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[Serusers] Mediaproxy in public IP e two client behind same Nat
by pistara_bros＠tin.it 20 Sep '07

20 Sep '07

Hi, I want add others information about my test, providing also the log of MediaProxy WEB GUI. Windows Client used : X-Lite Version 3.0 build 29712 First scenarious : Client_A and Client_B running on Windows XP in the same LAN behind same NAT. Client_A call Client_B and During the conversation MediaProxy WEB GUI (http://smile.ing.uniroma1. it/serweb/mediaproxy/) show me following dump: Call | Caller |Called |Via address |status |Duration|.... ------------------------------|---------------- |---------------------------|-------------------| F:j2me@151. 100.122.144 | - | - |151.100.122.144:60012|inactive| 0'59'' |.... To:enzo@151.100.122.144| - | - | | | | And after 60 seconds the session expired and MediaProxy WEB GUI don't show it. Result: the RTP packet dont' pass through the Mediaproxy, but the packet go directly from one computer to another of my Lan. Second scenarious: Windows Client used : X-Lite Version 3.0 build 29712 Linux Client used : X-Lite Version 2.0 build 1105d Client_A runnning on WindowsXP and Client_B running on Linux Fedora in the same LAN behind same NAT. Client_A call Client_B and During the conversation MediaProxy WEB GUI (http://smile.ing.uniroma1.it/serweb/mediaproxy/) show me following dump: Call | Caller |Called --------------------------------|-------------------------- |--------------------------- F:j2me@151.100.122.144 | 85.39.206.210: 62969 |85.39.206.210:62947 To: enzo(a)151.100.122.144 | | |Via address |status |Duration|.... |151.100.122.144:60010 |active | 1'25'' |.... Result: the RTP packet pass through the Mediaproxy, that acts as "man in the middle" anf forward RTP packet versus clients of my Lan. Why MediaProxy, in windowz -> windowz client communications doesn't work????? Orazio >Hi to All, >I'm investigating again about Mediaproxy and SER , but in >this test i'm using the >the testbed of smile.ing.uniroma1.it whic use >SER and Mediaproxy in a public ip 151.100.122.144 >I'm using Client_A (X- >lite) running on WindowsXP 192.168.123.10 >and Client_B (X-lite) running >on WindowsXP 192.168.123.168 >My Lan use a router with a public ip >83.211.41.253 >So the clients are behind the same Nat. >Using ethereal i >see that during the conversation the RTP packet dont' pass through the >Mediaproxy in public ip (151.100.122.144) but the packet go directly >from one computer to another of my Lan. >So the question is : >what SER >architecture (with Mediaproxy or with SEMS or other..) i have to use in >order to >have a proxy RTP that work as man in the middle and manage all >RTP Traffic between the two clients? >Please help me. >thanks.

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[OpenSER-Users] Free BSD 6.0 and SEAS problem.
by Kuddusi 20 Sep '07

20 Sep '07

Hi, I am using default seas configuration. When I tried to start openser seas module can not be loaded althoug module is compiled without problem. I tried 1.2.0 and 1.2.2 and the same problem occurred. It seems that it could not find "dprintf" function. How can I solve this problem? root@~: openser -ddddddddddd 0(43499) loading module /usr/local/lib/openser/modules/tm.so 0(43499) DEBUG: register_pv: tm 0(43499) xl_add_extra_spec: extra items list is not initialized 0(43499) xl_add_extra_spec: inserting extra item [T_branch_idx] at [0] 0(43499) xl_add_extra_spec: inserting extra item [T_reply_code] at [1] 0(43499) loading module /usr/local/lib/openser/modules/seas.so 0(43499) ERROR: load_module: could not open module </usr/local/lib/openser/modules/seas.so>: /usr/local/lib/openser/modules/seas.so: Undefined symbol "dprintf" and config is below Thanks, Kuddusi CIFTCIBASI. # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode fork=no log_stderror=yes # reply_to_via=1 listen = tcp:84.51.32.24:5060 listen = udp:84.51.32.24:5060 port=5060 # ------------------ module loading ---------------------------------- #set module path #mpath="/usr/local/lib/openser/modules/" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/seas.so" modparam("seas", "listen_sockets","84.51.32.24:5080") modparam("seas", "stats_socket","84.51.32.24:5088") #modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if(!as_relay_t("app_server_one")){ t_reply("500","No App Server"); } }

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[OpenSER-Users] Do we like the same books?
by voipexpert 20 Sep '07

20 Sep '07

I just joined Shelfari to connect with other book lovers. Come see the books I love and see if we have any in common. Then pick my next book so I can keep on reading. Click below to join my group of friends on Shelfari! http://www.shelfari.com/Register.aspx?ActivityId=16229378&InvitationCode=84… voipexpert Shelfari is a free site that lets you share book ratings and reviews with friends and meet people who have similar tastes in books. It also lets you build an online bookshelf, join book clubs, and get good book recommendations from friends. You should check it out. -------- You have received this email because voipexpert (sunkara.raviprakash.feb14(a)gmail.com) directly invited you to join his/her community on Shelfari. It is against Shelfari's policies to invite people who you don't know directly. Follow this link (http://www.shelfari.com/actions/emailoptout.aspx?email=Users@openser.org&ac…) to prevent future invitations to this address. If you believe you do not know this person, you may view (http://www.shelfari.com/voipexpert) his/her Shelfari page or report him/her in our feedback (http://www.shelfari.com/Feedback.aspx) section. Shelfari, 616 1st Ave #300, Seattle, WA 98104

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[Serusers] Creating a test scenario to test NATing..
by Piyush.Bansal＠relianceada.com 20 Sep '07

20 Sep '07

hi all, I wanna simulate a test scenario in which the sip clients are behind NAT (whether same or different NAT, in some private network), and the SER and MediaProxy are running on some public IP. My question is, in this case the client is not able to communicate with the server through the Firewall / NAT. My SIP messages are being dropped at the Firewall itself. Can anybody tell me, what could be the reason behind that????? I'm totally stuck at this point..... plz revert asap... thanx in advance.. The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

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[OpenSER-Users] openser + freeradius + cdrtool in freebsd
by Sam Wun 20 Sep '07

20 Sep '07

Hi, I found openser and freeradius in the freebsd ports tree, but the cdrtools is a tool for CD ISO in freebsd. Is this the right cdrtools for freesradius? In the CDRtools website, it said: http://download.dns-hosting.info/CDRTool/ "CDRTool provides mediation, accounting and tracing for Call Detail Records generated by OpenSER by using RADIUS protocol and OpenSER siptrace facility. " It seems that the CDrtools in freebsd ports tree is not the one for freeradius. Thanks W

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Re: [OpenSER-Users] OpenSER with MySQL Cluster
by Christian Schlatter 19 Sep '07

19 Sep '07

Kerker Staffan wrote: > hi > ok, thanks for the update. i kind of dropped that > track as i had all those errors. > > with your experience, is that a better solution than > using the t_replicate? i'm only synchronizing two servers. Well, there is a trade-off between the complexity of a mysql cluster deployment and the possible inconsistencies introduced by t_replicate. With mysql cluster, you can always be sure that both of your proxies have the same, consistent view of the registration state. This comes at the prize of deploying and maintaining the db cluster. t_replicate is a lightweight replication mechanism that doesn't ensure registration state consistency. I like the simplicity of this approach. Personally, I prefer having a consistent registration state. I also think that the mysql guys have done a great job in making the cluster deployment and management as easy as possible. /Christian > > best regards, > /staffan > > >> -----Original Message----- >> From: Christian Schlatter [mailto:cs@unc.edu] >> Sent: den 19 september 2007 15:42 >> To: Kerker Staffan >> Cc: Juha Heinanen; users(a)openser.org >> Subject: Re: [OpenSER-Users] OpenSER with MySQL Cluster >> >> >> Kerker Staffan wrote: >>> hi >>> i'm curious about this one. we used to run a dual Openser >> installation >>> with MySQL cluster, but this proved to very unstable (on >> the cluster >>> side). >>> is this working fine now? >>> >>> for instance, if one of the mysql cluster nodes was >> disconnected, this >>> could lead to complete cluster shut down, and nodes didn't handle >>> reconnect very good. >>> >>> so, we actually switched to the register replication instead... >> We're running a mysql 5.0.22 cluster for our two redundant >> openser proxies. This setup works very stable for about a year now. >> >> In the beginning our cluster was shutting down itself quite >> regularly e.g. after a network failure. Since then we >> >> - are running the management nodes on hosts that are >> connected to the data nodes through different network paths >> (this is essential, a cluster node needs to be able to solve >> the splitbrain problem) >> >> - have added 'StopOnError = 0' to the data node settings >> (this causes a data nodes to try to reconnect after it lost >> communication to its neighbors, per default, a data node just >> shuts itself down) >> >> - added enough DataMemory and IndexMemory for the data nodes >> (DataMemory = 1024M, IndexMemory = 64M) >> >> >> Overall I'm quite pleased with our mysql cluster solution, >> although I'd >> prefer a solution where the UAs do register with both of our >> proxies, so >> that there is no need to replicate registration state between >> the proxies. >> >> Our mysql cluster basically only contains the 'location' >> table since we >> store all account data on redundant LDAP/H350 directories. >> >> >> /Christian >> >> >> >>> best regards >>> /Staffan >>> >>> >>>> -----Original Message----- >>>> From: users-bounces(a)openser.org >>>> [mailto:users-bounces@openser.org] On Behalf Of Christian Schlatter >>>> Sent: den 18 september 2007 23:37 >>>> To: Juha Heinanen >>>> Cc: users(a)openser.org >>>> Subject: Re: [OpenSER-Users] OpenSER with MySQL Cluster >>>> >>>> >>>> Juha Heinanen wrote: >>>> ... >>>>> unfortunately you then cannot support presence. using >>>> another proxy >>>>> for presence is not a solution either, because same users >> need both >>>>> presence and other sip methods. >>>> What exactly prevents me from using a dedicated presence >>>> server? And why can't I support presence with db_mode=3? >>>> >>>> E.g. if I just want to deploy a simple PA using the PRESENCE >>>> module, I don't see any dependencies on USRLOC. >>>> >>>> I'm not sure if PUA_USRLOC depends on db_mode<3, but at least >>>> the documentation doesn't mention it. And it looks to me like >>>> I could use >>>> pua_set_publish() from PUA_USRLOC to send a PUBLISH to my >>>> dedicated presence server ;-) ... I haven't tried that though. >>>> >>>>> by the way, looks like presence related requests are by >> far the most >>>>> common ones if all sip UAs have presence turned on. it means that >>>>> presence would also create biggest db activity. >>>> Yes, I know, SIP presence is very chatty. >>>> >>>> >>>> /Christian >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>> >>

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[Serusers] Mediaproxy in public IP e two client behind same Nat
by pistara_bros＠tin.it 19 Sep '07

19 Sep '07

Hi to All, I'm investigating again about Mediaproxy and SER , but in this test i'm using the the testbed of smile.ing.uniroma1.it whic use SER and Mediaproxy in a public ip 151.100.122.144 I'm using Client_A (X- lite) running on WindowsXP 192.168.123.10 and Client_B (X-lite) running on WindowsXP 192.168.123.168 My Lan use a router with a public ip 83.211.41.253 So the clients are behind the same Nat. Using ethereal i see that during the conversation the RTP packet dont' pass through the Mediaproxy in public ip (151.100.122.144) but the packet go directly from one computer to another of my Lan. So the question is : what SER architecture (with Mediaproxy or with SEMS or other..) i have to use in order to have a proxy RTP that work as man in the middle and manage all RTP Traffic between the two clients? Please help me. thanks.

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[OpenSER-Users] UAC module with Asterisk
by Halomoan Chow 19 Sep '07

19 Sep '07

Dear All Now I want to try UAC module to solve the proxy authentication to Asterisk. Until now it still fail even the username/password and realm are already correct 100% Then I found this statement "For sending authentication credentials to the termination provider openser has the "uac" module. But be aware that this module does not always works as it does not increase the cseq when sending the INVITE with credentials (search the archive for details). E.g. the uac module wont work if the termination providers uses Asterisk (as Asterisk checks the cseq)" Actually why UAC module has this limitation and why this does not get solved until now? Thanks Regards Halomoan Zhou

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[Serusers] MeadiaProxy + SER + X-lite in the same Lan.
by pistara_bros＠tin.it 19 Sep '07

19 Sep '07

I want correct my previous posted sentences : "My guess is mediaproxy makes a test on the IPs (they are 192.168.147.2 and 192.168.147.3) and refuses to proxy the call." with "My guess is mediaproxy makes a test on the IPs (they are 192.168.123.168 and 192.168.123.10) and refuses to proxy the call." Does Anyone encounter this problem? could you share your opinion, with me?. Thanks. Orazio

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[OpenSER-Users] memory allocation issue
by Benko 19 Sep '07

19 Sep '07

Hello! I wanted to upgrade a server from 1.1.0 to 1.2.2 today but despite extensive testing beforehand i have some trouble: After starting with the new openser-binary, it didn't allow any registrations and answered with "500 server error occured". Logging said: Sep 18 17:23:12 [/sbin/openser] ERROR: sip_msg_cloner: cannot allocate \ memory_ Sep 18 17:23:12 [/sbin/openser] ERROR: new_t: out of mem:_ Sep 18 17:23:12 [/sbin/openser] ERROR: t_newtran: new_t failed_Sep 18 and 17:23:12 [/sbin/openser] ERROR: sip_msg_cloner: cannot allocate memory_ Sep 18 17:23:12 [/sbin/openser] ERROR: store_reply: can't alloc' clone \ memory_ after restarting it with "-m 128"(according to the faq) these errors are gone, but it still doesn't accept registrations and prints: Sep 18 17:24:16 [/sbin/openser] ERROR:tm:print_uac_request: no shmem_ Sep 18 17:24:16 [/sbin/openser] ERROR:tm:t_forward_nonack: failure to \ add branches_ Sep 18 17:24:16 [/sbin/openser] ERROR:tm:w_t_relay: t_forward_nonack \ failed_ [update] - after several restarts, the first error-messages appeared again. When i run the new version on port 5061(old production server 1.1.0 is running parallel on port 5060) i can register with the new server(on port 5061) without problems. I have the vague impression that the new instance can't free memory, but why does it work when i run it on a different port? Any clues? Regards Christian

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[OpenSER-Users] OpenSER with MySQL Cluster
by Vasile Zaharia 19 Sep '07

19 Sep '07

Hi *, Normally, OpenSER should work just fine with a MySQL Cluster, but I would like to know if anyone has tried this configuration yet. Are there any configuration settings I should be aware of? Thanks, Sica. ____________________________________________________________________________________ Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. http://mobile.yahoo.com/go?refer=1GNXIC

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[OpenSER-Users] Configuration of Permissions module
by Sergio Gutierrez 19 Sep '07

19 Sep '07

Hi everybody. Does anybody has any example about how to configure the permissions module to use address validation from database configuration tables? In our setup, we observe an inconsistent behaviour when defining subnets; fifo command subnet_dump does not show the expected subnet. The only case wich works well is wehn mask is set to 32. Thanks in advance for your attention. Sergio Gutierrez EPM Telecomunicaciones Medellin - Colombia

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[OpenSER-Users] example web_im send_im.html/php within openser 1.2.1 doensn't work
by CHEN XUEQIN 19 Sep '07

19 Sep '07

Hi all: I put example script file send_im.html/send_im.php in example/web_im direcotry into apache webroot. Then I launch firefox, fill in SIP Address and Message, click to send. But firefox don't response. The status bar indicated wait something. The fact is: 1. openser load mi_fifo.so correctly 2. the fifo file /tmp/openser_fifo, with the permission 0666, everyone can read and write 3. LOG message " DEBUG:mi_fifo:mi_fifo_server: entered consume", missing log "**** done consume" 4. SIP Address was user registered on openser. From Header was another user who registered on openser. 5. when I restart openser, replace fifo cmd with ":get_statistics:", send_im.php works Anyone know why? How to send message via fifo, pelease let me know Thanks in advance. Regards, Chen Xueqin

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Re: [OpenSER-Users] OpenSER with MySQL Cluster
by Helge Waastad 19 Sep '07

19 Sep '07

Hi My experiences with MySQL clusters is that you have to expand the index memory etc as far as you dear to enable adding new tables/columns etc. I guess yopu would not like to do that in a production evnvironment anyway... br hw

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[OpenSER-Users] user meeting (beer drinking in Vienna)
by Klaus Darilion 19 Sep '07

19 Sep '07

Hi! I proudly announce the first ser/openser/asterisk beer drinking evening in Vienna. When: Thursday (thirsty day) 20. September 2007, 19:00 CEST Where: Vienna, a bar in an inner district - exact location to be announced If you want to join please send a short reply so that I know for how many people I have to arrange a table. cu Klaus

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[Serusers] help needed in configuring ENUM with ENUM module
by geethanjali.csg＠wipro.com 19 Sep '07

19 Sep '07

Hi, I am trying to configure ser to use enum module. But I am getting the following error. 0(6192) get_record: lookup(5.6.7.8.e164.arpa, 35) failed 0(6192) enum_query(): No NAPTR record found for 5.6.7.8.e164.arpa. Can someone kindly give me some pointers to fix this error. I am trying this from past few days. Thanks, Geethanjali C S Wipro Technologies 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

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[OpenSER-Users] mediaproxy work for muti domains
by liu york 18 Sep '07

18 Sep '07

Dear all on my mind，mediaproxy can work for muti sip domains， like it's feature list―― -Ability to have different proxies for different SIP domains - Ability to use multiple proxy servers per SIP domain Now，using multiple proxy servers per SIP domain is ok for me，but i cant make it work with different SIP domains.Can somebody tell me how to do? mediaproxy.ini like below start = yes socket = /var/run/proxydispatcher.sock group = openser defaultProxy = domain://sip1.gteling.com the zone file in my dns like below mp1 IN A 64.77.236.132 mp2 IN A 210.51.163.165 sip1 IN A 64.77.236.132 sip2 IN A 210.51.163.165 _mediaproxy._tcp IN SRV 0 100 25060 mp1 how can i use use different medip proxy server for different sip domains. for example i want to use media mp1.gteling.com servise for sip domain sip1.gteling.com. many appreciation. 2007-09-19 york

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Re: [OpenSER-Users] WeSIP with openser 1.2.0 again
by Iñaki Baz Castillo 18 Sep '07

18 Sep '07

El Martes, 18 de Septiembre de 2007, Gregorio escribió: > Hi Elias et all, Please Gregorio, por future new questions to the list use the "New message" button in your Microsoft Office Outlook instead of replying to any other mail in the list and deleting body and subject. The mail you have sent has been created by pressing "Reply" over a mail with subject "Per branch timer and CANCEL on non final reply", so your mail appears in the same thread because it has a header: In-Reply-To: <293670490709180215t1536dff5ycd01a8bded247cfa(a)mail.gmail.com> For people using thread capable mail clients this is very uncomfortable. Even, you can see how wrong it is by looking for your mail in the mail archive: http://openser.org/pipermail/users/2007-September/thread.html (note how your mail "WeSIP with openser 1.2.0 again" hangs from "Per branch timer and CANCEL on non final reply", being it really wrong). So please, for future **new** questions use "New message" and just use "Reply" when you really want to reply a previous mail. PD: Using HTML and blue text is not very appreciated by people reading lost of mail each day. Please, use plain text. Regards. -- Iñaki Baz Castillo

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[OpenSER-Users] OSPToolkit Warning Messages
by Edson 18 Sep '07

18 Sep '07

Hi... I'm trying to compile OSP support for eather 1.2.2 or TRUNK. I have OpenSSL 0.9.8d on a OpenSUSE 10.2. I was wondering if the warning that pops are something that desire attention, since are related to the X509 part: ospopenssl.c: In function âbio_dump_cbâ: ospopenssl.c:597: warning: format â%08Xâ expects type âunsigned intâ, but argument 3 has type âstruct BIO *â ospopenssl.c:597: warning: format â%08lXâ expects type âlong unsigned intâ, but argument 4 has type âconst char *â ospopenssl.c:597: warning: format â%Xâ expects type âunsigned intâ, but argument 7 has type âlong intâ ospopenssl.c:604: warning: format â%08Xâ expects type âunsigned intâ, but argument 3 has type âstruct BIO *â ospopenssl.c:604: warning: format â%08lXâ expects type âlong unsigned intâ, but argument 4 has type âconst char *â ospopenssl.c:604: warning: format â%Xâ expects type âunsigned intâ, but argument 7 has type âlong intâ ospopenssl.c: In function âOSPPSSLLoadCertsâ: ospopenssl.c:659: warning: passing argument 2 of âd2i_X509â from incompatible pointer type ospopenssl.c:689: warning: passing argument 2 of âd2i_X509â from incompatible pointer type Any glues or nothing to worry about??? Edson

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[OpenSER-Users] snmpstats agent failing to connect to master agent
by Gabi Mahu 18 Sep '07

18 Sep '07

Hello all, I am currently trying to get the snmpstats module up and running, for monitoring purposes. I followed the instructions in the module documentation, I studied Net-SNMP references and I searched web resources for several days now, but still I couldn't succeed in starting the openser module properly. I have no idea what the problem could be. The NET-SNMP part of the equation seams to work properly: I am using the 5.4.1 version. I managed to play around with the tools from the package, in local and remote scenarios too. I was able to send and receive traps, get and set different values in the MIB tree, load new MIBs... On the openser part, I'm not doing something special. I have a config file that does nothing, except load the snmpstats module and set the limits for the dialogs and the unparsed data traps. That's it. I tried with fork on yes and on no, because I found out from my web searches that this could be a problem, but all I got was some logging in the no fork scenario, saying that the snmp agent from snmpstats could not connect to the master agent, the snmpd that runs on the same machine. The "check for a proper configuration" step in the module documentation failed also, before this (I got no output in the snmpd window after openser was started, until that point, everything was like in the documentation), so this couldn't work actually, but i still did it anyway. The problem must be in the configuration files, but it escapes me. This is the content of the files (yes, both in the same location...): snmpd.conf ----------------------------------- com2sec notConfigUser default public group notConfigGroup v1 notConfigUser group notConfigGroup v2c notConfigUser view systemview included .1.3.6.1.2.1.1 view systemview included .1.3.6.1.2.1.25.1.1 access notConfigGroup "" any noauth exact systemview none none syslocation localhost (edit /etc/snmp/snmpd.conf) syscontact Root <root@localhost> (configure /etc/snmp/snmp.local.conf) pass .1.3.6.1.4.1.4413.4.1 /usr/bin/ucd5820stat rocommunity public localhost trapsink localhost:162 public trap2sink localhost:162 public authtrapenable 1 master agentx ---------------------------------------------- snmpstats.conf ------------------------------- agentXSocket tcp:localhost:705 ---------------------------------------------- And the way I start snmpd now is: snmpd -f -Lo -M /usr/local/share/mibs -m ALL -d -x tcp:localhost:705 I did try several versions including the one found in the module documentation. This is the context of my problem. Hopping that there is a small, but obvious mistake, in my config files, due to my lack of attention or something, and not a more serious thing, I thank you for your time. Gabriel Mahu --------------------------------- Check out the hottest 2008 models today at Yahoo! Autos.

