pass other params to the force_rtp_proxy() function? Regards, -ovi

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[Serusers] Modifying "From" field in INVITE message
by İlker Aktuna (Koç.net) 08 Jun '06

08 Jun '06

Hi, I need to modify From field of INVITE messages. Is that possible ? How ? The username is too long because of some database issues; but the outgoing caller id (from uri) should be last 8 digits of the username. How can I do that ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/060A060D/0C074E08/00064D0A/152…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] SIP/2.0 484 Address Incomplete
by ram 08 Jun '06

08 Jun '06

Hi all I have downloaded from openser and iam trying to integrate voice mail with asterisk I have read all the docs in the document site after config, and people recomendation iam able to run the openser successfully and able to fix the problem calling out side but when the local user not available, iam sending to asterisk voice mail and i get error SIP/2.0 484 Address Incomplete in my x-lite client any suggestions ram

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RE: [Serusers] novice issue
by Benjamin.George＠t-systems.com 08 Jun '06

08 Jun '06

Hi, Check library (.so files) paths in ser.cfg file and if not correct, give the correct path for each library. Then run ser again, it will definitely work. Regards, Benjamin. -----Original Message----- From: Alexandr Dubovikov [mailto:shurik@start4.info] Sent: Thursday, June 08, 2006 5:14 PM To: Shyamsundar, Purkayastha (Purkayastha)** CTR ** Cc: 'serusers(a)lists.iptel.org' Subject: Re: [Serusers] novice issue On Thu, Jun 08, 2006 at 04:48:24PM +0530, Shyamsundar, Purkayastha (Purkayastha)** CTR ** wrote: > Hi all > > Just started using ser but i am stuck at the first step itself > > i downloaded ser-0.9.6_linux_i386.tar > > untared it , set the cfg file and modules in correct path as per cfg file > and ran ./ser from the sbin directory > > and it says > > ERROR in cfg file ( 24 errors) at first, check you configuration with "ser -c" and set the debug level to 4 in your ser.cfg , after it you can see where is a mistake. Wbr, -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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