sr-users June 2006

sr-users@lists.kamailio.org

203 participants
495 discussions

Re: [Serusers] How to start a VOIP business ?

by Roberto Pereyra

Argentine roberto 2006/6/21, ram <talk2ram(a)gmail.com>: > > Hi > > where are you located ? > > > ram > > > > On 6/21/06, Roberto Pereyra <pereyra.roberto(a)gmail.com> wrote: > > > > Subcription to any International whole sale VOIP > > carrier > > > > > > > > Please, somebody knows any to try ?. > > roberto pereyra > > 2006/6/21, Greger V. Teigre <greger(a)teigre.com>: > > > > There was an old SNMP module way back, but nobody wanted to maintain > it... > > It is most definitely a need. > > I know people use various monitoring tools like nagios. > > For version 0.10.x (CVS head), the new XMLRPC interface can be used for > > monitoring also, but it's more of a pull than push... > > g-) > > > > > > Benjamin.George(a)t-systems.com wrote: > > > > > > > > > > Hi, > > > > > > > > I doubt there are some monitoring/management tools already available or > > packed with SER. I am not very sure, whether SER components support SNMP > or > > not. But you can develop your own application by considering which are all > > the parameters (for example: Bandwidth usage, BHCA, call logs, etc) has to > > be monitored in a communication environment apart from normal system > > parameters like CPU usage, Memory consumption, Threads, I/O, etc. > > > > > > > > Normally what happens is, whoever supplies the network components provide > > the performance management/monitoring tools or adapters/api to integrate > > with other industry leading tools or support for SNMP. > > > > > > > > Or otherwise, you can use any good products which are available in the > > market (but they are not et-al freeware! L L). > > > > Some of tools are: > > > > BMC Patrol > > HP Open view > > CA Unicenter > > Mercury > > ProactiveNet > > > > Etc… > > > > > > > > If you have any idea about any free tools which can be used to monitor SER > > environment, please do let me know that. I am also curious to know/get > such > > a tool. > > > > > > > > Regards, > > > > Benjamin. > > > > ________________________________ > > > > > > From: ram [mailto:talk2ram@gmail.com] > > Sent: Tuesday, June 20, 2006 2:26 PM > > To: George, Benjamin > > Subject: Re: [Serusers] How to start a VOIP business ? > > > > > > > > > > Hi > > > > > > > > > > > > can you tell me where can get below said tools ? > > > > > > > > > > > > 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 > > > > > > > > > > > > ram > > > > > > ________________________________ > > > > _______________________________________________ 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 > > > > > > > > > -- > Ing. Roberto Pereyra > ContenidosOnline > Looking for Linux Virtual Private Servers ? Click here: > http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426&a_bid=56 > _______________________________________________ > > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -- Ing. Roberto Pereyra ContenidosOnline Looking for Linux Virtual Private Servers ? Click here: http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426&a_bid=56

18 years, 4 months

RE: [Serusers] How to start a VOIP business ?

by Benjamin.George＠t-systems.com

Hi, I doubt there are some monitoring/management tools already available or packed with SER. I am not very sure, whether SER components support SNMP or not. But you can develop your own application by considering which are all the parameters (for example: Bandwidth usage, BHCA, call logs, etc) has to be monitored in a communication environment apart from normal system parameters like CPU usage, Memory consumption, Threads, I/O, etc. Normally what happens is, whoever supplies the network components provide the performance management/monitoring tools or adapters/api to integrate with other industry leading tools or support for SNMP. Or otherwise, you can use any good products which are available in the market (but they are not et-al freeware! :-( :-(). Some of tools are: 1. BMC Patrol 2. HP Open view 3. CA Unicenter 4. Mercury 5. ProactiveNet Etc... If you have any idea about any free tools which can be used to monitor SER environment, please do let me know that. I am also curious to know/get such a tool. Regards, Benjamin. _____ From: ram [mailto:talk2ram@gmail.com] Sent: Tuesday, June 20, 2006 2:26 PM To: George, Benjamin Subject: Re: [Serusers] How to start a VOIP business ? Hi can you tell me where can get below said tools ? 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 ram

18 years, 4 months

[Users] RTPPRXY or MEDIA Proxy

by ram

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

18 years, 4 months

[Serusers] (no subject)

by Aman Sachdeva

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

18 years, 4 months

[Serusers] could not rev. resolve IP......

by Shenen Shenen

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

18 years, 4 months

RE: [Serusers] checking source ip and dest ip

by İlker Aktuna (Koç.net)

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. 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18 years, 4 months

RE: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723

by Benjamin.George＠t-systems.com

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.

18 years, 4 months

[Serusers] How to start a VOIP business ?

by Concy Pereira

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 > > ------------------------------

18 years, 4 months

RE: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723

by İlker Aktuna (Koç.net)

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. _____________________________________________________________________________________________________________________________________________

18 years, 4 months

[Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723

by ravi reddy

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.

18 years, 4 months

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sr-users June 2006