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Re: [OpenSER-Users] WeSIP with openser 1.2.0 again
by Elias Baixas 18 Sep '07

18 Sep '07

Hi Gregorio, can you send over the logs but with the 'debug' variable set to 7 ? I couldn't see the SIP messages in the logs you sent me. Elias Gregorio wrote: > > Hi Elias et all, > > > > I still get the parsing exception in the SEAS module when SIP MESSAGE > are sent from Openser to WeSIP. If I could know which header fail, > perhaps I could find a temporal solution and go on withw the > development, Elias can you tell me something about this? > > > > Thanks a lot for the help. > > Regards. > > Gregorio. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] OpenXCAP problem
by mika.saari＠wipsl.com 18 Sep '07

18 Sep '07

Hi, I have "successfully" installed OpenXCAP with latest OpenSER SVN. The OpenXCAP server manages to contact the SQL server. When I am using EyeBeam 1.5 with setting "http://openxcap.mydomain.com:8000/xcap-root/" I get next error to the syslog. Sep 18 14:17:21 openxcap openxcap[9902]: [-] error: Invalid Document Selector /xcap-root/resource-lists/users/testacc/buddies-resource-list.xml (the Document Selector context must be 'users' or 'global') What kind of misconfiguration I have done ? Thanks a lot, -Mika

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[OpenSER-Users] Per branch timer and CANCEL on non final reply
by Madalina Pitis 18 Sep '07

18 Sep '07

Hello, I have the following scenario: I receive an INVITE and do parallel forking, having two branches. What I would like to do then is set a timer for the first branch, and send a CANCEL on that branch in one of the following cases: 1. no final response is received on the first branch and the timer expires. In this case it is important that the timer is per branch, so not both branches expire, and not both are canceled. 2. a 181 Call is being forwarded reply is received on the second branch. Any suggestions are welcomed, because right now I have absolutely no idea how I could solve this. I read the documentation for the TM module, all kinds of timers, but they are all per transaction, not per branch, or at least so I understood. (I am using OpenSER 1.2) In conclusion, my questions are: 1. Can I set a per branch timer? (either from code or configuration script) 2. How can I send a CANCEL only on that branch when the timer expires? 3. How can I send a CANCEL on one branch when the other has received a non final reply, 181 in this case. (the problem is the actual generation and sending of the CANCEL message, not intercepting the 181 reply or finding out on which branch it was received). Thank you in advance, Madalina

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[OpenSER-Users] Openser with b2bua[query on media]
by Kaeman Chris 17 Sep '07

17 Sep '07

Hello, I have few questions before i proceed to get the integration of openser with the b2bua; I am confused that i need to use the mediaproxy module of the openser if i am taking up the media from the b2bua as i need the b2bua as openser is not performing topology hiding. Cheers, KChris --------------------------------- Try the revolutionary next-gen Yahoo! Mail. Click here.

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[OpenSER-Users] [CDRTool] Call limit per user/domain and fromuser/authuser
by Iñaki Baz Castillo 17 Sep '07

17 Sep '07

Hi, I've started reading CDRTool documentation (very complete and long). Let me two basic questions about it: 1) Does CDRTool allow limiting simultaneous calls from/to a user/domain? I mean something as "fixed number of lines" per client. 2) I plan to use this scenario: sip-users <-> OpenSer <-> Asterisk B2BUA <-> CDRTool <-> SIP_to_PSTN provider So the sip-user will just authenticate against OpenSer, which will route call to Asterisk, and Asterisk as B2BUA will authenticate against CDRTool, so the "from_user" will be set by Asterisk but the "caller ID" will be the sip-user's original caller ID. Does CDRTool log and show the caller ID (not just the fromuser) in the web report? Or maybe it's not necessary to set the "from_user" in Asterisk? does CDRTool allow digest auth different of the "from_user"? does CDRTool require "check_from()"? Thanks a lot for any help. Best regards. -- Iñaki Baz Castillo

1 0

[OpenSER-Users] Searchable Mailing List Archives
by Norman Brandinger 17 Sep '07

17 Sep '07

Everyone: I've set up a mirror of the OpenSER mailing lists on Nabble. One of the main reasons for this was to obtain search capabilities. If you're interested in this, please take a look at: http://www.nabble.com/OpenSER---the-Open-Source-SIP-Server-f26014.html Their home page is http://www.nabble.com Regards, Norm p.s I would like to thank the folks at Nabble for importing the historical archives into their systems.

2 1

RE: [OpenSER-Users] Openser and Voice master
by Dan 17 Sep '07

17 Sep '07

Hi Ram, I had tried with many providers having Voice Master but have not been able to work with any of them. People have tried opening username and password account, only prefix based and only IP based authentication. But none have worked, VM keep rejecting calls. As far as I understand VM checks IP Address/username in contact field as well, which I am unable to change. I can change username and or IP Address in from field using uac_replace_from but could not find anything to change the same in contact field of invite message. Only Scenarios where I am able to send calls via VM are: 1.) I have both the endpoint and openser's IP Address to be opened at VM. ( In case I have IP only authentication account at VM end) 2.) I send prefix from the endpoint end itself and not added at openser's level. (In case I have prefix only authentication account at VM end). But in both scenario's openser becomes useless and can easily be bypassed by client as VM will allow calls directly from ENDPOINT as well. Regards, Dan _____ From: ram [mailto:talk2ram@gmail.com] Sent: Monday, September 17, 2007 9:46 PM To: Dan Subject: Re: [OpenSER-Users] Openser and Voice master how are u trying should not be problem as for i know ram On 9/11/07, Dan <fiedler.dan(a)gmail.com> wrote: Hi, Has anybody able to successfully able to route calls to voicemaster from openser. I tried all method but none worked. From username/password to IP based authentication to prefix based authentication. Regards Dan _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Serusers] MeadiaProxy + SER + X-lite in the same Lan.
by pistara_bros＠tin.it 17 Sep '07

17 Sep '07

Hi to all, I'm newbie using SER and need your help. This is my testbed: SER running on 192.168.123.186 MediaProxy running on 192.168.123.186 Client_A X-lite running under WinowsXP on 192.168.123.168 Client_B X-lite running under WinowsXP on 192.168.123.10 I want to route RTP packets between 2 computers in the same network through Mediaproxy. I know this may have no sense, but it's just a test configuration. The call use_media_proxy() seems to have no effect, the RTP packets go directly from one computer to another. My guess is mediaproxy makes a test on the IPs (they are 192.168.147.2 and 192.168.147.3) and refuses to proxy the call. Is mediaproxy working just between different networks or it's an error in my configuration file? But if i use this Enviroment: SER running on 192.168.123.186 MediaProxy running on 192.168.123.186 Client_A X-lite running under Linux Fedora on 192.168.123.168 Client_B X-lite running under WinowsXP on 192.168.123.10 The call use_media_proxy() correctly and the RTP packets pass through the mediaproxy, infacty using tools session.py I see the session active and traffic packet amount. Why, with client under Windows, Media proxy doesn't work correctly? Any suggestions, please? Thanks in advance. Orazio

1 0

[Serusers] SER + RADIUS authentication example?
by Mike Trest - Personal 17 Sep '07

17 Sep '07

Hi, I have build my SER 2.0.0 with radius client to leverage existing authentication application. Can someone point me to any HOWTO use radius clent call authentication? Or, can some thoughtful person share their own ser.cfg example? Thanks ..mike.. --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000771-1, 09/01/2007 Tested on: 9/7/2007 9:44:06 AM avast! - copyright (c) 1988-2007 ALWIL Software. http://www.avast.com

2 1

[Serusers] UA With Presence Capabilities
by Abdul Qadir 17 Sep '07

17 Sep '07

Hi, I want to try SER 2.0.0 RC 1 with presence capabilities. Can any one tell me some clients with presence support which I can configure and test with SER presence module. Thanking you in advance. Best Regards, Abdul Qadir --------------------------------- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.

2 1

[Serusers] Communication Problem btw SER and User Agent
by Adeyeye O. Michael 17 Sep '07

17 Sep '07

Hi, Here is an indication that my SER is running: micadeyeye@asmicom:/usr/src/Research-Progress$ ser Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 196.24.224.76 [196.24.224.76]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 196.24.224.76 [196.24.224.76]:5060 Aliases: tcp: asmicom.local:5060 tcp: localhost:5060 udp: asmicom.local:5060 udp: localhost:5060 --------------------------------------------------------------------------------------------------- But each time I tried connecting to it from a USER AGENT (ZAP, UCTIMSCLIENT), it doesn't interact with the client. I checked my dump(via wireshack) but only REGISTER messages are sent with no reply from SER. Can anyone help me? I have both SER and client sitting on same PC (using same port-5060).

2 1

[Serusers] TLS problem ser 2_0_0 - ERROR: tls_init.c:366: Unable to set the memory allocation functions
by Tomasz Zieleniewski 17 Sep '07

17 Sep '07

Hi, I did everything according to the tls module description when I start SER I get the following error: What do I have to do to make things work? Thanks in advance. Tomasz Aug 24 18:42:39 sen ser[26840]: tls: _init_tls_h: compiled with openssl version "OpenSSL 0.9.8c 05 Sep 2006" (0x0090803f), kerberos support: off, compression: on Aug 24 18:42:39 sen ser[26840]: tls: init_tls_h: installed openssl library version "OpenSSL 0.9.8c 05 Sep 2006" (0x0090803f), kerberos support: off, zlib compression: on compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -m64 -DL_ENDIAN -DTERMIO -O3 -Wa,--noexecstack -g -Wall -DMD32_REG_T=int -DMD5_ASM Aug 24 18:42:39 sen ser[26840]: ERROR: tls_init.c:366: Unable to set the memory allocation functions Aug 24 18:42:39 sen ser[26840]: could not initialize tls, exiting...

2 1

[Serusers] Loadbalancing for interco
by inge 17 Sep '07

17 Sep '07

Hi all, Is there a way to have something like a loadbalancing on SER for outgoing calls ? I want to distribute the calls between two gateways. Ideally, with a coefficient (ie. 60% for the first and 40% for the second). Thanks for your support. Adrien .L

5 10

[OpenSER-Users] Quick question about RTPproxy and Mediaproxy
by Arya 17 Sep '07

17 Sep '07

Hello for those of you that use openser as a SIP proxy/registrar do you also have RTPproxy or Mediaproxy configured? -- Thank You

2 1

[OpenSER-Users] SIP web dialer
by Schumann Sebastian 17 Sep '07

17 Sep '07

Hello all, I am currently trying to make my openser Registrar/Presence solution work with several clients. Can anyone recommend web applets which are capable of SIP (at least to register/unregister and make calls) based on Java or (not prefered but also possible) ActiveX? Furthermore, some functins to support SIMPLE/XCAP acc. RFC would be perfect. I searched a lot but no satisfying solution was found yet. OpenSource is preferred of course, but other solutions are possible. My thesis requires besides the local application testing also this opportunity :-\ Thanks a lot for your help! Regards Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 2 588 13332 (tel) +421 910 643010 (mobile) +49 175 1925928 (mobile) sebastian.schumann(a)t-com.sk http://www.t-com.sk <http://www.t-com.sk/>

1 0

[OpenSER-Users] snmpstats agent failing to connect to master agent
by Gabi Mahu 17 Sep '07

17 Sep '07

Hello all, I am currently trying to get the snmpstats module up and running, for monitoring purposes. I followed the instructions in the module documentation, I studied Net-SNMP references and I searched web resources for several days now, but still I couldn't succeed in starting the openser module properly. I have no idea what the problem could be. The NET-SNMP part of the equation seams to work properly: I am using the 5.4.1 version. I managed to play around with the tools from the package, in local and remote scenarios too. I was able to send and receive traps, get and set different values in the MIB tree, load new MIBs... On the openser part, I'm not doing something special. I have a config file that does nothing, except load the snmpstats module and set the limits for the dialogs and the unparsed data traps. That's it. I tried with fork on yes and on no, because I found out from my web searches that this could be a problem, but all I got was some logging in the no fork scenario, saying that the snmp agent from snmpstats could not connect to the master agent, the snmpd that runs on the same machine. The "check for a proper configuration" step in the module documentation failed also, before this (I got no output in the snmpd window after openser was started, until that point, everything was like in the documentation), so this couldn't work actually, but i still did it anyway. The problem must be in the configuration files, but it escapes me. This is the content of the files (yes, both in the same location...): snmpd.conf ----------------------------------- com2sec notConfigUser default public group notConfigGroup v1 notConfigUser group notConfigGroup v2c notConfigUser view systemview included .1.3.6.1.2.1.1 view systemview included .1.3.6.1.2.1.25.1.1 access notConfigGroup "" any noauth exact systemview none none syslocation localhost (edit /etc/snmp/snmpd.conf) syscontact Root <root@localhost> (configure /etc/snmp/snmp.local.conf) pass .1.3.6.1.4.1.4413.4.1 /usr/bin/ucd5820stat rocommunity public localhost trapsink localhost:162 public trap2sink localhost:162 public authtrapenable 1 master agentx ---------------------------------------------- snmpstats.conf ------------------------------- agentXSocket tcp:localhost:705 ---------------------------------------------- And the way I start snmpd now is: snmpd -f -Lo -M /usr/local/share/mibs -m ALL -d -x tcp:localhost:705 I did try several versions including the one found in the module documentation. This is the context of my problem. Hopping that there is a small, but obvious mistake, in my config files, due to my lack of attention or something, and not a more serious thing, I thank you for your time. Gabriel Mahu ____________________________________________________________________________________ Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7

1 0

[Serusers] SER File transfer
by Allan Chao ( 趙廷睿 ) 17 Sep '07

17 Sep '07

Dear All, How should i do if i want to transfer a "file " to my friend (just like msn) in SIP protocol , does there any spec(oma, 3gpp etc.) to metion it ? allan

1 0

[OpenSER-Users] Openser as Loadbalancer with NAT support example
by Tung Tran 16 Sep '07

16 Sep '07

Hi all, I am studying how to setup the Loadbancer with OpenSER and NAT support, If anyone has done it please share your configuration, Thanks so much Regards Tung

1 0

[OpenSER-Users] rtpproxy recording
by nhadie 16 Sep '07

16 Sep '07

Hi, All how can i run rtpproxy to save recorded conversation locally? i tried: /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media but got nothing under /var/media after making calls /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media -S /var/media also got nothing? has anyone tried it? thank you regards, nhadie

1 0

[OpenSER-Users] rtpproxy recording
by nhadie 16 Sep '07

16 Sep '07

Hi, All how can i run rtpproxy to save recorded conversation locally? i tried: /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media but got nothing under /var/media after making calls /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media -S /var/media also got nothing? has anyone tried it? thank you regards, nhadie

1 0

[OpenSER-Users] RE: Trunk Compile Error 2
by Edson 16 Sep '07

16 Sep '07

Just an update info: # make modules=modules/auth_radius modules;echo $? make: *** [modules] Error 2 2 # And yes, I need Radius support... Edson >-----Original Message----- >From: Edson [mailto:4lists@gmail.com] >Sent: sexta-feira, 14 de setembro de 2007 23:01 >To: 'users(a)openser.org' >Subject: Trunk Compile Error 2 > >I'm trying to compile the trunk version. > >After updating I run >make distclean >and >make exclude_modules="auth_diameter h350 ldap mi_xmlrpc osp > postgres unixodbc" modules install > >but after I always get: > >make[1]: Leaving directory >`/usr/local/src/VoIP/openser/openser/trunk/modules/auth_db' > >make: *** [modules] Error 2 > >What is this "Error 2" ? In this same machine I can compile, without a >glance, branche 1.2 with this same 'make' command. > >Thanks for any help.... > >Edson

3 7

[Serusers] Per-cent Sign in Attributes
by Martin Hoffmann 15 Sep '07

15 Sep '07

Salut, sorry for getting lazy now. In the config snipped | append_to_reply("%$digest_challenge"); what is the point of the per-cent sign in front of the attribute name? Thanks, Martin, who though he had figured all this out. Apparently not.

4 19

[OpenSER-Users] Problem with SRV lookups
by Tim Madorma 15 Sep '07

15 Sep '07

Hi, I have a simple application where I am trying to an SRV record to load balance towards a few different destination servers. I set up my DNS server to resolve to four different destination A record URIs - something like: _sip._udp.aimcidfilter.com 180 IN SRV 1 50 5061 crunch.com. _sip._udp.aimcidfilter.com 180 IN SRV 1 50 5061 spinner.com. _sip._udp.aimcidfilter.com 180 IN SRV 2 50 5061 homer.com. _sip._udp.aimcidfilter.com 180 IN SRV 2 50 5061 bart.com. In my configuration file, I am putting a domain name in my call to seturi as follows: seturi("sip:feature_fs@aimcidfilter.com:6010"); and then I call route to send the INVITE. I was expecting to see a NAPTR and SRV record attempts, but when I sniffed DNS, I only saw A record attempts. Any advice would be very helpful. thanks, Tim ------------------------------------ My configuration file looks as follows: # # TWC OpenSER SIP Proxy for Caller ID - openser_cid.cfg # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) maxbuffer=1048576 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #reply_to_via=1 children=4 log_facility=LOG_LOCAL4 dns_use_search_list=no dns_servers_no=2 # One for the ENUM server and one for other DNS lookups user="sipproxy" group="sipproxy" disable_dns_blacklist=yes # FOR TESTING ONLY dns_try_ipv6=no # FOR TESTING ONLY # Local IP Address and port to listen to # Set the following IP addresses to the value of the local e1000g0 interface listen=udp:65.185.232.62:5060 # Alias IP address/port pair values will be in the Req URI when the # OpenSER is being load balanced with a BIG IP server # Set the IP addresses to the value of the associated blade # of the BIG IP that is load balancing for the SIP Proxy server. alias=65.185.233.104:5060 # LOAD OpenSER MODULES mpath="/sw/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "mi_fifo.so" loadmodule "xlog.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- modparam("acc", "log_level", 2) modparam("acc", "log_flag", 1) modparam("acc","report_cancels", 1) modparam("acc","failed_transaction_flag", 1) modparam("acc","log_extra", "req_uri=$rU") modparam("tm", "fr_timer", 5) modparam("tm","fr_inv_timer",5) modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages within a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { route(1); }; if (method=="OPTIONS") { sl_send_reply("200", "OK"); exit; } if (uri =~ "^sip:feature_fs@" || method=="NOTIFY") { xlog("L_DBG", "TWC: received incoming message:\n <$mb>\n"); seturi("sip:feature_fs@aimcidfilter.com:6010"); setflag(1); route(1); } else { sl_send_reply("404", "Not Found"); }; } ##################################################### # Default Message Handler ##################################################### route[1] { # Send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

3 6

[Serusers] SER Implementation
by Tuhin 15 Sep '07

15 Sep '07

Hi, I am from Bangladesh. I have downloaded and configured SER for proxy purpose. My server is working fine in test mode. Now I need some more features using SER. I wants to implement CDR and credit control module with this. My OS is Redhat Ent 4. What are the applications I need to to do so, pls let me know. Rgds Tuhin

1 0

[OpenSER-Users] Trunk Compile Error 2
by Edson 15 Sep '07

15 Sep '07

I'm trying to compile the trunk version. After updating I run make distclean and make exclude_modules="auth_diameter h350 ldap mi_xmlrpc osp postgres unixodbc" modules install but after I always get: make[1]: Leaving directory `/usr/local/src/VoIP/openser/openser/trunk/modules/auth_db' make: *** [modules] Error 2 What is this "Error 2" ? In this same machine I can compile, without a glance, branche 1.2 with this same 'make' command. Thanks for any help.... Edson

1 0

[OpenSER-Users] registration and redirection
by Mike Clark 14 Sep '07

14 Sep '07

Ok, I am brand new to openser and don't really have ti routing anything yet. But I did have a dumb or at least naive question. Can openser be configured as a very simple load balancer and redirect registration requests, in a round robin manner, to a set of Asterisk servers? I know having openser be the registrar is the best solution in the long run, but this may solve an issue quickly. I looked at the dispatch module and I know it will do want I want for invites, but didn't know if registration requests could be redirected. Thanks, Mike Clark

1 0

[OpenSER-Users] memory error sems with openser
by laurent schweizer 14 Sep '07

14 Sep '07

Hello, I user openser with sems , and each 2 week a memory error. the problem is that I don't have more trafic that usualy. Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmUtils.cpp)(write_to_socket)(614): write to unix socket: completed Sep 13 13:39:46 dns2 ./openser[3673]: t_uac: short of cell shmem Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmEventQueue.cpp)(processEvents)(71): event processed Sep 13 13:39:46 dns2 ./openser[3673]: unix_server_loop: Command 't_uac_dlg' failed with return value -1 Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmSession.cpp)(run)(292): 2D9B7F95-46E9218200068CF4-97601B30(a)212.203.123.82 dlg.getUACTransPending() = 1 Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmServer.cpp)(run)(93): revents = 1 Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmCtrlInterface.cpp)(cacheMsg)(186): recv-ed: <500 Server error occurred (2/FIFO/UAC> Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmInterfaceHandler.cpp)(handleRequest)(187): response from Ser: 500 Server error occurred (2/FIFO/UAC Sep 13 13:39:46 dns2 Sems[1254]: Error: (AmInterfaceHandler.cpp)(handleRequest)(214): reply is missing headers: <500 Server error occurred (2/FIFO/UAC> Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmServer.cpp)(run)(93): revents = 0 Sep 13 13:40:15 dns2 ./openser[3794]: ERROR:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 408 Sep 13 13:40:19 dns2 Sems[1254]: Debug: (UserTimer.cpp)(checkTimers)(97): Timeout Event could not be posted, session does not exist any more. Sep 13 13:43:19 dns2 ./openser[3764]: ERROR: new_t: out of mem: Sep 13 13:43:19 dns2 ./openser[3764]: ERROR: t_newtran: new_t failed Sep 13 13:44:23 dns2 ./openser[3748]: ERROR: new_t: out of mem: Sep 13 13:44:23 dns2 ./openser[3748]: ERROR: t_newtran: new_t failed Sep 13 13:44:28 dns2 ./openser[3781]: ERROR: new_t: out of mem: Sep 13 13:44:28 dns2 ./openser[3781]: ERROR: t_newtran: new_t failed Sep 13 13:44:47 dns2 ./openser[3765]: ERROR: new_t: out of mem: Sep 13 13:44:47 dns2 ./openser[3765]: ERROR: t_newtran: new_t failed Sep 13 13:45:09 dns2 ./openser[3747]: ERROR: new_t: out of mem: Sep 13 13:45:09 dns2 ./openser[3747]: ERROR: t_newtran: new_t failed

4 3

[Serusers] SER+Mediaproxy + X-LITE in the same LAN.
by pistara_bros＠tin.it 14 Sep '07

14 Sep '07

Hi all, i'm newbie in SER, I want to route RTP packets between 2 computers with x-lite client voip in the same network through Mediaproxy. I know this may have no sense, but it's just a test configuration. The call use_media_proxy() seems to have no effect, the RTP packets go directly from one computer to another, no packet trough Mediaproxy. This is my environment: SER and Mediaproxy on the same PC (192.168.123.186) X-lite client_A running on 192.168.123.10 X-lite client_B running on 192.168.123.168 Client_A call Client_B that accept, but mediaproxy refuses to proxy the call. This is my ser.cfg # ----------- global configuration parameters ------------------------ debug=3 # debug level listen=192.168.123.186 port=5060 log_facility=LOG_LOCAL6 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser: heslo@localhost/ser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local1/lib/ser/modules/mysql.so" loadmodule "/usr/local1/lib/ser/modules/sl.so" loadmodule "/usr/local1/lib/ser/modules/tm.so" loadmodule "/usr/local1/lib/ser/modules/rr.so" loadmodule "/usr/local1/lib/ser/modules/maxfwd.so" loadmodule "/usr/local1/lib/ser/modules/usrloc.so" loadmodule "/usr/local1/lib/ser/modules/registrar.so" loadmodule "/usr/local1/lib/ser/modules/textops.so" loadmodule "/usr/local1/lib/ser/modules/uri.so" loadmodule "/usr/local1/lib/ser/modules/uri_db.so" loadmodule "/usr/local1/lib/ser/modules/domain.so" loadmodule "/usr/local1/lib/ser/modules/auth.so" loadmodule "/usr/local1/lib/ser/modules/auth_db.so" #MEDIA PROXY module needed loadmodule "/usr/local1/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local1/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- auth params -- 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") #SQL settings modparam("auth_db", "db_url", "mysql://ser: heslo@localhost/ser") modparam("usrloc", "db_url", "mysql://ser: heslo@localhost/ser") #MEDIAPROXY settings #--------------------------- # configurazione moduli #--------------------------- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy"," natping_interval", 30) modparam("mediaproxy","mediaproxy_socket"," /var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics"," /usr/local1/etc/ser/sip-clients") modparam("mediaproxy"," rtp_asymmetrics","/usr/local1/etc/ser/rtp-clients") modparam ("registrar", "nat_flag", 6) # -- 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"); break; }; if (msg:len >= 2048 ) { 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=="INVITE" && client_nat_test("3")) { # record_route_preset("192.168.123.186:5060;nat=yes"); #} else if (method!="REGISTER") { # record_route(); #}; if (method=="INVITE" ) { if (client_nat_test("3")) { record_route_preset("192.168.123.186:5060; nat=yes"); } else record_route_preset ("192.168.123.186"); } else if (method!="REGISTER") { record_route(); }; if (method==" BYE" || method=="CANCEL") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply ("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; if (!uri==myself) { route(4); route(1); break; }; # 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=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method==" REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; 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 t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact: [ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri! =myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (! isflagset(8)) { setflag(8); use_media_proxy(); }; }; } 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(); }; } And this is Mediaproxy Log: Listening for commands on local socket `/var/run/mediaproxy.sock' Listening for remote commands on `192. 168.123.186:25060' Remote commands are allowed from: 192.168.123.186, 127.0.0.1 Using IP address `192.168.123.186' for the RTP/RTCP proxy request B390Opu7cB 192.168.123.168:25500:audio 192.168.123.168 sip. antech.it local 192.168.123.186 remote unknown-agent info=from: orazio@sip.antech.it,to:j2me@sip.antech.it, fromtag:As94,totag: session B390Opu7cB: started. listening on 192.168.123.186:52010 execution time: 1.71 ms lookup B390Opu7cB 192.168.123.186:25500:audio 192.168.123.186 sip.antech.it local sip.antech.it unknown unknown-agent info=from:orazio@sip.antech.it,to:j2me@sip.antech.it,fr omtag:As94, totag:oePf execution time: 0.57 ms session B390Opu7cB: caller signed in from 192.168.123.168:25500 (RTP) (will return to 192.168.123.168: 25500) status execution time: 0.14 ms session B390Opu7cB: 2/0/0 packets, 344/0/0 bytes (caller/called/relayed) session B390Opu7cB: ended (did timeout). delete B390Opu7cB info= execution time: 0.14 ms and this is dump of session.py [root@sip mediaproxy]# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------------------------------- 192.168.123.168:25500 - 192.168.123.186:52010 - ?.?.?.?:? inactive 0'32" GSM Audio 344/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.9.0 Is mediaproxy working just between different networks or it's an error in my configuration file? Please help me. Any suggestions? Orazio

1 0

[Serusers] how to record conversation using rtpproxy
by nhadie 14 Sep '07

14 Sep '07

Hi, All how can i run rtpproxy to save recorded conversation locally? i tried: /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media but got nothing under /var/media after making calls /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media -S /var/media also got nothing? has anyone tried it? thank you regards, nhadie

1 0

[OpenSER-Users] domain name in route header
by jh＠tutpro.com 14 Sep '07

14 Sep '07

klaus, if you want to support domains in route headers, then you need to do naptr and srv lookups for the domain when handling loose routing. it is not enough to compare the domain to the ones domain module knows about. domain in route header may namely be anything as long as it resolves to your proxy. for this reason, i don't like the idea of domains in route headers. it hurts performance far too much. -- juha

1 0

[OpenSER-Users] help with outbound proxy
by Dawid Jarosławski 14 Sep '07

14 Sep '07

Hello, I'm trying to make outbound proxy for my network using OpenSER. Currently I've configured OpenSER as sip registrar/rtpproxy and it works fine even with NAT traversal. The problem is that I don't how shuould I reconfigure my config to make it work as outbound proxy. I've tried many changes with no results. The first problem is how to correctly handle the register method. My config is based on: http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html Can anyone help me? Best regards, Dawid Jaroslawski

2 5

[OpenSER-Users] Patch for openserunix.c
by Patrick Miccio 14 Sep '07

14 Sep '07

Hi everyone, I use 1.1.1 notls, in my testing environment I came across some strange errors when using a home made script in combination with the openserunix util. First of all there were thousands of files like"/tmp/OpenSER.XXXXXX" but looking around with google I found a patch for that: https://sourceforge.net/tracker/?func=detail&atid=743022&aid=1708301&group_… Then with the home made script I was still getting lots of socket errors in syslog, a colleague of mine fixed the openserunix.c ############################################################ --- openserunix.c.ori 2007-09-13 19:54:00.406861237 +0200 +++ openserunix.c 2007-09-13 19:54:07.407260167 +0200 @@ -97,6 +97,7 @@ goto err; } + from_len = sizeof(from); len = recvfrom(sock, buffer, BUF_SIZE, 0, (struct sockaddr*)&from, &from_len); if (len == -1) { fprintf(stderr, "Error in recvfrom: %s\n", strerror(errno)); ############################################################ maybe this is useful for someone since even the newest OpenSER version hast the "buggy" openserunix util :-) cheers, Patrick.

4 3

[OpenSER-Users] [presence] "in-dialog" SUBSCRIBE without "Route" record
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I have presence in OpenSer same box. modparam("presence","server_address","sip:85.92.0.210") I've a problem because I get "in-dialog" SUBSCRIBE messages without "Route" header: When my SIP client (Twinkle or X-Lite) starts up it sends a SUBSCRIBE NOT-in-dialog: ---------------------------------------------------------------------------------------------------------- # U 2007/09/10 17:54:08.294012 222.21.235.18:5060 -> 85.92.0.210:5060 SUBSCRIBE sip:ibc@sip.domain.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKcsmbires Max-Forwards: 70 To: <sip:ibc@sip.domain.net> From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 636 SUBSCRIBE Contact: <sip:angel@192.168.1.58> Accept: application/pidf+xml Event: presence Expires: 30 User-Agent: Twinkle/1.1alpha2 Content-Length: 0 # U 2007/09/10 17:54:08.296369 85.92.0.210:5060 -> 222.21.235.18:5060 SIP/2.0 202 OK Via: SIP/2.0/UDP 192.168.1.58;rport=5060;branch=z9hG4bKcsmbires;received=222.21.235.18 To: <sip:ibc@sip.domain.net>;tag=10.14184.1189439648.1 From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 636 SUBSCRIBE Expires: 30 Contact: <sip:85.92.0.210> Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 # U 2007/09/10 17:54:08.297746 85.92.0.210:5060 -> 222.21.235.18:5060 NOTIFY sip:angel@222.21.235.18:5060 SIP/2.0 Via: SIP/2.0/UDP 85.92.0.210;branch=z9hG4bKfaad.3f308931.0 To: sip:angel@sip.domain.net;tag=wddxn From: sip:ibc@sip.domain.net;tag=10.14184.1189439648.1 CSeq: 2 NOTIFY Call-ID: bdrmoffsihrozyj(a)192.168.1.58 Content-Length: 342 User-Agent: OpenSER (1.2.2-notls (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:85.92.0.210> Subscription-State: active;expires=30 Content-Type: application/pidf+xml ---------------------------------------------------------------------------------------------------------- And after 60 minutes (presence refresh in X-Lite) the following SUBSCRIBE from X-Lite is this sent. Note that: - This is a IN-DIALOG SUBSCRIBE (has totag and same callerd ID as before). - This in-dialog SUBSCRIBE hasn't "Route" header !!!!!!!!!!! ---------------------------------------------------------------------------------------------------------- # U 2007/09/10 17:54:35.390759 222.21.235.18:5060 -> 85.92.0.210:5060 SUBSCRIBE sip:85.92.0.210 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKpgezzksl Max-Forwards: 70 To: <sip:ibc@sip.domain.net>;tag=10.14184.1189439648.1 From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 638 SUBSCRIBE Contact: <sip:angel@192.168.1.58> Accept: application/pidf+xml Event: presence Expires: 30 User-Agent: Twinkle/1.1alpha2 Content-Length: 0 ---------------------------------------------------------------------------------------------------------- So, because this second SUBSCRIBE hasn't a "Route" header the funcion "loose_route()" doesn't match it and it's routed as NOT-in-dialog message. So, what to do??? I've a in-dialog message without "Route" header, so it's not routed by "loose_route()". I've tryied with this: if (loose_route() || has_totag()) { but it generates a loop. Does OpenSer store the SUBSCRIBE dialog info for 60 minutes? could it recognize this establishd dialog? or should I route this in-dialog message as NOT in-dialog? Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 1

[OpenSER-Users] About presence server in other host
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I will separate the presence server in a different host, and I'd like to know your opinions: - For SUBSCRIBE messages main OpenSer will "rewriteuri(presence_serve)" and will NOT apply "record_route()" in order to avoid in-dialog NOTIFY to pass through main OpenSer. Is it ok? - "server_address" for Presence module will have: "sip:presence-server:5060" (of course presence module will just exist in presence-server). - Any other suggestion? In which cases is necessary to do this separation? with how much SIP traffic is it interesting? Thanks for any help. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[OpenSER-Users] record_route_preset("x.x.x.x:5060; nat=yes"); and search("^Route:.*; nat=yes")
by Halomoan Chow 14 Sep '07

14 Sep '07

Dear All I don't understand why in the Openser+MediaProxy i need record_route_preset("x.x.x.x:5060;nat=yes"); and search("^Route:.*;nat=yes") What is the purpose while I can do client_nat_test to see if the request is come from a nat-ed client. If I remove search("^Route:.*;nat=yes") from below statement, I can do re-INVITE without any problem because it still have client_nat_test to find out the nat in between. if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); use_media_proxy(); }; Hope someone can give me a clue. Thank you. Regards, Halomoan

2 5

Re: [OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 14 Sep '07

14 Sep '07

On Thu, Sep 13, 2007 at 04:51:22PM +0200, Klaus Darilion wrote: > no - two: Right, my mistake. > Two identical NAPTR. One in RFC 3761 syntax and one in 2916 syntax. > Looks like enum_query() supports both syntax. Yes, it is my problem. I am sending this information for other users who will crawl this archive. :) > Solution 1: fix the NAPTR I sent request to their DNS administrators, it is not my site. > Solution 2: use enum_query("e164.arpa.","+sip"); (not sure about this one) I can confirm this. Works well before they update their DNS records. Thank you for help. SAL

1 0

Re[2]: [OpenSER-Users] OPTIONS problem
by Ion Minzu 14 Sep '07

14 Sep '07

Hello Bogdan-Andrei, Friday, September 14, 2007, 10:15:55 AM, you wrote: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: SIP Request: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: method: <OPTIONS> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: uri: <sip:820999@voip.md> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: version: <SIP/2.0> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=2 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: Found param type 235, <rport> = <n/a>; state=17 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: end of header reached, state=5 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: Via found, flags=2 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: this is the first via Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: After parse_msg... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: preparing to run routing scripts... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=100 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:parse_to:end of header reached, state=9 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBUG:parse_to: display={}, ruri={sip:820999@voip.md} Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:820999@voip.md] Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: to body [sip:820999@voip.md ] Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: get_hdr_field: cseq <CSeq>: <1> <OPTIONS> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: get_hdr_body : content_length=0 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: add_param: tag=gd60cu Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:parse_to:end of header reached, state=29 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBUG:parse_to: display={}, ruri={sip:checksip@sheik:37176} Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=200 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: find_first_route: No Route headers found Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: loose_route: There is no Route HF Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: rewrite_uri: Rewriting Request-URI with 'sip:820999@172.17.6.103' Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: t_newtran: msg id=106 , global msg id=105 , T on entrance=0xffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:sl:sl_reply_error: error text is I'm terribly sorry, server error occurred (1/SL) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]:check_via_address(Y.Y.Y.Y, sheik, 0) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBG:sl:run_sl_callbacks: callback id 0 entered Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: trace_sl_onreply_out: ================================= Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: trace_sl_onreply_out: trace off... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:destroy_avp_list: destroying list (nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: receive_msg: cleaning up Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:0,tl=0xb5e44354 next=0xb5e3bac8 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:0,tl=0xb5e3bac8 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:1,tl=0xb5e3bb34 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:4,tl=0xb5e44344 next=0xb5e3bab8 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:4,tl=0xb5e3bab8 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: SIP Request: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: method: <BYE> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: uri: <sip:X.X.X.X:5060;r2=on;lr=on;ftag=1455B87C-17C9> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: version: <SIP/2.0> > Hi, > Have you checked the openser logs to get more details about the error? > regards, > bogdan > Ion Minzu wrote: >> Hello , >> >> I have an working openser, i installed and configured in Nagios check_sip >> plugin, the problem is that openser answer to OPTIONS message with the error >> 500. >> Help me >> >> # >> U Y.Y.Y.Y:49813 -> X.X.X.X:5060 >> OPTIONS sip:820999@voip.md SIP/2.0 >> Via: SIP/2.0/UDP sheik:49813;rport >> From: sip:checksip@sheik:49813;tag=gd60cu >> To: sip:820999@voip.md >> Call-ID: dailgg@sheik >> CSeq: 1 OPTIONS >> Contact: sip:checksip@sheik:49813 >> Content-length: 0 >> Max-Forwards: 70 >> User-agent: check_sip 1.01 >> Accept: text/plain >> >> # >> U X.X.X.X:5060 -> Y.Y.Y.Y:49813 >> SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). >> Via: SIP/2.0/UDP sheik:49813;rport=49813;received=Y.Y.Y.Y. >> From: sip:checksip@sheik:49813;tag=gd60cu >> To: sip:820999@voip.md;tag=420087ba42f57437a013c6a41fc4e4c8.b0df >> Call-ID: dailgg@sheik >> CSeq: 1 OPTIONS >> Server: OpenSer (1.1.1-notls (i386/linux)). >> Content-Length: 0. >> >> >>

2 1

[OpenSER-Users] need help
by luhan 14 Sep '07

14 Sep '07

hi master-hand: i had rencontred a problem.when i configured my Openser work together with wesip,use two linux OS,one use to run openser,other run wesip, this is my configure file: openser: ------------------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) children=8 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_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" "openser.cfg" 154L, 3866C 14,1 Top # 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) { 61,1 49% }; 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(2); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ if(!as_relay_t("app_server_one")){ t_reply("500","Application Server error"); } 121,3-17 98% } ----------------------------- and this is wesip: ----------------------------- <Server port="8005" shutdown="SHUTDOWN" debug="0"> <Service name="Tomcat-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="8080" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" addresses="localhost:5060" pathName="com.voztele" protocol="tcp" > <ExtraProperties> <Property key="javax.sip.SER_ADDRESS" value="192.168.7.15" /> <Property key="javax.sip.SER_PORT" value="5080" /> <Property key="STATISTICS" value="true" /> <Property key="javax.sip.STACK_NAME" value="app_server_one" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="localhost" debug="0" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> ----------------------------------------------------------------------- at the debug model: found those tips: WARNING: using only the first listen address (no fork) 0(0) init_unixsock_socket: No unix domain socket will be opened ....... bind (Cannot assign requested address) -1(0) unable to open server sockets on dispatcher -1(0) ERROR:init_mod_child: Error while initializing module seas, rank 1 -1(0) main_dontfork: init_child failed would you help me! thank you very much. All my best regards. ------------------------------ (luhan china)卢涵技术部 E-mail: luhan(a)bjybkl.com phone: 010-59828003 mobile: 13810262476 ------------------------------

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[OpenSER-Users] OPTIONS problem
by Ion Minzu 14 Sep '07

14 Sep '07

Hello , I have an working openser, i installed and configured in Nagios check_sip plugin, the problem is that openser answer to OPTIONS message with the error 500. Help me # U Y.Y.Y.Y:49813 -> X.X.X.X:5060 OPTIONS sip:820999@voip.md SIP/2.0 Via: SIP/2.0/UDP sheik:49813;rport From: sip:checksip@sheik:49813;tag=gd60cu To: sip:820999@voip.md Call-ID: dailgg@sheik CSeq: 1 OPTIONS Contact: sip:checksip@sheik:49813 Content-length: 0 Max-Forwards: 70 User-agent: check_sip 1.01 Accept: text/plain # U X.X.X.X:5060 -> Y.Y.Y.Y:49813 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). Via: SIP/2.0/UDP sheik:49813;rport=49813;received=Y.Y.Y.Y. From: sip:checksip@sheik:49813;tag=gd60cu To: sip:820999@voip.md;tag=420087ba42f57437a013c6a41fc4e4c8.b0df Call-ID: dailgg@sheik CSeq: 1 OPTIONS Server: OpenSer (1.1.1-notls (i386/linux)). Content-Length: 0. -- Best regards, Ion Minzu, Specialist Tehnologii Informationale, I.S."Centrul de Telecomunicatii Speciale", tel:250-540, 820922 (office), 069501208 (mob), 382869185 (ICQ) mailto:ion.minzu@cts.md

2 1

[OpenSER-Users] Makefile "bug" in the latest 1.2.2 tarball...
by Geir O. Jensen 14 Sep '07

14 Sep '07

Hey, Downloaded the 1.2.2-tls tarball, extracted and did a make all - install. Everything seemed to work fine untill I tried to start it, whereupon it complained that tls was not built in. Looking at the Makefile I spotted the #TLS=1 - whence uncommented and rebuilt made everything a.o.k. I'm guessing if I did an export TLS=1 before make all it would also work? But when you give out a tarball that's supposed to include support for TLS I think I shouldn't need to do anything else than make all... Correct? So I'm calling this a bug :D // sip:gojensen@uninett.no | h323: 004710012 pstn: +47 73 55 79 23 | fax: +47 73 55 79 01

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[OpenSER-Users] How to automate the new user registration process?
by lu 14 Sep '07

14 Sep '07

Hi all, Is there a simple way/opensource tool that can handle all the new VoIP users registration to the Server side automatically without human intervention/manual entry? iam using Openser 1.2 with Mysql 5.0 database. cheers.

2 1

[OpenSER-Users] Do most SIP devices support SRV DNS lookup?
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I'd like to know about experiences using SRV DNS: - Do most SIP device support it? - Will I have unexpected surprises with some SIP phones or softphones because don't support it? - How often do the SIP devices supporting SRV have it enabled by default? Of course I will test it by myself, but I'd like to be warned about it :) Thanks for any response. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[OpenSER-Users] tls errors
by Jeremy George 14 Sep '07

14 Sep '07

Hello All, I could use some help with tls. Any pointers to a possible solution would be greatly appreciated. Please let me know if any further information would be helpful. Thanks. Everything seems to register correctly. Openserctl ul show seems correct (see below.) For a short time all is well. I can call from/to Polycom hard phones. (Eyebeam never seems to make calls, though it does register correctly.) Then calls stop working, and I begin getting the following error messages. Sep 12 16:17:16 spongy openser[30644]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Sep 12 16:17:16 spongy openser[30644]: ERROR: tcpconn_connect: tcp_blocking_connect failed Sep 12 16:17:16 spongy openser[30644]: ERROR: tcp_send: connect failed Sep 12 16:17:16 spongy openser[30644]: msg_send: ERROR: tcp_send failed Sep 12 16:17:16 spongy openser[30644]: ERROR:tm:t_forward_nonack: sending request failed Sep 12 16:17:53 spongy openser[30644]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:17:53 spongy openser[30644]: ERROR:tm:w_t_relay: t_forward_nonack failed Sep 12 16:17:59 spongy openser[30629]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:17:59 spongy openser[30629]: ERROR:tm:w_t_relay: t_forward_nonack failed Sep 12 16:18:25 spongy openser[30625]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:18:25 spongy openser[30625]: ERROR:tm:w_t_relay: t_forward_nonack failed I thought the errors might be caused by an underlying tcp configuration issue on the machine, so I reconfigured the proxy and uas to use tcp (but not tls) and I had no trouble. Googling and looking through the archives seems to indicate I'm not the first person to have this problem. http://www.ip-phone-forum.de/archive/index.php/t-100539.html (I don't know if there is an answer here; unfortunately, I can't read German :) Spongy% sudo openser -V version: openser 1.2.2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 2304 2007-05-25 16:36:07Z bogdan_iancu $ main.c compiled on 11:33:05 Sep 1 Spongy% sudo openserctl ul show Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=5 max_slot=1 AOR:: didama Contact:: sip:didama@75.56.10.117:37522;transport=tls Q= Expires:: 227 Callid:: acae15de-44fdabfd-e712aa40(a)75.56.10.117 Cseq:: 38 User-agent:: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 AOR:: jeremy Contact:: sip:jeremy@75.56.10.115:5060;transport=udp Q= Expires:: 85 Callid:: 00036b09-50520002-40d733bc-0d83cc25(a)75.56.10.115 Cseq:: 140 User-agent:: Cisco-CP7960G/8.0 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:128.91.3.19:5060 Methods:: 4294967295 AOR:: jessica Contact:: sip:jessica@75.56.10.116:55250;transport=tls Q= Expires:: 1196 Callid:: ef97ec67-46682f92-799b9cc1(a)75.56.10.116 Cseq:: 84 User-agent:: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 AOR:: peggy Contact:: sip:peggy@75.56.10.113:1210;transport=TLS;rinstance=dd48392409b03c0a Q= Expires:: 3567 Callid:: M2NhMmI1OGZkZThhYjUzMmZiZTcxY2M0MjhlODE1ZDQ. Cseq:: 4 User-agent:: eyeBeam release 1013t stamp 43069 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 5951 AOR:: helen Contact:: sip:helen@75.56.10.116:55250;transport=tls Q= Expires:: 1197 Callid:: f556d6ba-86bff9c9-f62d758c(a)75.56.10.116 Cseq:: 84 User-agent:: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 Contact:: sip:helen@75.56.10.117:37522;transport=tls Q= Expires:: 228 Callid:: dd7e9645-83cfde68-f0f8341f(a)75.56.10.117 Cseq:: 38 User-agent:: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 - Jeremy

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[OpenSER-Users] 513 Massage too big
by Arya 14 Sep '07

14 Sep '07

Hello everyone I have a problem connecting to OpenSER from my public IP. I can connect and make calls to OpenSER inside my LAN but when I connect from outside LAN I get the 513 Massage too big massage. I than changed the openSER default config's section that checks a message size to a bigger number and I got the I get the "483 - too many Hops" error. I changed the config back to default and tested it with ngrep and I tried connecting to the server from a server in another state. The SIP server is on 192.168.1.109 and the computer trying to connect is on 192.168.1.100 And ngrep gave me these massages: Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 67.186.60.123:29296 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. -- Thank You

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Re: [OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 13 Sep '07

13 Sep '07

Hello, On Wed, Sep 12, 2007 at 05:32:15PM +0200, Klaus Darilion wrote: > 1. I have forgotten to tell you to use the "-W byline" switch too in > ngrep (much more readable SIP logs) I can resend you new logs if needed. > 2. Nevertheless, I have found the error: In the openser proxy your > create a second branch: Take a look at the logs - there are 2 INVITEs > sent from openser to ser - the only different in these INVITEs is the > branch parameter in the topmost Via header. > > The first is: > branch=z9hG4bK41a.25464943.0 > the second is: > branch=z9hG4bK41a.25464943.1 > > Thus, openser does parallel forking. Then, when Grandstream answers the > second branch (branch=z9hG4bK41a.25464943.1) openser will CANCEL the > first branch. Right. There is no problem with CANCEL but with 2 INVITEs. I have this in my openser log: 2(13970) DEBUG: t_lookup_request: no transaction found 2(13970) DBG: trans=0xb60d6e80, callback type 1, id 1 entered 2(13970) DEBUG:dialog:build_new_dlg: new dialog on hash 2386 2(13970) DBG: trans=0xb60d6e80, callback type 1, id 0 entered Is this a right place where the second fork is done? > Thus, you have either bug in your config or in your provisioning (maybe > you have the user twice in alias table or location table). Which command can create this second INVITE? What I need to find in my config? This user is not local, it is found by ENUM. My ENUM configuration: if (uri=~"^sip:\+[0-9]+@") { if (enum_query("e164.arpa.")) { xlog("L_ERR", "ENUM[e164.arpa]: [$rm] from user $fu to $ru\n"); route(5); return; }; if (enum_query("e164.arpa.", "voice")) { route(5); return; }; if (enum_query("e164.bts.sk.")) { route(5); return; }; }; Thanks. SAL

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[Serusers] FW: problem in DNS lookup from enum
by geethanjali.csg＠wipro.com 13 Sep '07

13 Sep '07

Hi, I am configuring ser to use enum module. I am getting this error when dns lookup is done from ser. 0(27798) get_record: lookup(8.7.6.e164.arpa, 35) failed 0(27798) enum_query(): No NAPTR record found for 8.7.6.e164.arpa. This is my zone file: $TTL 3D @ IN SOA e164.arpa. root( 4 ; serial (d. adams) 3H ; refresh 15M ; retry 1W ; expiry 1D ) ; minimum IN NS localhost IN NS e164.arpa. localhost A 127.0.0.1 e164.arpa. A 10.200.220.200 $ORIGIN 1.2.3.4.e164.arpa. IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:abcd@e164.arpa!" . $ORIGIN 8.7.6.e164.arpa. IN NAPTR 20 100 "u" "E2U+sip" "!^.*$!sip:xyz@e164.arpa!" . named.conf file contains, zone "e164.arpa" IN { type master; file "/var/named/named.enum"; }; This is my enum code in ser.cfg. loadmodule "/usr/local/lib/ser/modules/enum.so" modparam("enum", "domain_suffix", "e164.arpa.") if (method == "INVITE" && uri =~ "sip:\+[0-9]*+@.") { if (enum_query("e164.arpa")) { route(1); }; }; Its possible to ping the domain e164.arpa from the m/c where ser is installed. Any help in this regard is appreciated. Regards Geethanjali 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

1 0

[OpenSER-Users] Installing OpenSER in multiprocessor machine
by John Barry 13 Sep '07

13 Sep '07

Hello, I would like to know if there are any especial considerations when compiling, configuring, and running OpenSER v1.2.2 in a multiprocessor machine. I am using a quad-processor machine (Intel Xeon @ 3.00GHz) with Red Hat Enterprise Linux AS release 4 (2.6.9-42.0.3.ELsmp #1 SMP x86_64 x86_64 x86_64 GNU/Linux). I am also using MySQL DB for authentication in the same machine. I will appreciate any advice or information. Bests, JB _________________________________________________________________ Check out some new online services at Windows Live Ideas—so new they haven’t even been officially released yet. http://ideas.live.com

2 1

[Serusers] problem in DNS lookup from enum
by geethanjali.csg＠wipro.com 13 Sep '07

13 Sep '07

Hi, I am configuring ser to use enum module. I am getting this error when dns lookup is done from ser. 0(27798) get_record: lookup(8.7.6.e164.arpa, 35) failed 0(27798) enum_query(): No NAPTR record found for 8.7.6.e164.arpa. This is my zone file: $TTL 3D @ IN SOA e164.arpa. root( 4 ; serial (d. adams) 3H ; refresh 15M ; retry 1W ; expiry 1D ) ; minimum IN NS localhost IN NS e164.arpa. localhost A 127.0.0.1 e164.arpa. A 10.200.220.200 $ORIGIN 1.2.3.4.e164.arpa. IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:abcd@e164.arpa!" . $ORIGIN 8.7.6.e164.arpa. IN NAPTR 20 100 "u" "E2U+sip" "!^.*$!sip:xyz@e164.arpa!" . named.conf file contains, zone "e164.arpa" IN { type master; file "/var/named/named.enum"; }; This is my enum code in ser.cfg. loadmodule "/usr/local/lib/ser/modules/enum.so" modparam("enum", "domain_suffix", "e164.arpa.") if (method == "INVITE" && uri =~ "sip:\+[0-9]*+@.") { if (enum_query("e164.arpa")) { route(1); }; }; Its possible to ping the domain e164.arpa from the m/c where ser is installed. Any help in this regard is appreciated. Regards Geethanjali 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

1 0

Re: [Serusers] WARNING: xl_mod_init: more IP 10.0.0.21 not used and ERROR: init_tcpudp_sock: bind: Address already in use [98]
by Jiri Kuthan 13 Sep '07

13 Sep '07

At 11:55 12/09/2007, Frank Wang wrote: >I downloaded ser 2.0.0-rc1 source code from <http://www.iptel.org/ser_2_0_0_release_candidate_1>http://www.iptel.org/ser_2_0_0_release_candidate_1 and install on my Linux step by step according the tutorial on the WEB. > >When I running SER, I got follow messages: > >[root@sipmcu ser]# ../../sbin/ser -f ser.cfg >Listening on > udp: <http://10.0.0.21>10.0.0.21 [<http://10.0.0.21>10.0.0.21]:5060 > tcp: <http://10.0.0.21>10.0.0.21 [<http://10.0.0.21>10.0.0.21]:5060 >Aliases: > tcp: <http://sipmcu.leadtek.com.cn:5060>sipmcu.leadtek.com.cn:5060 > udp: <http://sipmcu.leadtek.com.cn:5060>sipmcu.leadtek.com.cn:5060 > *: mytestser.org:* > >WARNING: no fork mode > 0(3078) init_tcp: using epoll_lt as the io watch method (auto detected) > 0(3078) Maxfwd module- initializing > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used unimportant. (I checked the code and I was a bit puzzled by it, but it really shold not matter much.) > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) ERROR: init_tcpudp_sock: bind: Address already in use [98] This is the problem. Maybe you have a SER instance running from the last time and you are starting the script again, for which reason SER can't be restarted -- it can't claim the same port number twice. > 0(3078) ERROR: ctl: mod_init: init ctrl. sockets failed > 0(3078) init_mod(): Error while initializing module ctl >ERROR: error while initializing modules This could have same reason. > >could anybody tell me how about it? Thanks a lot. > >Regards, >Frank >_______________________________________________ >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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Re: [OpenSER-Users] WeSIP with openser 1.2.0 again
by Elias Baixas 13 Sep '07

13 Sep '07

Hi Gregorio, posting the SIP message that caused the problem would help me reproduce it, also some OpenSER logs would help. It seems to be related to the binary-encoding of the message. try out sipSpy to capture messages if you like. remove any field which you find confidential (lest Max-Forwards...) from the SIP message send it to me privately if you want. regards, Elias Gregorio wrote: > > Hi everyone, > > > > Hi! > > Just installed WeSIP beta-0.1.4 and openser 1.2.0 in the same machine, > configuration looks alright because some SIP Servlets receive > messages. Now I am getting an exception in the SEAS module when I > receive MESSAGE, I think this problem is similar to a previous one in > the mailing list (Sat Apr 28 16:07:34 CEST 2007, [WeSIP with openser > 1.2.0 again]). Does someone know anything about it? > > > > This is the exception: > > > > Thanks for helping. > > > > java.lang.ArrayIndexOutOfBoundsException: 5942 > > at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) > > at > com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) > > at > com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) > > at > com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) > > at > com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) > > at > com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) > > at > com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) > > at > com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:357) > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > Exception in thread "ExpressMChannel[1]" java.lang.NullPointerException > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > java.lang.ArrayIndexOutOfBoundsException: 5942 > > at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) > > at > com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) > > at > com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) > > at > com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) > > at > com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) > > at > com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) > > at > com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) > > at > com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:357) > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] redirect server and handling 302 messages
by erik.linder＠accenture.com 12 Sep '07

12 Sep '07

Hello, I'm trying to make openser process 302 messages though I have problems(part of .cfg file below). I'll briefly explain the call flow I'd like to have: UA1 ---INVITE---> Session border controller ---INVITE--->SIP PROXY1 SIP PROXY1---INVITE--->REDIRECT SERVER---302 MOVED--->SIP PROXY1 SIP PROXY1---INVITE--->SIP PROXY2---INVITE---> Session border controller---INVITE---> UA2 This is how I want it to work. However managed to get a weird call flow were the 302 is forwarded back to the session border controller by the proxy1 which then trigger an INVITE sent to the SIP proxy2. But that is not what I want. My 302 message looks like this 20:12:34.314705 IP 192.168.200.101.sip > 192.168.200.102.sip: SIP, length: 488 E.....@.@.%....e...f........SIP/2.0 302 Moved temporarily Via: SIP/2.0/UDP 192.168.200.102;branch=z9hG4bK1658.2543f3d6.0 Via: SIP/2.0/UDP 192.168.200.10:5060;branch=z9hG4bKpis9m130dgj04b08v7k1.1 To: "22222"<sip:22222@btsip.bt.com>;tag=cc0e227ef05fff2dd833db745f3905a6.25f 2 From: "111"<sip:11111@btsip.bt.com>;tag=SDvto0501-df614a6e Call-ID: SDvto0501-515b91532ff7927a8345f39add52762f-06ahc50 CSeq: 1 INVITE Contact: , sip:22222@mydomain.com;maddr=192.168.200.103:5060;side=term> Content-Length: 0 192.168.200.101 is redirect server 192.168.200.102 is the proxy1 192.168.200.103 is proxy2 192.168.200.10 is session border controller 1. I would like to have proxy1 process the 302 message but there seems (from other mails on the mailing list) that openser does not allow routing depending on maddr in the Contact field. Sic :-( <- how can this routing decision been taken , using rewritehostport after get_redirect("*") is kind of ignored. The only workaround I have is to change the mydomain.com to the IP of proxy2 to get the automatic redirect 2. How can I block the redirect to the session border controller and send an invite right away to the proxy2? Here are the relevant part of the logic I use in proxy1 to handle the 302 I'm using this to trigger the forward to the redirect server: lookup("aliases"); #dest is remote if (!lookup("location")) { t_on_failure("3"); xlog("L_INFO","redirect invite to SLS\n"); #change orig to term in request URI $avp(s:request_uri_to_use)=$ru; avp_subst("$avp(s:request_uri_to_use)", "/orig/term/"); avp_pushto("$ru","$avp(s:request_uri_to_use)"); #forward("udp:192.168.200.101:5060"); #exit; rewritehostport("192.168.200.101:5060"); if(!t_relay()) { sl_reply_error(); } return; and this to try to handle the 302 that is received but it does not behave like I want: failure_route[3] { xlog("L_WARN","redirect FAILUREROUTE: time_t=$Ts messageid=$mi remark=here we are $bR\n"); if(t_check_status("3[0-9][0-9]")) { get_redirects("*"); xlog("L_WARN","^method=$rm ^ruri=$ru ^messageid=$mi^remark=2status is 3xx\n--$br--\nbR((($bR)))\n"); #rewritehost("$(ru{uri.maddr})"); serialize_branches(0); } xlog("L_WARN","nextbranch\n"); #next_branches(); #t_on_failure("3"); #t_relay(); return; #t_reply("404", "No branches left"); } Help would be appreciated Cheers erik This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited.

2 2

[OpenSER-Users] authorization problems with multiple servers
by Ben Smithurst 12 Sep '07

12 Sep '07

Hi We have a problem with clients authorizing (whether for INVITE or REGISTER) against a pair of openser servers, served via either round robin A records, or multiple equal preference SRV records. The problem in a nutshell is client sends INVITE to server 1 server 1 replies with authorization challenge client sends INVITE with auth to server 2 server 2 doesn't like the 'nonce' value Now, this only happens via our nat proxy, as far as we can tell, so I think the problem lies there (and I have contacted the vendor of that system). However I wonder if there is any way to make openser work in such a setup, in case other client do the same thing (which IMO is quite a stupid thing to do) ... would the valid 'nonce' values need to be stored in the DB? Incidentally we are using mysql authentication and we are using usrloc in db_mode 3, with both openser machines pointed at the same mysql server. We're using a pretty recent openser SVN revision (dev15). Any tips would be appreciated. thanks -ben -- Ben Smithurst http://www.bensmithurst.com/

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[OpenSER-Users] WeSIP with openser 1.2.0 again
by Gregorio 12 Sep '07

12 Sep '07

Hi everyone, Hi! Just installed WeSIP beta-0.1.4 and openser 1.2.0 in the same machine, configuration looks alright because some SIP Servlets receive messages. Now I am getting an exception in the SEAS module when I receive MESSAGE, I think this problem is similar to a previous one in the mailing list (Sat Apr 28 16:07:34 CEST 2007, [WeSIP with openser 1.2.0 again]). Does someone know anything about it? This is the exception: Thanks for helping. java.lang.ArrayIndexOutOfBoundsException: 5942 at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) at com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266 ) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser .java:357) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[1]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) java.lang.ArrayIndexOutOfBoundsException: 5942 at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) at com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266 ) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser .java:357) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595)

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[OpenSER-Users] Wanted: VoIP Engineer for Warsaw !
by laurent schweizer 12 Sep '07

12 Sep '07

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>and AVM/FRITZ!Box <http://www.fritz-shop.ch/>. Due to our rapid growth, for our new Polish subsidiary we are currently seeking for: VOIP SPECIALIST Place of work: Warsaw *Requirements:* - Graduation from college or university with a Bachelor's degree (preferably IT) - Experience with PHP - Practical knowledge of C and C++ - Practical knowledge of Mysql and Postgresql - Linux experience - Knowledge of IP Networks, UDP, TCP - Experience with tools for Network analysis like Ethereal - VoIP basic knowledge, VoIP servers, configuration of devices - Fluency in English - Ability to interact with individuals and groups at all levels - Detail oriented and analytical - Strong verbal and written communication skills Knowledge of Perl, Java, Asterisk / SER / OPENSER, ability to configure routers, Cisco or Patton gateways, knowledge of SIP and STUN protocol, knowledge of NAT problems, of outbandProxy, knowledge of monitoring tools like Cactus, Nagios, MRTG or high availability tools like DRBD, Hearthbeat would be an additional asset. Interested individuals are requested to send their resume to: ewa.ciesla(a)peoplefone.com In your application please put the following statement: I hereby agree for processing the following personal information strictly for recruitment purposes in accordance with the regulation regarding the protection of personal data passed on the following date: 29.08.97r. Dz.U nr 133 poz. 883.

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[OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 12 Sep '07

12 Sep '07

Hello, I have problems calling from my phones connected to openser to grandstream phones on another university to ser. Problem is, that my openser generates a "CANCEL" SIP packet sent to grandstream phone, but no CANCEL received from my phone. Wireshark shows something like this: 1189587144.843962 158.195.15.100 -> 158.197.16.77 SIP/SDP Status: 200 OK, with session description 1189587144.845528 158.197.16.77 -> 158.195.15.100 SIP Request: CANCEL sip:+421xxxx44903@domain.sk 1189587144.853654 158.195.15.100 -> 158.197.16.77 SIP Status: 200 canceling Communication before is OK. 158.197.16.77 is my openser server and 158.195.15.100 is peers phone. Any idea, why openser can generate this packet? Can openser generate a SIP packet not received from client? Calling from my phone to another phone on same university works well. Thank you. SAL

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[Serusers] WARNING: xl_mod_init: more IP 10.0.0.21 not used and ERROR: init_tcpudp_sock: bind: Address already in use [98]
by Frank Wang 12 Sep '07

12 Sep '07

I downloaded ser 2.0.0-rc1 source code from http://www.iptel.org/ser_2_0_0_release_candidate_1 and install on my Linux step by step according the tutorial on the WEB. When I running SER, I got follow messages: [root@sipmcu ser]# ../../sbin/ser -f ser.cfg Listening on udp: 10.0.0.21 [10.0.0.21]:5060 tcp: 10.0.0.21 [10.0.0.21]:5060 Aliases: tcp: sipmcu.leadtek.com.cn:5060 udp: sipmcu.leadtek.com.cn:5060 *: mytestser.org:* WARNING: no fork mode 0(3078) init_tcp: using epoll_lt as the io watch method (auto detected) 0(3078) Maxfwd module- initializing 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) ERROR: init_tcpudp_sock: bind: Address already in use [98] 0(3078) ERROR: ctl: mod_init: init ctrl. sockets failed 0(3078) init_mod(): Error while initializing module ctl ERROR: error while initializing modules could anybody tell me how about it? Thanks a lot. Regards, Frank

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Fwd: [OpenSER-Users] Transaction Deletion
by Brandon Armstead 12 Sep '07

12 Sep '07

---------- Forwarded message ---------- From: Brandon Armstead <brandon.armstead(a)gmail.com> Date: Sep 7, 2007 12:06 PM Subject: Re: [OpenSER-Users] Transaction Deletion To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Hello Bodgan, There is no noticable delay however the only thing I see is that one of the RR'ed proxy's responds sends out 200 OK rather than the original. You can see an example sipcap of this at this url: http://pastebin.ca/685897 . If you could let me know if you see anything that looks rather, odd, towards the end you can see the last few transactions and where there is no response to 216.82.224.203's 200 OK, where it flips from 216.82.224.202 to 216.82.224.203 from the RR. Also then later the PSTN follows with a BYE, due to no response, once again thanks for your help! Regards, Brandon On 9/7/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > > Hi Brandon, > > Is there any noticeable delay between the 200OK and ACK ? > > regards, > bogdan > > Brandon Armstead wrote: > > Hello Users, > > > > I have a weird scenario where the PSTN is responding with 200 OK, > > however Openser Never forwards on the ACK, after several > > retransmissions of 200OK to Openser, PSTN responds with BYE, anyone > > know what might be going on? I've played with the timers to no avail, > > any help is appreciated, thank you! The log below is as follows: > > > > 375 Sep 6 12:18:10 openser /usr/local/sbin/openser[17385]: DEBUG: > > timer routine:1,tl=0x2b78e4e73588 next=(nil), timeout=261 > > 376 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > timer routine:2,tl=0x2b78e4e733b8 next=(nil), timeout=262 > > 377 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > wait_handler : removing 0x2b78e4e73338 from table > > 378 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > delete transaction 0x2b78e4e73338 > > 379 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > wait_handler : done > > 380 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: > > DEBUG:parse_to:end of header reached, state=10 > > 381 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: > > get_hdr_field: <To> [26]; uri=[sip:openser] > > 382 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: to > > body [<sip:openser> ] > > 383 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: > > get_hdr_body : content_length=0 > > 384 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: > > DEBUG:destroy_avp_list: destroying list (nil) > > 385 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: > > DEBUG:forward_reply: found module tm, passing reply to it > > 386 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_check: start=0xffffffffffffffff > > 387 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > add_param: tag=gK0288a919 > > 388 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: > > DEBUG:parse_to:end of header reached, state=29 > > 389 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > get_hdr_field: <To> [53]; uri=[ sip:17144828306@openser] > > 390 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: to > > body [<sip:17144828306@openser>] > > 391 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: hash 53903 label 1526431020 branch 0 > > 392 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: no matching transaction exists > > 393 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: failure to match a transaction > > 394 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_check: end=(nil) > > > > It seems like its not matching the transaction, because a timer is > > removing the transaction? Any ideas? Thanks! > > -- > > Brandon Armstead > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > -- Brandon Armstead -- Brandon Armstead

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Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 12 Sep '07

12 Sep '07

Arya, it means you have a routing error in the script - you need to debug your script to see where the routing logic is broken - try using xlog() messages to see the execution path of your script. I cannot help you with more than this. regards, bogdan Arya wrote: > any idea how I can fix the problem? > > On 9/11/07, *Arya* <arya6000(a)gmail.com <mailto:arya6000@gmail.com>> > wrote: > > I used ngrep but it just shows that the massage keeps bouncing > server to client, and eventually it says massage too big > > > On 9/11/07, * Bogdan-Andrei Iancu* < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > use ngrep or tcpdump to get a network trace from the proxy > server, on > the loopback interface. > > Regards, > Bogdan > > Arya wrote: > > Can you tell me how I can trace it? could this be an issue > with my > > router? its the Linksys Wr54G and its set on DMZ to the server > > > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro> > > <mailto: bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>> wrote: > > > > Hi Arya, > > > > I suspect you have a routing problem that cause the > request to loop on > > openser (via loopback interface) - openser keeps sending the > > request to > > itself. That's why you get a "Message to big" or "too > many hops".... > > > > run a trace on lo to see if this is what's happening. > > > > regards, > > bogdan > > > > Arya wrote: > > > Hello everyone > > > > > > I have a problem connecting to OpenSER from my public IP. > > > > > > I can connect and make calls to OpenSER inside my LAN > but when I > > > connect from outside LAN I get the 513 Massage too big > massage. > > I than > > > changed the openSER default config's section that > checks a message > > > size to a bigger number and I got the I get the "483 - > too many > > Hops" > > > error. > > > > > > I changed the config back to default and tested it with > ngrep and I > > > tried connecting to the server from a server in another > state. > > > > > > The SIP server is on 192.168.1.109 > <http://192.168.1.109> <http://192.168.1.109 > <http://192.168.1.109>> > > <http://192.168.1.109> and the > > > computer trying to connect is on 192.168.1.100 > <http://192.168.1.100> > > < http://192.168.1.100> <http://192.168.1.100 < > http://192.168.1.100>> > > > > > > And ngrep gave me these massages: >

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[OpenSER-Users] 'while' statement in config file
by Daniel-Constantin Mierla 12 Sep '07

12 Sep '07

Hello, although is more part of devel, the decision will affect users of future stable releases, so I posted on both lists. Maybe some of you noticed that in the ongoing process of improving the pseudo-variables system, the AVPs and HDRs can be accessed with dynamic index. That means next lines are correct in the config file. $var(i) = 1; xlog("$(avp(i:20)[$var(i)])"); if($(avp(i:20)[$var(i)]) == 3) { } In the past, the index was static, as a number. The question I want to address is whether makes sense to add 'while' statements to iterate through avps or hdrs. More or less, the cycles can be done via recursive routes. The block: $var(i) = 0; while($(avp(i:20)[$var(i)]) != null) { ..... $var(i) = $var(i) +1; } is equivalent with: $var(i) = 0; route(1); .... route[1] { if(($(avp(i:20)[$var(i)]) == null) return; ... $var(i) = $var(i) +1; route(1); } 'while' may look more attractive, but could open the pandora box, people asking for more and more, that will turn config language is something very complex (not saying that is simple now :-) ). The drawback with recursive routes is the fixed number of routes that can be defined with the standard openser: 40 (number can be increased if openser is recompiled), because for each cycle another route has to be used. Waiting for your opinions... Cheers, Daniel

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[OpenSER-Users] add/remove headers and append_branch
by Madalina Pitis 12 Sep '07

12 Sep '07

Hello, I have a problem trying to remove a header using remove_hf() after an append_branch(). I am using 1.2 version of OpenSer. The scenario with the problem is as described in the folowing lines. When I receive an INVITE I try a parallel fork (one leg towards PSTN, the other towards IP), at which time I append a "Redirection" header. The PSTN leg will be answered with a 486 response, due to the presence of the "Redirection" header. When the IP leg expires, I receive a 408 response, at which time I do a serial fork, and resend the message towards the PSTN destination, this time without the "Redirection" header. The problem is that I cannot seem to be able to remove (or find) the header. I tried everywhere (before appending the new branch, after, in branch_route, before relaying), as you will see below. I tried searching the mailing lists but nothing turned up. I would greatly appreciate any help, for I am running out of ideas. Here is a simplified version of my configuration script, containing only the significant bits of the routing blocks: #--------------------------------------------------- # main routing logic #--------------------------------------------------- route { # check maxfwd and message lenght # do loose routing if (method=="INVITE") { # rewrite ruri rewriteuri(<PSTN destination>); # signal forking append_hf("Redirection: count=5\r\n"); append_branch(); revert_uri(); t_on_failure("1"); if (!t_relay()){ xlog("acc_info", " ERROR"); sl_reply_error(); } } else if (method=="ACK" || method=="BYE" || method=="CANCEL") { # default message handler } else { # forbidden message handler } } #--------------------------------------------------- # Failure route relay #--------------------------------------------------- route[1] { xlog("acc_info", " Failure route relay"); # try removing it before t_relay if(is_present_hf("Redirection")){ xlog("acc_info", " Remove before t_relay"); remove_hf("Redirection"); } if (!t_relay()){ xlog("acc_info", " ERROR"); sl_reply_error(); } } #--------------------------------------------------- # Failure route handling #--------------------------------------------------- failure_route[1] { xlog("acc_info", " failure route handler"); if(t_check_status("408")) { t_on_branch("1"); rewriteuri(<PSTN destination>); # try removing it before append branch if(is_present_hf("Redirection")){ xlog("acc_info", " Remove before append branch"); remove_hf("Redirection"); } append_branch(); # try removing it after append branch if(is_present_hf("Redirection")){ xlog("acc_info", " Remove after append branch"); remove_hf("Redirection"); } route(1); } }

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[OpenSER-Users] append_hf() and failure_route issue
by Alex Hermann 11 Sep '07

11 Sep '07

Hello all, I'm having some trouble with failure routes and custom SIP headers. In FAILURE_ROUTE I set the ruri to a new destination. This destination needs a custom SIP header. When I add a SIP header with append_hf() (from within the failure_route), this header is correctly sent to the first IP of the destination on t_relay(). However, when this destination fails with 503 or timeout, the built-in DNS based failover doesn't add the custom SIP header to the new requests for subsequent IP's. When the SIP header is added from REQUEST_ROUTE the header is sent to all destinations by the DNS based failover mechanism. How do I get the DNS based failover to send my SIP headers all the time? (The same problem is present with uac_auth()) -- Greetings, Alex Hermann

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Re: [OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by Iñaki Baz Castillo 11 Sep '07

11 Sep '07

First of all sorry for this post not being in the thread it belongs to (I've not the initial mail). > But, I don't want to use the Asterisk dtmf features, > I want to use the SIP INVITE/REFER features from the > UAC (actually polycom and snom phones). > > Can someboy shed me some light on this? You receive a call from Asterisk UAC and want to transfer (blind or attender) to other UACregistered in OpenSer. Of course you don't need Asterisk native transfer, SIP allows blind and attended transfer. The only you need is that your SIP devices allow both of them (unfortunatelly many SIP devices just allow blind transfer, but in most cases reading the device manual you'll find how to do a attended transfer). The point here is that Asterisk allows receiving a REFER, so don't worry for that (I use your same escenary with no problem and SIP transfers). Maybe you find useful this links: http://tech-invite.com/Ti-sip-service-4.html http://tech-invite.com/Ti-sip-service-5.html http://dev.openwengo.com/trac/openwengo/trac.cgi/wiki/SipTransfer Regards. -- Iñaki Baz Castillo

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Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 11 Sep '07

11 Sep '07

Hi, use ngrep or tcpdump to get a network trace from the proxy server, on the loopback interface. Regards, Bogdan Arya wrote: > Can you tell me how I can trace it? could this be an issue with my > router? its the Linksys Wr54G and its set on DMZ to the server > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Arya, > > I suspect you have a routing problem that cause the request to loop on > openser (via loopback interface) - openser keeps sending the > request to > itself. That's why you get a "Message to big" or "too many hops".... > > run a trace on lo to see if this is what's happening. > > regards, > bogdan > > Arya wrote: > > Hello everyone > > > > I have a problem connecting to OpenSER from my public IP. > > > > I can connect and make calls to OpenSER inside my LAN but when I > > connect from outside LAN I get the 513 Massage too big massage. > I than > > changed the openSER default config's section that checks a message > > size to a bigger number and I got the I get the "483 - too many > Hops" > > error. > > > > I changed the config back to default and tested it with ngrep and I > > tried connecting to the server from a server in another state. > > > > The SIP server is on 192.168.1.109 <http://192.168.1.109> > <http://192.168.1.109> and the > > computer trying to connect is on 192.168.1.100 > <http://192.168.1.100> <http://192.168.1.100 <http://192.168.1.100>> > > > > And ngrep gave me these massages: > > > > > > > > > -- > Thank You

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[OpenSER-Users] Openser and Voice master
by Dan 11 Sep '07

11 Sep '07

Hi, Has anybody able to successfully able to route calls to voicemaster from openser. I tried all method but none worked. From username/password to IP based authentication to prefix based authentication. Regards Dan

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[OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by perl＠libkvm.org 11 Sep '07

11 Sep '07

Hello, I am configuring a platform unig OpenSER 1.2 with Asterisk 1.2. I have all the SIP phones registered on OpenSER. I use the realtime setup on Asterisk using a view on MySQL. I need to be able to process the attended transfer between two UAC registered on OpenSER, having the first received a call from the Asterisk side, so that the second UAC gets the call transferred. I also need to do this as a blind transfer. But, I don't want to use the Asterisk dtmf features, I want to use the SIP INVITE/REFER features from the UAC (actually polycom and snom phones). Can someboy shed me some light on this? I need to evaluate if this is possible, or if it needs lots of sources modifications. Thank you very much.

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[Serusers] ERROR: warning_builder: buffer size exceeded
by Kostas Marneris 10 Sep '07

10 Sep '07

Hello all, I have occasionally the following warning mesgs at my SER log : Sep 10 14:34:27 serhost ser[17957]: ERROR: warning_builder: buffer size exceeded Sep 10 14:34:27 serhost ser[17957]: WARNING: warning skipped -- too big I noticed that the R-URI in that case is : R-URI=<sip:XXXXXXXXXX@mydomain.com;user=phone;transport=UDP> when usually the R-URI is : <sip:XXXXXXXXXX@mydomain.com;transport=UDP> <sip:XXXXXXXXXXXXX@mydomain.com> Anyone seen those mesgs before ? Can this explained by the extra field 'user=phone' which the SIP UA adds at the R-URI of the INVITE mesg ? thanks Kostas

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[OpenSER-Users] Problem with rtpproxy
by Jeferson Prevedello 10 Sep '07

10 Sep '07

Hi Guys, When I originate a call for an user behind of a NAT the call is without audio in the two ways. I believe that the problem is related to rtpproxy, however did not find nothing of wrong in my configuration. Some tip? openser.cfg: # ------------- !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # Nathelper with RTPproxy service run: ps -aux | grep rtp root 26098 0.0 0.0 2532 428 ? Ss 10:40 0:00 /usr/bin/rtpproxy root 11143 0.0 0.0 2536 352 ? Ss 11:30 0:00 /usr/bin/rtpproxy -l xxx.xxx.xxx.xxx -s udp:xxx.xxx.xxx.xxx 7890 logs: Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11859]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11860]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11861]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11862]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Regards Jeferson

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[Serusers] Need SIP proxy to translate UDP<->TCP
by Thermal Wetland 10 Sep '07

10 Sep '07

Hello, I would like to hire someone to setup a SIP UDP<->TCP proxy using SER. SER will sit between an Asterisk box and Microsoft Exchange 2007. Asterisk speaks UDP & the Exchange box speaks TCP. I will make a fresh fully patched CentOS 5 box available for the work. Payment could be through PayPal or Rent a Coder or similar. I would expect an SER expert to be able to do this in less than 2 hours. Please let me know if you are interested - thermalwetland at gmail dot com. Thanks, Thermal

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Re: [OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Hi Bogdan I got it fixed. The problem is on this line: if(!t_relay("0x05")) if I change to if(!t_relay()) everything is ok. Don't know what is 0x05 mean. Regards, Halomoan On 9/10/07, Halomoan Chow <halomoan(a)gmail.com> wrote: > > Hi > > My Openser version is openser-1.2.0-notls and the Mediaproxy is > mediaproxy-1.9.0 > I was using the openser.cfg that was generated from > http://www.sipwise.com/wizard/ for Openser + Mediaproxy only. > > Just only simple call from A to B cause segmentation fault. > > I don't know how to locate the core file and get the backtrace :( > > Regards, > > Halomoan > > On 9/10/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > > > > Hi Halomoan, > > > > provide full details about your openser version - version, how did you > > get it (source, binaries, etc). > > also locate the core file and get the backtrace. > > > > Regards, > > Bogdan > > > > Halomoan Chow wrote: > > > Dear All > > > > > > I'm running into segmentation fault error on Openser 1.2.x with > > > Mediaproxy. > > > What is the first thing I should take a look into to find out which > > > line or which module causing this segmentation fault? > > > Thank you. > > > > > > Regards, > > > > > > Halomoan > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

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Re: [Serusers] [Serdev] Release of SER 2.0 (and future 2.1)
by Greger V. Teigre 10 Sep '07

10 Sep '07

Ok, I conclude then that final release of SER 2.0 is aimed at October 15. That should allow us to align SER, serctl, SERWeb, rtpproxy, and SEMS. SER, SERWeb, and SEMS will have independent releases, and they will be aligned to be interoperable. SER, rtpproxy, and serctl will have a common release. The release will include the new buildsystem that was tested in release candidate 1. The config file and features available work with sercmd, serctl, as well as with SERWeb. SEMS support is planned, but pending some patches. All components will be combined in a one-file download with sources for all components, the SIP Express Bundle. I urge everybody with patches, bug reports, or tasks related to this release to prioritize those now! g-) Greger V. Teigre wrote: > Guys, > My last post concerned documentation. I suggested decoupling > documentation from the CVS to allow us to work independently with code > and docs. > > I suggest we move forward on the decoupling and set a date for SER 2.0 > release. My suggestion is that we target October 15. This will allow > people to allocate time to do the last clean-ups, bug-fixes, and patches. > > I also suggest that we set a release date for 2.1. AFAIU, there is > already quite a lot of code that has gone into the trunk and as 2.1 is a > minor release and we strive for ser.cfg compatibility with 2.0, we > shouldn't really wait too long. > > I suggest mid-January 2008. How does that work for you? > g-) > _______________________________________________ > Serdev mailing list > Serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev > > >

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[OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Dear All I'm running into segmentation fault error on Openser 1.2.x with Mediaproxy. What is the first thing I should take a look into to find out which line or which module causing this segmentation fault? Thank you. Regards, Halomoan

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[OpenSER-Users] How do I obtain or log the return code in a failure_route[]?
by Juan Carlos Castro y Castro 10 Sep '07

10 Sep '07

I know I can test the SIP return code with an "if", but how do I log it, or, alternatively, pass it to a exec_dset() script? I tried something like log(1, "Failure Route, status is $status\n"); And I get a "bad logargument" when I start openser. Clues? Juan

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[OpenSER-Users] Openser Signaling and Routing
by Kaeman Chris 08 Sep '07

08 Sep '07

Hello, Is this possible? Call comes to openser for handling signaling, and is handed over to B2BUA for topology hiding and media handling. But the routing(selection of gw to terminate to the supplier) is used from openser LCR. Or after the call is given to B2BUA the routing is to be performed by the B2BUA. Thanks!! Regards, KChris --------------------------------- 5, 50, 500, 5000 - Store N number of mails in your inbox. Click here.

2 1

[OpenSER-Users] Empty $du
by Andreas Sikkema 07 Sep '07

07 Sep '07

Hi, I'm trying to save the destination IP address of an INVITE message received and processed by OpenSER in the acc table. I've added an extra column and told the acc module about the new column. When I looked at the contents of the new column it was empty and adding an xlog() call to the config file also showed an empty $du pseudo variable. I know the OpenSER version I'm using is not the very latest, but it should be new enough to properly fill $du. I'm calling acc for received INVITE and BYE messages and showing $du with xlog() in the last line of INVITE processing for my testcase when the destination should be available. What am I doing wrong? -- Andreas Sikkema

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[OpenSER-Users] modparam("usrloc", "nat_bflag", 6) and setflag(6) doesn't update flags column
by Halomoan Chow 07 Sep '07

07 Sep '07

Dear all I followed tutorial from SER-Getting Start document for Mediaproxy to solve nat problem. I read on the Openser document http://www.openser.org/docs/modules/1.2.x/usrloc.html#AEN83 I have to use modparam("usrloc", "nat_bflag", 6) instead of modparam("registrar", "nat_flag", 6) And from the documentation, it said "When SIP clients attempt to REGISTER with our SIP proxy we need a way to tell the registrar module to store NAT information for any particular UA. We do this by using flag 6, which has been arbitrarily chosen (but defined earlier in the loadmodule parameter section). We could have specified another integer here, but flag 6 seems to be the accepted standard for nat_flag. If the nat_flag is set before calling the save() function to store contact information, SER will preserve the NAT contact information as well as set the flags column in the MySQL location table. By doing so, we can call lookup(location) when processing messages and flag 6 will be set for NATed clients." I have a NAT UA is detected as NAT-ed UA by Openser with below rule and do setflag(6); before it does save("location"); Why in mysql, location table, i see the flags column still 0 instead of 6 ? Did I miss something? Please help :( # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress= paul(a)onsip.org $ debug=3 fork=no log_stderror=yes #alias="halodns.powere2e.com" listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no 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/domain.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/xlog.so" loadmodule "/usr/local/lib/openser/modules/mi_fifo.so" modparam("mi_fifo","fifo_name","/tmp/openser_fifo") modparam("auth_db|domain|uri_db|usrloc", "db_url", " mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper|registrar", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) <- set to 6 modparam("rr", "enable_full_lr", 1) . . . . route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- xlog("L_NOTICE","$Cxg Enter Route[2]: REGISTER $Cxx\n"); sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("2")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; }

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[OpenSER-Users] uac_auth, how to sending request in failure route?
by Cseke Tamas 07 Sep '07

07 Sep '07

Hello, I 'd like to relay calls to another sip proxy, which requires authentication. But i can't forward the 407 reply to the caller, beacause they cant authenticate themselves, so i have to authenticate the call myself, without the caller knows anything about it. witch uac_auth, i can add the proxy-authorization header, but i can't send the new request to the proxy. i try this way: route[1] { t_on_failure("1"); rewritehostport("another.proxy:5060"); if (!t_relay()) { sl_reply_error(); } exit; } failure_route[1] { xlog("L_WARN", "FAILURE Route 3 AUTH required...trying..\n"); uac_auth(); if(t_newtran()) { xlog("L_WARN", "Creating new transaction successfully...\n"); } else { xlog("L_WARN", "Creating new transaction failed...\n"); }; t_relay(); } When the proxy challange authentication with 407, the call is passed to failure route properly, but t_relay failed with the following error: 0(27975) ERROR:tm:t_forward_nonack: no branch for forwarding 0(27975) ERROR:tm:w_t_relay: t_forward_nonack failed I think that's mean that the transaction doesn't exists, becacause it acknowleged alrealy, am i right? i think i have to t_newtran(), but it cannot be done in failure route, Can you suggest any method to send the request again with the credentials? Thanks any help, Tamas

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[OpenSER-Users] my small security breach REGISTER
by Marc LEURENT 07 Sep '07

07 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a security matter with my configuration (default one), it's possible to register using login/password and to set anything in the contact field. So if you have an account 106/password, it's possible to be 105 in the location database! How is it possible to deny that kind of matter..? Thanks Is it useful to use: method_filtering of the REGISTRAR module Or is it better to so something whith the values below and a compare function?? $ct - reference to body of contact header $ar - realm from Authorization or Proxy-Authorization header $au - username from Authorization or Proxy-Authorization header if ($ct != $au@$ar) { sl_send_reply("403", "User and login must be the same"); }; Best Regards, Marc LEURENT # U 82.127.0.79:1045 -> 88.191.45.91:5060 REGISTER sip:sd-7501.dedibox.fr;user=phone SIP/2.0. Via: SIP/2.0/UDP 82.127.0.79:1046;branch=z9hG4bK5808036470869310420. From: <sip:105@sd-7501.dedibox.fr:5060;user=phone>;tag=c0a80101-38c0e7. To: <sip:105@sd-7501.dedibox.fr:5060;user=phone>. Call-ID: 29eb6e9-c0a80101-5-17(a)192.168.95.70. CSeq: 90 REGISTER. Max-Forwards: 70. Expires: 3600. Contact: <sip:105@82.127.0.79:1046;user=phone>. Authorization: Digest username="106", realm="sd-7501.dedibox.fr", nonce="46dfceb402cad04812873b855bc50ea65aa99ed5", uri="sip:sd-7501.dedibox.fr", response="7dca83fd358a9aea3a963f4a71ea5c9e", algorithm=MD5, qop=auth, cnonce="38c102", nc=00000001. User-Agent: THOMSON ST2030 hw0 fw1.56 00-0E-50-4E-AF-C4. Allow-Events: refer,dialog,message-summary,check-sync,talk,hold. Content-Length: 0. . AOR:: 105 Contact:: sip:105@82.127.0.79:1046;user=phone Q= Expires:: 194 Callid:: 29eb6e9-c0a80101-5-17(a)192.168.95.70 Cseq:: 92 User-agent:: THOMSON ST2030 hw0 fw1.56 00-0E-50-4E-AF-C4 Received:: sip:82.127.0.79:1045 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:88.191.45.91:5060 Methods:: 4294967295 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG39AIqjpLE0HiOBYRAiUKAJ9Ilv+Zpbzw89tqWgwmHyVjU/DXugCgjEh8 5XQKEAeiF/L4RWszGC2/yzQ= =SXE9 -----END PGP SIGNATURE-----

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[OpenSER-Users] is_uri_host_local() returns 1 or 4294967295 (always TRUE !!!)
by Iñaki Baz Castillo 07 Sep '07

07 Sep '07

Hi, in order to minimize the SQL queries for functions "is_uri_host_local()" and "is_from_local()" I want to exec they just one time and store the result in a variable, but the result is unexpected: $var(is_uri_host_local) = is_uri_host_local(); xlog("------------is_uri_host_local = $var(is_uri_host_local)\n"); So if the URI is local then the result is 1. But if the URI is not local then result is 4294967295. ¿?¿?¿?¿ Obviously it makes me difficult for me to do: if ($var(is_uri_host_local)) because always is true and I need to convert the values and so. Why "is_uri_host_local()" doesn't return -1 in case of being not local domain? Or maybe the issue is when I store the result in a variable? but why? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Transaction Deletion
by Brandon Armstead 07 Sep '07

07 Sep '07

Hello Users, I have a weird scenario where the PSTN is responding with 200 OK, however Openser Never forwards on the ACK, after several retransmissions of 200OK to Openser, PSTN responds with BYE, anyone know what might be going on? I've played with the timers to no avail, any help is appreciated, thank you! The log below is as follows: 375 Sep 6 12:18:10 openser /usr/local/sbin/openser[17385]: DEBUG: timer routine:1,tl=0x2b78e4e73588 next=(nil), timeout=261 376 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: timer routine:2,tl=0x2b78e4e733b8 next=(nil), timeout=262 377 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: wait_handler : removing 0x2b78e4e73338 from table 378 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: delete transaction 0x2b78e4e73338 379 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: wait_handler : done 380 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG:parse_to:end of header reached, state=10 381 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: get_hdr_field: <To> [26]; uri=[sip:openser] 382 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: to body [<sip:openser> ] 383 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: get_hdr_body : content_length=0 384 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG:destroy_avp_list: destroying list (nil) 385 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG:forward_reply: found module tm, passing reply to it 386 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_check: start=0xffffffffffffffff 387 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: add_param: tag=gK0288a919 388 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG:parse_to:end of header reached, state=29 389 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: get_hdr_field: <To> [53]; uri=[sip:17144828306@openser] 390 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: to body [<sip:17144828306@openser>] 391 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: hash 53903 label 1526431020 branch 0 392 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: no matching transaction exists 393 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: failure to match a transaction 394 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_check: end=(nil) It seems like its not matching the transaction, because a timer is removing the transaction? Any ideas? Thanks! -- Brandon Armstead

2 1

[OpenSER-Users] Dispatcher + NAT
by kokoska rokoska 07 Sep '07

07 Sep '07

Hi all, I'm currently succesfully running OpenSER, but user population grow up and I'm affraid I need laod balancing :-) The projected scenario is: USERS <-> (NAT) <-> LB <-> P1 <-> GWs <-> P2 <-> R1 <-> R2 Where LB is Load Balancer (OpenSER with Dispatcher & Path module), and P1-P2 is proxies (OpenSER) and R1-R2 registrars (OpenSER) And the problem I need to solve is NAT handling. I can correctly process registrations with help of Path module on LB and enabling of using it on R1,R2. But, what should I do with calls? My idea is to process NAT handling on P1-P2, but I don't know how. Both proxies are unable (I mean) detect correct user IP address. Could you tell me, please, what to do? Do NAT handling on LB? It is unnecessary added load to LB... Thanx in advance! kokoska.rokoska

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[OpenSER-Users] is_from_user_enum() errors
by Eric Hiller 07 Sep '07

07 Sep '07

Hello all, I have a rather strange issue, I cannot use "is_from_user_enum()" without first using the variable $fu somewhere; or else I get the error "0(5858) get_parsed_from(): From header is not properly parsed". Originally I had an xlog there and it all worked fine, but when I took it away I started getting the error. Now I just have " $var(bleh) = $fu;" there and it makes it work fine. I had a forum post @ http://www.voipuser.org/forum_topic_10875.html that shows my entire config. Thanks for any help, Eric _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

2 1

[OpenSER-Users] CDRTool Price is all 0.0000
by Chahn Kim 06 Sep '07

06 Sep '07

Hi all. CDRTool shows all call records with price o.oooo. Syslog logs following: Error: Cannot find rates for callid=, domain=koreanet.us, destination=1847, profile=441, app=audio Can anyone help? Thanks in advance John Chahn Kim email: johnk(a)koreanet.us

1 0

[OpenSER-Users] Limiting maximum simultaneous calls per user
by Sebastian Mangelkramer 06 Sep '07

06 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear list, actually I'm searching for a solution to limit the max. simultaneous calls per user. I`ve searched the internet and list-archives for possible solutions. But until now i haven`t found a solution. What i exactly need is to define the max. simultaneous calls per user (e.g. 2,4,6,8,...). I know, that openSER is transaction-stateful, not call-stateful but is there a way to monitor the number of RTP-Sessions per user from Mediaproxy and set the "max_calls"-variable via an AVP? My setup is openSER 1.2.1-notls + Mediaproxy Has anyone implemented such a solution or anyone know such one? Perhaps i had to place a kind of SBC/B2BUA in front of my SIP-Proxy? Many thanks in advance, kind regards Sebastian -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (Darwin) iD8DBQFG3tU87EKQHe5FugIRAi8LAJ0eOVGO10FDjZcTMc7q24TXdfElDgCdHM0g l2ZanHEt/eX9yZTw/jVQxGc= =7Mz4 -----END PGP SIGNATURE-----

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[OpenSER-Users] OpenXCAP install and config problem?
by KevinKinnan 06 Sep '07

06 Sep '07

I am not sure whether or not my procedure is right as server doesnt work at the moment, but I will keep working on it. Here is the Fedora installation [1] Fedora 6 linux install; [2] pre-dependencies package; Because I installed OpenIMSCore and Openser Presence module before, so some dependencies was already installed. In this case I just installed clearsilver from Add&Remove Package, and copy lxml folder(1.3.4 download from lxml site) to /usr/lib/python2.5/site-packages/. [3] install OpenXCAP, followed installation guide/Debain unstable installation By now, that's it. Coz no daemon can be run from /etc/init.d/openxcap. Also, I haven't modified anything in the config.ini and openxcap files Cheers, Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

1 0

[OpenSER-Users] OpenXCAP install and config problem?
by KevinKinnan 06 Sep '07

06 Sep '07

I am trying to install openXCAP on the Fedora 6. Here is my installation procedure: python ./setup.py install; copy config.init to /etc/openserxcap/ copy openxcap to /etc/init.d/ run /etc/init.d/openxcap start update openser config file modparam("presence_xml", "force_active", 0); modparam("presence_xml", "integrated_xcap_server", 0); I havent change anything in config.ini and openxcap it shows some errors "start-stop-daemon command not found". It seems that the openxcap.ini is writed for Debian based Linux as start-stop-daemon is for Debain. How can I run the start file on a red hat based linux? Do I need to do any other modification in some files? Thank You! Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

2 1

[OpenSER-Users] Please help in Openser behind NAT design
by Halomoan Chow 06 Sep '07

06 Sep '07

Hi All I'm a new member and trying to use Openser as my VOIP "router" I plan to make following configuration: UA <---> Openser <---> Firewall/NAT <--INTERNET--> Firewall/NAT <---> UA a.a.a.a/24 x.x.x.x y.y.y.y b.b.b.b/24 a.a.a.a and b.b.b.b is internal IP not internet route-able x.x.x.x and y.y.y.y is Firewall/NAT external IP number I know there is mediaproxy and RTPProxy to solve the UA b.b.b.b that behind NAT but I don't know if my decision to put Openser behind NAT is correct or not. How to archive the above diagram with Openser ? If above Openser location is wrong, where should the Openser reside in the diagram? Install in Firewall/NAT or in INTERNET (dedicated server) Thank you for your help and sorry if my question is too newbie Halomoan

1 0

Re: [Serusers] Can't make a single call. Help me please.
by Mike Trest - Personal 06 Sep '07

06 Sep '07

As a long time VoIP and Internet engineer, we have all seen various port-blocking activities at different levels. These are almost always done to attain economic or political goals. (Remember PANAMA and SINGAPORE). Seldom is it ever for security reasons or for conservation of resources. It is the first question I ask whenever I start to assist a gateway operator who is testing via a cable operator. Therefore, it is understandable that we would be groping in the dark without that bit of bank ground information. Perhaps we should add a guideline note to our FAQ which recommends placing a sniffer/etherial between the originating sip device and the IP network as well as an ethereal/tcpdump on the SER end. ..mike.. At 01:34 AM 9/3/2007, Harmeet Singh wrote: >Dear Greg > >I tried ekiga, brujula, and couple other services on my internet >yesterday. None worked. Same problem - REGISTER is OK but can't make >calls. Then after searching google again, I found that my ISP (Rogers >Cable) is dropping/blocking the INVITE messages. That's why I was able >to register and not make any calls. Then I found on the internet about >a workaround, which is to not use 5060 port. So, I just tried that. To >my surprise, Any port other than 5060 just works! It beats me really. > >Thanks again for your help. I really appreciate it. > >Harmeet --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000771-1, 09/01/2007 Tested on: 9/6/2007 10:25:41 AM avast! - copyright (c) 1988-2007 ALWIL Software. http://www.avast.com

1 0

[OpenSER-Users] openSER rtpproxy video call - please help
by Volodymyr Fedyk 06 Sep '07

06 Sep '07

Hi, I've tried this openser configs from http://www.voip-info.org/wiki/view/OpenSER+v1.2.x+and+RTPProxy and generated by http://sipwise.com/wizard/ and I always face with the next issue: 2 UAs are behind NAT; each UA could call another; IM works; Audio works; but looks-like video streams are handled by openSER and rtpproxy incorrectly, and I'm unable to make a video call to any ua, - I'm using x-lit sp phone, and it shows "Could not start video" on one side, and "No matching video codec" on the other side... Has anybody faced with such issues later or has any ideas ? best regards, Dorin

3 2

[OpenSER-Users] Almost the End / End calls OpenSER + mediaproxy + freeradius
by Marc LEURENT 06 Sep '07

06 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello evrybody! I have one little problem... When a call badly ends (like hard disconnection or killall mysoftphone) mediaproxy has a timeout of 10sec, after the timeout, it sends a message to freeradius to stop accounting (so CDRTool billing too), but DON'T say to openser to stop the call, so the connection to my SIP/PSTN provider is still up (and its billing too) Is there any way to say to openser to send BYEs when mediaproxy detects the end of a badly ended call? Best Regards! PS: J-20 before VoN Europe 2007 - Rome Extract of the INSTALL file: """If the media session timeouts (there is no BYE that closes the session), MediaProxy will contact back the proxy dispatcher and pass the session media information to be logged. Using this architecture, only the proxy dispatcher needs to have direct access to the database. This addresses the issue of the MediaProxy servers being distributed and not having access to the database.""" -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3rpwqjpLE0HiOBYRAmbmAJ9UiDDi6UWYg4v+KLR020MwG/ks2ACggg8w aDQX87YSNxjEvy5vcSSeWto= =OCrH -----END PGP SIGNATURE-----

1 1

[OpenSER-Users] openser as proxy relay
by Asim Riaz 06 Sep '07

06 Sep '07

Hi, Could any body give me some idea to what need to be done in openser for relaying all SIP packets to another third party SIP registrar and Gateway. Scenario XYZ SIP Reg------------------openser----------------------UA Thanks in advance

3 2

[Serusers] serweb login problem
by Harmeet Singh 06 Sep '07

06 Sep '07

Hi I am facing this strange problem, if anyone can help me on this I will appreciate. Here is my configuration - Fedora 7, Apache 2.2.4, php 5.2.2, ser/serweb 0.9.6 When I go to localhost/serweb, enter login info, and submit, I get a download window pop-up asking me to save the php file. Shouldn't Apache be catching this and render the php to html? The login page itself is a php page and that renders ok. So, Apache is configured correctly for php rendering. What else am I missing here? Any help is appreciated. Thanks Harmeet

1 0

[OpenSER-Users] XCAP setting problem?
by Adrian Georgescu 05 Sep '07

05 Sep '07

You may obtain the XCAP server and the configurations from http:// openxcap.org The latest OpenSER svn version works with it and the sample configuration. Adrian >>>>> I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server? Thank You! Kevin

1 0

[OpenSER-Users] XCAP setting problem?
by KevinKinnan 05 Sep '07

05 Sep '07

The PUBLISH request works when using the following configurations(no XCAP): loadmodule "sl.so" loadmodule "mysql.so" loadmodule "presence.so" loadmodule "presence_xml.so" modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("presence_xml", "force_active", 1) modparam("presence_xml", "pidf_manipulation", 1) It doesn't if starting XCAP by updating these, the server can run properly, but cann't handle PUBLISH. loadmodule "xcap_client.so" modparam("presence_xml", "xcap_table", "xcap_xml") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "xcap_server", "localhost:5065") I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server? Thank You! Kevin _________________________________________________________________ Windows Live Spaces 中最年轻的成员！ http://miaomiaogarden2007.spaces.live.com/

1 0

[OpenSER-Users] mi_xmlrpc compilation issues
by Purna Chandar M 05 Sep '07

05 Sep '07

Hi, Im trying to compile mi_xmlrpc module in openser but could not succeed. The following errors are displayed mi_xmlrpc.c: In function 'xmlrpc_process': mi_xmlrpc.c:161: error: 'xmlrpc_server_abyss_rpc2_handler' undeclared (first use in this function) mi_xmlrpc.c:161: error: (Each undeclared identifier is reported only once mi_xmlrpc.c:161: error: for each function it appears in.) mi_xmlrpc.c:167: error: 'xmlrpc_server_abyss_default_handler' undeclared (first use in this function) Im using Suse 10.1; The libraries xmlrpc-c-0.9.10-36, libxml-1.8.17-385, libxml2-devel-2.6.23-13, xmlrpc-c-devel-0.9.10-36 are properly installed. Few discussions on the mailing list concluded that the module mi_xmlrpc compiles well on Debian without any dependency issues. Is it true?? Can somebody tell me whether Im missing any dependencies?? Please let me know how to successfully compile this module. Thanks and regards, Purna Chandar M

3 3

Re: [OpenSER-Users] mi_xmlrpc compilation issues
by Kriti Wadhwa 05 Sep '07

05 Sep '07

2 1

[OpenSER-Users] NatPing stop working
by Marcello Lupo 05 Sep '07

05 Sep '07

Hi to all, i have openser 1.2.1 active from some months.. Suddenly on yesterday the natping stop to work. I have 140 contacts in user location almost all of them natted. There is some limit to user in location table at which the ping stops or a particular server load? There is a way to have something in the logs about it?? I didn't found anything in the logs. This is my config: modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) With a restart of openser all come back to work but i'm afraid if it will happen again? Thanks for help, Bye, Marcello

2 14

[OpenSER-Users] Opnser rejecting BYE packet
by Dan 05 Sep '07

05 Sep '07

Scenario 1: U 212.34.169.220:5060 -> 78.45.146.34:5060 BYE sip:test@78.45.146.34 SIP/2.0. Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Content-Length: 0. # U 78.45.146.34:5060 -> 212.34.169.220:5060 SIP/2.0 404 User Not Found. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 req_src_ip=78.45.146.34 req_src_port=5060 in_uri=sip:78.45.146.34;ftag=aa313a3c;lr out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". Scenario 2: U 59.176.78.110:50496 -> 210.68.65.78:5060 BYE sip:210.68.65.78 SIP/2.0. Route: <sip:@209.67.171.10;transport=udp>. Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. User-Agent: Cisco ATA v2.15 ata18x (020927a). Content-Length: 0. . # U 210.68.65.78:5060 -> 59.176.78.110:50496 SIP/2.0 478 Unresolvable destination (478/TM). Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. In both cases BYE is not getting accounted for. Can anyone help me on this ?? Regards Dan

4 19

Re: [OpenSER-Users] Broken "BYE" returned from Asterisk on TLS implementation ?
by David Loh 05 Sep '07

05 Sep '07

Hi, Arrggghh .. that's one of my attempts to eliminate the broken "BYE" problem... that's ngrep was captured when I set "modparam("rr", "enable_double_rr", "0");", I've paste another ngrep to http://pastebin.ca/674450, this time the double RR header is enabled. And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt the post is "openser"). Even though double RR header is enabled, but for BYE it's still doesn't process properly :( For the .cfg file line #130 onward, I did tried t_relay, forward and force_send_socket, but none of this will do the trick (force_send_socket was complaining TLS error due to missing certificate (?) ) Would appreciate if anyone could enlighten me why is this happen ? Thanks, David Loh Klaus Darilion wrote: > But the INVITE you posted at http://pastebin.ca/673392 also has only > one Record-Route header. > > regards > klaus > > David Loh schrieb: >> Hi, >> >> Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK >> ...but when asterisk disconnected the call and send BYE back to OpenSER, >> the TLS RR header wasn't present, the only 2 RR header was >> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" .... >> I'm puzzled ... is there any command to 'fix' this? >> >> >> Regards, >> David Loh >> >> Klaus Darilion wrote: >>> The openser proxy should add 2 record-route header (TLS and UDP = >>> double record route). This is why it does not work. >>> >>> regards >>> klaus >>> >>> David Loh schrieb: >>>> Hi All, >>>> >>>> Greeting. >>>> >>>> I've been struggle with OpenSER TLS implementation for more than a >>>> week, since I've ported from UDP to TLS, everything work fine >>>> except the "BYE" request from Asterisk (loose route), my >>>> implementation was something like below: >>>> >>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] >>>> >>>> My OpenSER.cfg already configured to listen on two port which is :- >>>> "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or >>>> even voicemail) having no problem, >>>> but when the callee disconnect the call, caller will never get hang >>>> up :( >>>> >>>> I've attached my ethereal trace/ngrep to pastebin, >>>> http://pastebin.ca/673392 >>>> >>>> Wondering if anyone can help me with the broken "BYE" that returned >>>> from Asterisk ? >>>> Line #131, supposedly this line should have contain 2 Via header, >>>> one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", >>>> but somehow the TLS via header was gone !! (compare to previous ACK >>>> (Line #117) /INVITE (Line #51). >>>> Due to the missing TLS via header, OpenSER log file was complaining >>>> "protocol/port mis-match". >>>> >>>> The last BYE request (Line #256) is actually firing from Client, >>>> which contain the "TLS" via. >>>> >>>> >>>> I've even tried "force_send_socket" to port 5061 (instead of 5060) >>>> from loose route, but it complaining TLS certificate error, >>>> since Asterisk doesn't support TLS natively, I've no clue why is >>>> the ACK/INVITE/CANCEL work but not BYE. >>>> if (loose_route) { >>>> .... >>>> if(is_method("BYE")) { force_send_socket(IP:5061); } >>>> } >>>> >>>> >>>> Has any one gone through of this kinda OpenSER over TLS + Asterisk >>>> setup, >>>> I'm really appreciate if you can share your experience with me, or >>>> pin point what's the mistakes I made here. >>>> >>>> Thanks in advance. >>>> >>>> Regards, >>>> David Loh >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > >

2 5

[OpenSER-Users] OpenSER with PostgreSQL
by Cliff Flood 04 Sep '07

04 Sep '07

Hi, I have an Ubuntu 6.06 install with PostgreSQL 8.1 and OpenSER 1.2.2 which I am testing at the moment. I have tested OpenSER with MySQL and it worked fine but when when I want to switch to PostgreSQL I'm having some difficulty. I have used the postgressqldb.sh script to initialise the database and it has completed without error. When I start OpenSER I get the following lines in my PostgreSQL logs: LOG: connection received: host=127.0.0.1 port=37610 LOG: connection authorized: user=openser database=openser LOG: SSL SYSCALL error: EOF detected LOG: could not receive data from client: Connection reset by peer LOG: unexpected EOF on client connection I have looked at OpenSER's logs with debugging turned on but I don't see anything that looks like an error but I would be happy to provide them. `openserctl moni` returns: ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. My openser.cfg looks like this: debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 listen=192.168.20.151 dns=no rev_dns=no port=5060 mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "auth.so" loadmodule "auth_db.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) 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("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # 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("host.doman.tld", "subscriber")) { # www_challenge("host.domain.tld", "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; } Any and all help appreciated. -- Cliff Flood Systems Administrator Jazinga Inc.

2 2

[OpenSER-Users] Strange Provider SIP - DSP fields in trying
by Marc LEURENT 04 Sep '07

04 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...) When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced! OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP! Do you know how I can delete or modify TRYING/RINGING SDP? Thanks Below an example of a RINGING with SDP # U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712(a)192.168.95.47. Contact: <sip:212.129.6.65:5060>. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" <sip:mleurent@sd-7501.dedibox.fr>;tag=542924903. Record-Route: <sip:88.191.45.91;lr;ftag=542924903>. Server: Cirpack/v4.39a (gw_sip). To: <sip:0614730696@sd-7501.dedibox.fr>;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3TdoqjpLE0HiOBYRAjnkAKCgzB4p89oTKhxuvZcLlklTme9w3ACeP+TW FI+LkwgbaJ1XJTIm3FUSMTg= =Cp53 -----END PGP SIGNATURE-----

2 4

[OpenSER-Users] MediaProxy 1.9.0
by Mike O'Connor 04 Sep '07

04 Sep '07

Hi Guys I upgraded my mediaproxy to 1.9.0 and found out sometime later that the data usage update at call end seems to have stopped. I'm using direct mysql (not radius) for the end of call updates. No error message in the logs and a manual test of mysql show nothing wrong. Any idea. Mike

1 0

RE: [OpenSER-Users] Opnser rejecting BYE packet
by Dan 04 Sep '07

04 Sep '07

Hi Ravi, Did what you suggested, but still same error. Regards Dan _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: Tuesday, September 04, 2007 3:51 PM To: Dan Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hello Dan In yur openser.cfg In Below this this Section route { if(loose_route()) { # mention here BYE section also with fix_nated_contact(); } if(method=="CANCEL" ) { } else if (method=="INVITE") { } else if (method=="REGISTER") { } else if ( method=="BYE" ) # here u missing in this section route(1); exit; } On 9/4/07, Dan <fiedler.dan(a)gmail.com> wrote: Where exactly? Regards Dan _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: Tuesday, September 04, 2007 2:27 PM To: Dan Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hi Dan Problem in ur openser,cfg file On 9/4/07, Dan <fiedler.dan(a)gmail.com> wrote: My Config File: listen=72.65.56.136 #alias=0.0.0.0 alias=dstpl.com check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=6 fifo="/tmp/openser_fifo" fifo_db_url="postgres://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/postgres.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/uac.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 diest authentication mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" # loadmodule "/usr/local/lib/openser/modules/lcr.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" #loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "postgres://openser:openserrw@localhost/openser") # ----------------- setting module-specific parameters --------------- # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) modparam("maxfwd", "max_limit", 32) modparam("auth_radius","radius_config","/etc/radiusclient-ng/radiusclient.co nf") modparam("auth_radius", "service_type", 15) modparam("avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- Accounting param -- modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 16) modparam("acc", "radius_flag", 1) modparam("acc", "log_level", 1) # modparam("acc", "radius_missed_flag", 1) modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc modparam("acc", "radius_extra", "Sip-Source-IP-Address=$si") # -- 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("avpops", "avp_aliases", "c_uri=i:888; s_ip=i:999" ) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if((method=="INVITE") || (method=="ACK") || (method=="BYE") || (method=="CANCEL")) { setflag(1); } 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; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); exit; }; if (!uri==myself) { route(5); route(1); exit; }; if (uri==myself) { if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="BYE") { route(1); exit; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); exit; }; 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" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!radius_www_authorize(" <http://72.65.56.136> 72.65.56.136")) { www_challenge("72.65.56.136", "0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[3] { # ------------------------------------------------------------------------ # INVITE Handler # ------------------------------------------------------------------------ if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(5); route(1); exit; }; if (!allow_trusted() && uri=~"^sip:[0-9]*@.*") { if (!radius_www_authorize(" 72.65.56.136")) { www_challenge("72.65.56.136", "0"); exit; }; }; if (uri=~"^sip:[0-9]*@.*"){ route(11); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; if (method=="CANCEL") { route(1); exit; }; route(5); route(1); } route[11]{ # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- strip(0); prefix(""); rewritehostport(" 206.234.108.104:5060 <http://206.234.108.104:5060> "); route(5); route(1); } -----Original Message----- From: users-bounces(a)openser.org [mailto: <mailto:users-bounces@openser.org> users-bounces(a)openser.org] On Behalf Of Dan-Cristian Bogos Sent: Tuesday, September 04, 2007 1:30 PM To: users openser.org Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hello Inaki, yes, it is almost sure that it does (except receiving this answer from remote party). This is the reason all the people place loose-route block before location query and this is why I have assumed that it does not match the loose-routing and ends up to location query automatically. DanB On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Like Dan said, > send us your configuration, you may have forbidden something! > > Dan a écrit : > > *_Scenario 1:_* > > > > > > > > > > > > U 212.34.169.220:5060 -> 78.45.146.34:5060 > > > > BYE sip:test@78.45.146.34 SIP/2.0. > > > > Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. > > > > CSeq: 2 BYE. > > > > Via: SIP/2.0/UDP 212.34.169.220:5060. > > > > From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. > > > > Call-ID: 8246bf729f62a754. > > > > To: 58.68.96.11< sip:test@78.45.146.34 <mailto:sip:test@78.45.146.34> >;tag=aa313a3c. > > > > Content-Length: 0. > > > > > > > > # > > > > U 78.45.146.34:5060 -> 212.34.169.220:5060 > > > > SIP/2.0 404 User Not Found. > > > > CSeq: 2 BYE. > > > > Via: SIP/2.0/UDP 212.34.169.220:5060. > > > > From: < <mailto:sip:1217773456@78.45.146.34> sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. > > > > Call-ID: 8246bf729f62a754. > > > > To: 58.68.96.11< sip:test@78.45.146.34 <mailto:sip:test@78.45.146.34> >;tag=aa313a3c. > > > > Server: OpenSer (1.0.1 (i386/linux)). > > > > Content-Length: 0. > > > > Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 > > req_src_ip=78.45.146.34 req_src_port=5060 > > in_uri=sip:78.45.146.34;ftag=aa313a3c;lr > > out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". > > > > > > > > > > > > *_Scenario 2:_* > > > > > > > > U 59.176.78.110:50496 -> 210.68.65.78:5060 > > > > BYE sip:210.68.65.78 <http://210.68.65.78> SIP/2.0. > > > > Route: <sip:@209.67.171.10;transport=udp>. > > > > Via: SIP/2.0/UDP 59.176.78.110:50496 . > > > > From: sip:user1@210.68.65.78;tag=4154163258. > > > > To: > > <sip:12818495027@210.68.65.78 ;user=phone>;tag=3005360702044301069847989. > > > > Call-ID: 4202237689(a)192.168.1.156. > > > > CSeq: 3 BYE. > > > > User-Agent: Cisco ATA v2.15 ata18x (020927a). > > > > Content-Length: 0. > > > > . > > > > > > > > # > > > > U 210.68.65.78:5060 -> 59.176.78.110:50496 > > > > SIP/2.0 478 Unresolvable destination (478/TM). > > > > Via: SIP/2.0/UDP 59.176.78.110:50496. > > > > From: sip:user1@210.68.65.78;tag=4154163258. > > > > To: > > < <mailto:sip:12818495027@210.68.65.78> sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. > > > > Call-ID: 4202237689(a)192.168.1.156. > > > > CSeq: 3 BYE. > > > > Server: OpenSer (1.0.1 (i386/linux)). > > > > Content-Length: 0. > > > > Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 > > req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. > > > > > > > > > > > > In both cases BYE is not getting accounted for. Can anyone help me on > > this ?? > > > > > > > > > > > > Regards > > > > Dan > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org <http://enigmail.mozdev.org> > > iD8DBQFG3Q9/qjpLE0HiOBYRArgmAKCec0eFK2QQyBmV5eGoCWRmUOMM8ACgmFgp > tOKvu9D6afXK2f8if8FABnM= > =66HB > -----END PGP SIGNATURE----- > > _______________________________________________ > 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 -- Thanks &Regards Ravi Prakash Sunkara -- Thanks &Regards Ravi Prakash Sunkara

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[Serusers] SER+PATTON1+PATTON2+PATTON3
by pokerino76 04 Sep '07

04 Sep '07

I do not have the possibility to use Patton 4960 with 1xPRI. I wished an alternative method for Patton1 + patton2 + patton3: 8+8+8=32. thanks This is the code for 2 Patton Hi! On the pattons you can created Error Messages when not circuit channel is available And on Ser you have to register to Failure Route to the route of the first patton and the failure route point to the second Patton. PATTON1: profile sip default map causes to-sip no-circuit-channel-available to 486 map causes to-sip resources-unavailable to 486 service hunt-group HUNT-BRI cyclic timeout 1 drop-causes normal-unspecified drop-causes no-circuit-channel-available drop-causes network-out-of-order drop-causes temporary-failure drop-causes switching-equipment-congestion drop-causes access-info-discarded drop-causes circuit-channel-not-available drop-causes resources-unavailable route call 1 dest-interface ISDN_01 route call 2 dest-interface ISDN_02 route call 3 dest-interface ISDN_03 SER: Ser.cfg: if (uri=~ ". *sip: 012345 ") # Patton 1 {t_on_failure (1); t_relay_to_udp (11.11.11.10, 5060); break; } failure_route [1] { if (method== " INVITE ") { if (t_check_status (600|486)) #### ErrorCode from Patton 1 { #####call from Patton 1 -- 486 (Busy) ret -- call switch to Patton 2 ###### if (uri=~ ". *sip: 012345 ") # Costumer xy { append_branch (); t_relay_to_udp (11.11.11.11, 5060); ##Unterroute ist hier nicht notwendig, break; ##da sich die Verrechnung hier nicht aendert } } } Which code to use for 3 patton? ------------------------------------------------------ Leggi GRATIS le tue mail con il telefonino i-mode di Wind http://i-mode.wind.it/

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[OpenSER-Users] Openser rejecting BYE packet
by Dan 04 Sep '07

04 Sep '07

Scenario 1: U 212.34.169.220:5060 -> 78.45.146.34:5060 BYE sip:test@78.45.146.34 SIP/2.0. Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Content-Length: 0. # U 78.45.146.34:5060 -> 212.34.169.220:5060 SIP/2.0 404 User Not Found. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 req_src_ip=78.45.146.34 req_src_port=5060 in_uri=sip:78.45.146.34;ftag=aa313a3c;lr out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". Scenario 2: U 59.176.78.110:50496 -> 210.68.65.78:5060 BYE sip:210.68.65.78 SIP/2.0. Route: <sip:@209.67.171.10;transport=udp>. Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. User-Agent: Cisco ATA v2.15 ata18x (020927a). Content-Length: 0. . # U 210.68.65.78:5060 -> 59.176.78.110:50496 SIP/2.0 478 Unresolvable destination (478/TM). Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. In both cases BYE is not getting accounted for. Can anyone help me on this ?? Regards Dan

2 2

[OpenSER-Users] call duration limit
by Nick 'TARANTUL' Novikov 04 Sep '07

04 Sep '07

Hi, ppl! I have installed OpenSER and radius server. OpenSER authorized all call through radius. How I can limit call duration by "Session-Timeout" attribute from radius ? -- tarantul

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[Serusers] Route SIP Calls To Asterisk
by Kevin Kibuku 03 Sep '07

03 Sep '07

Hi Team, Please someone guide me in how I could forward for instance a call to 1111111 made on the SER to an * server. I have tried quite a lot of things but nothing seems to work. Thanks -- Kevin Kibuku, Intersat Africa Limited Internet Via Satellite Cell: +254720789158

2 2

[OpenSER-Users] t_relay after failure_route -- problem
by Zahid Mehmood 03 Sep '07

03 Sep '07

Hi All, I am trying to implement serial forking and am using the example on voice-systems website as a guide. The problem I am experiencing is as follows. Failure route rewrite ruri with the next uri to try and calls route[17] route 17 performs a lookup("location"), finds the usrloc information and rewrites ruri, however, when t_relay is called it uses the original ruri (one before the lookup) to do t_relay. Another issue I'm experiencing is that when processing call after failure route, ruri_user_avp. Once i rewrite the uri in failure route, I have to use the following to manually write a value for ruri_user_avp. Can someone explain why this is happening? Thanks in advance for your help!! -- Zahid ] Some relevant config and debug info: modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") route[18] { # used to process pstn/ off system calls ...... if (!is_avp_set("$avp(i:1402)")) { xlog("L_INFO","$ci: route[22] ** setting ruri_user_avp **\n"); $avp(i:1402) = $rU; xlog("L_INFO","$ci: route[22] ** setting ruri_user_avp = $avp (i:1402) **\n"); } ........ } #end route 18 route[11] { # load all numbers asscociated with the pilot number in $avp (serial_fork) array xlog("L_INFO","$ci: route[11] - M=$rm RURI=$ru F=$fu T=$tu IP=$si \n"); $avp(serial_fork) = "sip:*6" + $rU + "@sipdev.oursiphost.com"; if (avp_db_query("select username from fork where pilot = $rU order by pri" , "$avp(serial_fork)")) { xlog("L_INFO", "route[11] $ci: method $rm r-uri <$ru> \n"); xlog("L_INFO", "route[11] serial_fork = $avp (serial_fork) remote-user = $rU \n"); t_on_failure("5"); route(17); exit; } else { xlog("L_ERR", "route[11] failed to load serial_fork info.\n"); } avp_print(); # debugging only setflag(8); } # end of route[11] route[17] { xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si \n"); $avp(s:inv_timeout) = "10"; if (lookup("location")) { # currently online registered UA. xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si -- inside location lookup \n"); if (!t_relay()) { sl_reply_error(); }; return; } else { # not online if (does_uri_exist()) { # but exists xlog("L_INFO", "$ci: Subscriber is offline [$ru]\n"); sl_send_reply("480", "Temporarily Unavailable."); return; } xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si -- outside location lookup \n"); }; if (uri=~"^sip:(([0-9()+-])|(%20))+@") { subst_user('/[()+-]//g'); # strip out the punctuation. subst_user('/%20//g'); # strip out the white space. route(18); # all numerics just go to route(18) -- off system calls return; }; } # end route[17] failure_route[5] { xlog("L_INFO","$ci: failure_route[5] - M=$rm RURI=$ru F=$fu T=$tu IP=$si\n"); avp_print(); if (avp_pushto("$ruri", "$avp(serial_fork)")) { append_branch(); xlog("L_INFO","$ci: failure_route[5] $rm r-uri <$ru> from <$fu> to <$tu>\n"); avp_delete("$avp(serial_fork)"); t_on_failure("5"); route(17); } else { xlog("L_INFO","$ci: failure_route[5] $rm r-uri <$ru> from <$fu> to <$tu>\n"); } } Debug info Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: failure_route[5] - M=INVITE RURI=sip: 10512@10.1.12.172:8891 F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039c28, flags=0x0003 Aug 31 10:54:56 mousse openser[6815]: INFO: name=<inv_timeout> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<10 / 2> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039bf0, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10511@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039bb8, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10210@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b80, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10510@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b48, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:*610512@sipdev.oursiphost.com / 31> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b18, flags=0x0083 Aug 31 10:54:56 mousse openser[6815]: INFO: name=<is_pilot> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<y / 1> Aug 31 10:54:56 mousse openser[6815]: DEBUG:avpops:pushto_avps: 1 avps were processed Aug 31 10:54:56 mousse openser[6815]: ******* setting for branch 0 flags 0 Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: failure_route[5] INVITE r-uri <sip: 10511(a)sipdev.oursiphost.com> from <sip:10510@sipdev.oursiphost.com> to <sip:10512@sipdev.oursiphost.com;user=phone> Aug 31 10:54:56 mousse openser[6815]: DEBUG:avpops:delete_avps: 1 avps were removed Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: route[17] - M=INVITE RURI=sip: 10511(a)sipdev.oursiphost.com F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 Aug 31 10:54:56 mousse openser[6815]: rewrite_uri: Rewriting Request- URI with 'sip:10511@10.1.12.172:8891' Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: route[17] - M=INVITE RURI=sip: 10511@10.1.12.172:8891 F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 -- inside location lookup Aug 31 10:54:56 mousse openser[6815]: DEBUG: mk_proxy: doing DNS lookup... Aug 31 10:54:56 mousse openser[6815]: DEBUG:sip_resolvehost2: no port, no proto -> do NAPTR lookup! Aug 31 10:54:56 mousse openser[6815]: DEBUG:filter_and_sort_naptr: found valid SIP+D2U -> _sip._udp.sipdev.oursiphost.com Aug 31 10:54:56 mousse openser[6815]: DEBUG:do_srv_lookup: SRV (_sip._udp.sipdev.oursiphost.com) = mousse.cc.oursiphost.com:5060 Aug 31 10:54:56 mousse openser[6815]: DEBUG:sip_resolvehost2: found! Aug 31 10:54:56 mousse openser[6815]: check_via_address(10.1.12.174, 10.1.12.174, 0) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 500000 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[4]: 0xb6039eb4 (61500000) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 30 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[0]: 0xb6039ed0 (91) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:relay_reply: branch=0, save=1, relay=-1 >, CSEQ_N=<CSeq: 1> 2(a)sipdev.oursiphost.com;user=phone>ocal: using FROM=<From: "Zahid 10510" <sip: 10510(a)sipdev.oursiphost.com>;tag=BF787F6-F6DD44ED Aug 31 10:54:56 mousse openser[6815]: DEBUG: cancel_branch: sending cancel... Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 500000 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[4]: 0xb6039e20 (61500000) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 30 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[0]: 0xb6039e3c (91)

1 2

[OpenSER-Users] Bernard Le est absent(e).
by bernard.le＠orange-ftgroup.com 03 Sep '07

03 Sep '07

Je serai absent(e) du 03/09/2007 au 17/09/2007. Je répondrai à votre message dès mon retour. Contacter le service Ingénierie : Patrice Galleau au +33 1 44 55 69 36, patrice.galleau(a)orange-ftgroup.com ou Tony Capochichi au +33 1 44 55 66 34, tony capochichi(a)orange-ftgroup.com I will reply to your message when I'll be back. Please contact the Engineer department: Patrice Galleau at +33 1 44 55 69 36, patrice.galleau(a)orange-ftgroup.com or Tony Capochichi at +33 1 44 55 66 34, tony capochichi(a)orange-ftgroup.com ********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

1 0

[OpenSER-Users] Does Presence Module support 'policy document'?
by KevinKinnan 03 Sep '07

03 Sep '07

Does Presence Module support 'policy document'? That means the user can do some settings to forbid specified watcher or something like this. Thank you! Kevin _________________________________________________________________ Windows Live Custom Domain，您的免费电子邮局。 https://domains.live.com/default.aspx

2 1

[OpenSER-Users] OpenXCAP launch
by Adrian Georgescu 03 Sep '07

03 Sep '07

Hello, I would like to announce the launch of OpenXCAP.org OpenXCAP is an open source, easy extensible, fully featured XCAP server with TLS security and support for multiple realms. OpenXCAP server can be used in combination with a SIP server that supports PUBLISH/SUBSCRIBE/NOTIFY methods to provide a complete SIP SIMPLE server solution The server is written in the Python programming language and is based on the following RFCs: - The Extensible Markup Language Configuration Access Protocol RFC 4825 - Formats for Representing Resource Lists RFC 4826 - Usage for Manipulating Presence Document Contents RFC 4827 - Presence Authorization Rules draft-ietf-simple-presence-rules-10 The server supports TLS security, multiple back-end storage systems and works out of the box with OpenSER Proxy/Registrar/Presence server. The software is provided under the BSD license. I wish to thank here to my IETF friends who helped me with advice, testing and guidance during the development phase of this project. I would also like to thank to Mircea Amarascu, the author of the software, without his passion and dedication this project would not have been possible. The project is hosted at: http://openxcap.org The project is supported via the wiki collaboration system setup by AG Projects. To open ticket please Register first. We are looking forward your feedback from real-life field deployments. Kind regards, Adrian Georgescu AG Projects

2 1

[OpenSER-Users] Media control with openser
by srinivas Antarvedi 03 Sep '07

03 Sep '07

Hello all, Actually i have openser box which i used to make calls among my offices which spread different geographical places. i am using in the configuration file just to make changes to my rtp packets so without RTPPROXY and MEDIAPROXY modules i was able to make calls even my end UAC are behind NAT. but if i want to control my media between different UAs like a prepaid system as we used to see in case of mobile prepaid system which stops you from continuing your call when there was no balance left on ur mobile . can anybody give me a hint how can i do that ? if so what architecture it needs like any mediaproxy? will session border controller help me out? just give me a hint so that i can go ahead thanks once again -- Srinivas Antarvedi

3 2

[OpenSER-Users] Wanted: VoIP Engineer for Warsaw!
by laurent schweizer 03 Sep '07

03 Sep '07

Peoplefone AG jest firmą oferującą usługi telekomunikacyjne oparte na technologii Voice over IP (VoIP) w wyjątkowo niskich stawkach. Peoplefone jest certyfikowanym partnerem firm Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>i AVM/FRITZ!Box <http://www.fritz-shop.ch/>. W związku z naszym dynamicznym rozwojem, do nowo tworzonej spółki w Polsce poszukujemy osoby na stanowisko: SPECJALISTA D/S VOIP Miejsce pracy: Warszawa *Wymagania:* - wykształcenie wyższe techniczne (lub w trakcie studiów), najlepiej informatyczne - praktyczna umiejętność programowania w środowisku w PHP i językach C i C++ - znajomość Linux - znajomość mySQL i PostgreSQL - znajomość zagadnień związanych z sieciami IP, UDP, TCP - znajomość narzędzi do analizy sieci np.. Ethereal - wiedza z zakresu VoIP, serwerów i konfiguracji oprzyrządowania - umiejętność rozwiązywania problemów informatycznych - znajomość języka angielskiego w mowie i piśmie w stopniu zaawansowanym - umiejętność szybkiego uczenia się - chęć podnoszenia swoich kwalifikacji zawodowych - profesjonalne i rzetelne podejście do obowiązków służbowych, umiejętność pracy w zespole, samodzielność, komunikatywność, systematyczność, terminowość - doświadczenie zawodowe mile widziane *Dodatkowe wymagania (mile widziane):* - umiejętność programowania w PERL i JAVA - znajomość Asterisk / SER / OPENSER - umiejętność konfigurowania bramek i ruterów Cisco, Patton, lub innych - znajomość protokołów SIP lub STUN - znajomość problematyki NAT I funkcjonowania outband Proxy - znajomość narzędzi monitorujących jak Cactus, Nagios, MRTG - znajomość narzędzi jak DRBD, Hearthbeat - dyspozycyjność Wszystkich zainteresowanych prosimy o przesłanie życiorysu oraz listu motywacyjnego (zawierającego zgodę na przetwarzanie danych osobowych) *do 10 września br. ** *na adres: *ewa.ciesla(a)peoplefone.com* W temacie wiadomości wysyłanej pocztą elektroniczną prosimy umieścić nazwę stanowiska. Skontaktujemy się tylko z wybranymi kandydatami. Prosimy o dopisanie następującej klauzuli "Wyrażam zgodę na przetwarzanie moich danych osobowych zawartych w mojej ofercie pracy dla potrzeb niezbędnych do realizacji procesu rekrutacji (zgodnie z ustawą z dn. 29.08.97 roku o Ochronie Danych Osobowych Dz. Ust Nr 133 poz. 883)".

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Re: [Serusers] Can't make a single call. Help me please.
by Harmeet Singh 03 Sep '07

03 Sep '07

It's strange, but I do see the SIP INVITE messages being sent in the ethereal log. Why they don't reach to the server? SER is connected directly to the internet, without any NAT. On the other hand, REGISTER messages do get to SER and the client recieves the 200 OK and all intermediate replies as well. I have attached the clientside ethereal log for you to see. I am lost. Harmeet

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[Serusers] exec_dset
by Jesús Terán 02 Sep '07

02 Sep '07

Hello, I am using the exec_dset for access to my database mysql and change the uri in the calls, but I think this use more CPU processing, Are there other module to do the same? thanks Jesus Teran B.

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Re: [OpenSER-Users] OpenSER to PSTN
by Ricardo.ferreira 02 Sep '07

02 Sep '07

1 0

[OpenSER-Users] OpenSER to PSTN
by Arya 01 Sep '07

01 Sep '07

Hi Everyone Is there any way to send all the calls that come to OpenSER to PSTN lines (PRI, BRI) ? -- Thank You

2 1

Re: [OpenSER-Users] checking radius return attributes
by danish.samad＠vocalseeds.com 01 Sep '07

01 Sep '07

Hi Dan, Thanks for all your help and support, I really appreciate it. I was able to get it to work. Although I had to replace all VSA'a with the SIP-AVP attributes. For some reason the SIP-AVP's were not loaded due to the presence of other VSA's. Maybe it was because I was using a = sign in the value prefixes instead of : or #. Is this a known issue. Regards, Danish ps: Someone asked earlier about the radius server I was using. The radius server along with the billing solution is from advancedvoip.com, so far our experience with them has been quite good. > Hi Danish, > > sorry for replying to you so late, but I was out from the office today. > > Here you go: > > First of all, my bad, the returned value should be in the form: > return-code#0 if you want your value to be treated as an integer or > return-code:0 if 0 the AVP should be considered as string - has been a > while since I worked last time with those. By using a higher layer of > debug you should be able to see your avps converted inside when > received from radius. > The rest of your configuration looks good. Note that you can return > more AVP values with the same reply, so you can include the credit > amount as well: > SIP-AVP="return-code#0" > SIP-AVP="h323-credit-amount#4206" > > Hope you will rock now. > > Cheers, > DanB > > On 8/30/07, danish.samad(a)vocalseeds.com <danish.samad(a)vocalseeds.com> > wrote: >> Hi Dan, >> To test the functionality of SIP-AVP, I am trying to get theses avp's in >> the registration block, shown below. Basically on sucessful registration >> the radius server returns the following attributes in acces-accpet >> >> Attributes: >> SIP-AVP = "return-code=0" >> cisco-h323-credit-amount = "h323-credit-amount=4206" >> >> so after registration I am trying to get the return code value. The >> registration block is shown below >> >> route[2] { >> # ----------------------------------------------------------------- >> # REGISTER Message Handler >> # ---------------------------------------------------------------- >> xlog("L_NOTICE","Danish: Entering route[2]\n"); >> xlog("L_NOTICE","Danish: Register message IPsrcaddr [$si], >> RecvdIPaddr[$Ri]\n"); >> if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { >> xlog("L_NOTICE","Danish: In route[2] Found Nated >> contact\n"); >> setflag(6); >> setbflag(6); >> # if you want OPTIONS natpings uncomment next >> setbflag(7); >> fix_nated_register(); >> force_rport(); >> }; >> >> sl_send_reply("100", "Trying"); >> if (!radius_www_authorize("192.168.0.99")) { >> www_challenge("192.168.0.99", "1"); >> }; >> consume_credentials(); >> >> #avp_print(); >> xlog("L_NOTICE","Danish: Before is set check\n"); >> if(is_avp_set("$avp(s:return-code)")){ >> xlog("L_NOTICE","Danish: return code is set \n"); >> }; >> xlog("L_NOTICE","Danish: After is set check\n"); >> if(avp_check("$avp(s:return-code)", "eq/i:0/g")) >> { >> xlog("L_NOTICE","Danish: return code 0 received \n"); >> }; >> >> if (!save("location")) { >> sl_reply_error(); >> }; >> >> } >> >> In my logs I see >> >> 058) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 >> 4(11058) Danish: Before is set check >> 4(11058) Danish: After is set check >> 4(11058) DEBUG:avpops:ops_check_avp: no src avp found >> >> so both conditional avp checks fail and the last debug line is a bit >> worrying >> I traced the first debug line above to sterman.c->generate_avps() >> so just under this debug line I added this piece of code >> >> vp=rc_avpair_get(vp,attrs[A_SIP_AVP].v,0); >> if (vp != NULL) >> { >> DBG("DEBUG:auth_radius:generate_avps: strvalue %s\n", >> vp->strvalue); >> } >> else >> DBG("DEBUG:auth_radius:generate_avps: vp is null\n"); >> >> and this is what I got >> >> 4(11058) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair >> 225 >> 4(11058) DEBUG:auth_radius:generate_avps: vp is null >> >> hence openser is not loading the SIP-avp in $avp(s:return-code). I think >> either I am missing something in the openser script or the way I am >> returning the attribute. >> >> all relevant definitions in the cfg are given below: >> >> loadmodule "/usr/local/lib/openser/modules/auth_radius.so" >> loadmodule "/usr/local/lib/openser/modules/uri_radius.so" >> loadmodule "/usr/local/lib/openser/modules/avp_radius.so" >> loadmodule "/usr/local/lib/openser/modules/avpops.so" >> >> modparam("auth_radius|uri_radius|avp_radius", "radius_config", >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >> modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") >> >> modparam("uri_db|usrloc", "db_url", >> "mysql://openser:openserrw@localhost/openser") >> >> modparam("nathelper", "natping_interval", 20) >> modparam("nathelper", "ping_nated_only", 1) >> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") >> modparam("nathelper", "received_avp", "$avp(i:42)") >> modparam("nathelper", "sipping_bflag", 7) >> modparam("nathelper", "sipping_from", "sip:pinger@abc.org") >> >> modparam("registrar", "received_avp", "$avp(i:42)") >> modparam("usrloc", "db_mode", 2) >> modparam("usrloc", "nat_bflag", 6) >> >> Hope this explains everything and you would be in a better position to >> help. >> >> Regards, >> Danish >> ps: isnt there any sample script I can refer to for sip-avp radius >> configurations. >> >> >> > Hi Danish, >> > >> > can u post the full block where you are doing these checks? >> > >> > DanB >> > >> > On 8/29/07, danish.samad(a)vocalseeds.com <danish.samad(a)vocalseeds.com> >> > wrote: >> >> Hi Dan, >> >> Thanks for the explanation, things are more clear now. Now I am >> >> returning >> >> a SIP-AVP from radius (for authentication), this is what I see in the >> >> logs >> >> >> >> Attributes: >> >> SIP-AVP = "return-code=0" >> >> ....... >> >> >> >> In my script I do >> >> avp_print(); >> >> if(is_avp_set("$avp(s:return-code)")){ >> >> xlog("L_NOTICE","return code is set \n"); >> >> }; >> >> if(avp_check("$avp(s:return-code)", "eq/0/g")) >> >> { >> >> xlog("L_NOTICE","return code 0 received \n"); >> >> }; >> >> >> >> none of the checks pass and nothing is printed from avp_print. In >> >> openser >> >> logs I see this line >> >> >> >> 1(32550) DEBUG:auth_radius:generate_avps: getting SIP AVPs from >> avpair >> >> 225 >> >> >> >> am I missing some pre-requisites or what am I doing wrong. >> >> >> >> Regards, >> >> Danish >> >> >> >> >> >> > Hi Danish, >> >> > >> >> > for the moment you cannot process attributes other than SIP-AVP >> inside >> >> > openser script. This one is automatically converted into avp. >> >> > EG. A SIP-AVP attribute in the form of: "reason='No Credit'" should >> >> > automatically create an AVP named reason with value "No Credit". >> >> > >> >> > Hope that helps, >> >> > DanB >> >> > >> >> > >> >> > On 8/29/07, danish.samad(a)vocalseeds.com >> <danish.samad(a)vocalseeds.com> >> >> > wrote: >> >> >> Hi Dan, >> >> >> >> >> >> Thanks for your reply. >> >> >> >> >> >> Actually I am currently returning an integer status value in >> >> >> cisco-h323-return-code (a vendor specific attribute). So basically >> >> how >> >> >> can >> >> >> I extract the value returned in this attribute, compare it and >> then >> >> send >> >> >> and sl reply. >> >> >> >> >> >> Probably there is a way of loading this attr value in an avp and >> >> >> comparing >> >> >> it? >> >> >> >> >> >> Regards, >> >> >> Danish >> >> >> >> >> >> > Hello Danish, >> >> >> > >> >> >> > a solution would be returning a SIP-AVP, containing the reason >> for >> >> >> > failure, and then forward this to the client using sl. >> >> >> > Another one would be completely rewriting R-URI again from your >> >> >> > SIP-AVP and then send the request to an announcement server (eg: >> >> >> > ann:paymentrequired@your.server - which should play Payment >> >> Required >> >> >> > annoucement). >> >> >> > >> >> >> > Cheers, >> >> >> > DanB >> >> >> > >> >> >> > On 8/29/07, danish.samad(a)vocalseeds.com >> >> <danish.samad(a)vocalseeds.com> >> >> >> > wrote: >> >> >> >> Hi, >> >> >> >> >> >> >> >> I have configured openser-1.2 with a (commercial) third party >> >> radius >> >> >> >> server and it is working great. Now I have a requirement where >> I >> >> need >> >> >> to >> >> >> >> check a radius return attribute in case an access reject is >> >> received >> >> >> in >> >> >> >> response to an authorization request. Is there some way I can >> do >> >> >> this >> >> >> >> in >> >> >> >> the openser script. >> >> >> >> >> >> >> >> Basically I want to send a 402 payment required message to a >> sip >> >> >> client >> >> >> >> in >> >> >> >> case the call is rejected due low balance. In such cases radius >> >> >> returns >> >> >> >> a >> >> >> >> status response code with access reject. >> >> >> >> >> >> >> >> >> >> >> >> Thanks, >> >> >> >> Danish >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Users mailing list >> >> >> >> Users(a)openser.org >> >> >> >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> >> > >> >> >> >> >> >> >> >> > >> >> >> >> >> > >> >> >

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[OpenSER-Users] checking radius return attributes
by danish.samad＠vocalseeds.com 01 Sep '07

01 Sep '07

Hi, I have configured openser-1.2 with a (commercial) third party radius server and it is working great. Now I have a requirement where I need to check a radius return attribute in case an access reject is received in response to an authorization request. Is there some way I can do this in the openser script. Basically I want to send a 402 payment required message to a sip client in case the call is rejected due low balance. In such cases radius returns a status response code with access reject. Thanks, Danish

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Re: [Serusers] INFO Method
by Greger Viken Teigre 01 Sep '07

01 Sep '07

As I wrote: you don't, you need another component to do that. SER has a management interface that can be used to inject a message, but you still need an external component to do that. g-) ------- Original message ------- From: kapilg <kapilg(a)amdale.com> Sent: 31.8.'07, 13:34 > Oh Sorry! Actually I am using SER. > > > Well, if one has to send INFO using ser then how one can send? > > > > > Regards, > > > Kapil > > > > --------------- > > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Friday, August 31, 2007 4:19 PM > To: kapilg > Cc: serusers(a)iptel.org > Subject: Re: [Serusers] INFO Method > > > > > Wrong list, this is the ser, not openser list. > A proxy just proxies, it doesn't create messages. But maybe your media server? > g-) > > kapilg wrote: > > > Dear Users, > > > > > Could any one tell or give me one sample of sending INFO Method with MSCML as Message Body part? > > > I am working on one project in which XLite makes a call to a Media Server using openser proxy and after some seconds delay proxy forms one packet of INFO Method filling MSCML script in this and send to Media Server. This will be an ivr call. > > > > > Any help will be appreciated. > > > > > Thanks and Regards, > > > Kapil Goel > > > > > > --------------- > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Serusers] Can't make a single call. Help me please.
by Harmeet Singh 31 Aug '07

31 Aug '07

Hi allAfter spending whole week reading and googling I am writing here. Hope someone from here can help me.Here is my setup -I have a Fedora 7 desktop with SER, mysql, and rtpproxy running. This is connected directly to the cable modem. There is not NAT in between. Firewall and SELinux are disabled.I have X-Lite client on my Laptop connected to a Wi-Fi router on a different IP. This router has NAT but the firewall is disabled in the router.On the far end, my sister has X-Lite on her desktop which is also behind NAT.Both X-Lite clients can REGISTER and the SIP server sends 200 OK. But can't make calls.When all clients and SER are on the same network with local IP addresses, I can make calls between the X-Lite clients and SJPhone on different computers. No problem. And Grandstream phones also work in the local setup.Any help is appreciated.HS

2 2

[OpenSER-Users] avp_substr, re-index's lists?
by Brandon Armstead 31 Aug '07

31 Aug '07

Hello, Say I have something like the following: $avp(a[0]) = "27*" $avp(a[1]) = "22*" Then I run avp_substr like the following: avp_subst("$avp(a)/g", "/\*/[0-9]/g"); As you would expect it to simply strip * from the avp list values, instead it decides to rewrite the index in numerical order, so the avp's would now look like: [ BAD RESULTS ] $avp(a[0]) = "22"; $avp(a[1]) = "27"; however the [ EXPECTED RESULTS ] $avp(a[0]) = "27"; $avp(a[1]) = "22"; Could someone please elaborate on why OpenSER would re-numerate the list index on a simple avp_substr, also any ideas on how I can get around this and stop this from happening? Thanks!

3 3

[OpenSER-Users] Openser for voip+presence+IM
by Alejandro Cabrera Obed 31 Aug '07

31 Aug '07

Dear all, I'm using Debian Etch + Asterisk 1.2.13 as our VoIP server. I have several clients talking to each other but now I need to implement presence+instant messaging also. People told me that Asterisk does not support presence+IM, so I have to use another SIP server like OpenSER. I've never used OpenSER so I need to know the following: Is Openser enough to get just several X-Lite (softphones) clients using voip+presence+IM, without connecting to the PSTN ??? Or do I need the integration with Asterisk to get what I need ??? Really thanks Alejandro

2 1

[OpenSER-Users] Hiding topology between the end customer and supplier?
by Kaeman Chris 31 Aug '07

31 Aug '07

Hello, What is the best solution to use openser to hide the topology between the customer and supplier? Thanks. KChris. --------------------------------- Why delete messages? Unlimited storage is just a click away.

2 1

[OpenSER-Users] End calls - Send UPDATE
by Marc LEURENT 31 Aug '07

31 Aug '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good morning everybody, I have one problem, I have set up an architecture with OpenSER + Freeradius + CDRTool + rtpproxy + mysql and it works, thanks to you and special thanks to Dan-Christian Bogos (with his freeradius-cdrtool module)!! My problem, is that when a call don't end properly (example: hard disconnection or kill of the SIP client process), no BYE request are sent, and no accounting stop! So the cdr are not generated! So I was wondering if someone knew how to deals with it, like sending SIP UPDATE with openser (like SIP OPTIONS PING for keeping NAT traversal)??? Have a nice day! Marc LEURENT -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG186EqjpLE0HiOBYRAnNsAJ9EKyWLnFq6CnI9/X/gkwpeqi1LhwCgquq/ axFFAoAbDaMD3x4YOzERzE0= =tgmf -----END PGP SIGNATURE-----

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[OpenSER-Users] [OT] VON OpenSer course: Registration issues
by Iñaki Baz Castillo 31 Aug '07

31 Aug '07

Hi, first of all sorry for this OffTopic. I've tryed to contact VON organizers for my question but got no response, maybe people here can help me. It's just about the VON OpenSer course registration process. I filled the register form and select PayPal pay type. I wasn't answered about my PayPal account and so, and I got: * Registration Summary: Confirmation#: 60130446 Conference Type: OpenSER Admin Training Sessions Payment type: PayPal Ends with: Amount Charged: $315.00 Amount Paid: $0.00 Date: No more, I know no more. Could someody explain me when should I pay for the course and how can I know if I'm registered or not? I've already bought the plain tickets so... XD Thanks a lot and regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

6 8

Re: [OpenSER-Users] proxy1(client)-->Openser-->proxy2(supplier)
by Dan-Cristian Bogos 31 Aug '07

31 Aug '07

Hi Kaeman, as per the logs you have sent me (please in the future keep users(a)openser.org on cc so others can use also the results of our time invested work), it looks like you are not handing properly the RTP part. You have a combination between proxy and non proxy of RTP. If you check your packets you will see that for the case of INVITE from RemoteProxy1-> OpenSER -> RemoteProxy2, you will rewrite the connection IP with the OpenSER IP Address (RTP Sockets) so the RemoteProxy2 will think that he must speak with OpenSER, but for the case of 183 replied back (200 OK, same case) you will not rewrite the connection IP, and the RemoteProxy1 - call originator, will have the feeling that he must send the RTP packets directly to RemoteProxy2. Therefore you must decide the scenario you want: proxy all through your openser (rtpproxy) or proxy nothing and let the parties speak directly to each other and have only signaling passing through your openser server. Since you don't have a NAT case here, I thing it should be pretty easy to do the correct setup. Hope you did get my point. Cheers, DanB On 8/30/07, Kaeman Chris <kaemanchris(a)yahoo.co.in> wrote: > > Hi Daniel, > > I agree with you:). thanks for your reply. I see life in me NOW. > > Ok, find the debug with ngrep. I did that grepping the port 5060. Is that > OK? Let me know if you need with any specific keyword. > > Attached you will find the debug file. Thanks!! > > Regards, > KChris. >

2 2

Re: [OpenSER-Users] OpenSER and SIPPY b2bua
by vallimamod abdullah 31 Aug '07

31 Aug '07

Hi, On 8/29/07, TCB <tawandac(a)gmail.com> wrote on behalf of users-request(a)openser.org: > Anyone out there managed to use sippy b2bua with openser . Any pointers > would be most welcome. I am currently using Sippy as a light b2bua to control my gateway and it is working pretty well. There is unfortunately very little documentation on it but it looks like the project is not dead (cf. http://www.b2bua.org). Can you be more precise on what you want to know ? -- Regards, - vma .

4 4

[Serusers] SIP registration failed, status=408 (Request Timeout)
by 朗雀通信服务有限公司 31 Aug '07

31 Aug '07

hi: I installed ser-0.9.6 on centos 5. I start ser use ser -D -E. ser -D -E Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 121.14.136.107 [121.14.136.107]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 121.14.136.107 [121.14.136.107]:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: localhost:5060 udp: localhost.localdomain:5060 When i registered with sip ua but the "SIP registration failed, status=408 (Request Timeout)" return. The annex is the ser.cfg file and tested result with sipsak.

2 1

[Serusers] how to enable logs in SER
by Sandeep.U.Sharma＠relianceada.com 31 Aug '07

31 Aug '07

Can some help me on this critical issue. I'm using ser.0.9.6 version of SER on Solaris10 (sparc). how can we enable error & info logs through syslogd utility of UNIX? My another query "Is SER config Build system available for ser version 0.9.6" Sandeep Sharma The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

2 1

[Serusers] INFO Method
by kapilg 31 Aug '07

31 Aug '07

Dear Users, Could any one tell or give me one sample of sending INFO Method with MSCML as Message Body part? I am working on one project in which XLite makes a call to a Media Server using openser proxy and after some seconds delay proxy forms one packet of INFO Method filling MSCML script in this and send to Media Server. This will be an ivr call. Any help will be appreciated. Thanks and Regards, Kapil Goel

2 1

[Serusers] ser command fails
by Adeyeye O. Michael 31 Aug '07

31 Aug '07

Thanks, object files successfully created but ser command now found. Moreso, it prompted that i could use sudo apt-get install. if i do so, there might be conflict i think. besides when i did sudo.... before now, it appeared the server was running but no call went through. What could be wrong? *see below *micadeyeye@asmicom:/usr/src/sip_router$ ls action.c data_lump_rpl.c forward.d Makefile.sources proxy.c route_struct.o tcp_read.d action.d data_lump_rpl.d forward.h md5.c proxy.d rpm tcp_read.o action.h data_lump_rpl.h forward.o md5.d proxy.h script_cb.c tcp_server.h action.o data_lump_rpl.o freebsd md5global.h proxy.o script_cb.d test AUTHORS db gentoo md5.h pt.h script_cb.h test.c.bak ccopts.sh debian globals.h md5.o qvalue.c script_cb.o timer.c ccver.sh doc hash_func.c md5utils.c qvalue.d scripts timer.d cfg.lex dprint.c hash_func.d md5utils.d qvalue.h ser timer.h cfg.tab.c dprint.d hash_func.h md5utils.h qvalue.o ser.8 timer.o cfg.tab.d dprint.h hash_func.o md5utils.o README ser.cfg.5 TODO cfg.tab.h dprint.o INSTALL mem README.cfg socket_info.c trim.h cfg.tab.o dset.c ip_addr.c mod_fix.h README-MODULES socket_info.d tsend.c cfg.y dset.d ip_addr.d modparam.c re.c socket_info.h tsend.d ChangeLog dset.h ip_addr.h modparam.d receive.c socket_info.o tsend.h comp_defs.h dset.o ip_addr.o modparam.h receive.d solaris tsend.o config.h error.c ISSUES modparam.o receive.h sr_module.c types.h COPYING error.d lex.yy.c modules receive.o sr_module.d udp_server.c crc.c error.h lex.yy.d msg_translator.c re.d sr_module.h udp_server.d crc.d error.o lex.yy.o msg_translator.d re.h sr_module.o udp_server.h crc.h etc lock_alloc.h msg_translator.h re.o stats.c udp_server.o crc.o examples locking.h msg_translator.o resolve.c stats.d unixsock_server.c CREDITS fastlock.h lock_ops.h name_alias.h resolve.d stats.h unixsock_server.d CVS fifo_server.c lump_struct.h netbsd resolve.h stats.o unixsock_server.h daemonize.c fifo_server.d main.c NEWS resolve.o str.h unixsock_server.o daemonize.d fifo_server.h main.d openbsd route.c tags.h usr_avp.c daemonize.h fifo_server.o main.o parser route.d tcp_conn.h usr_avp.d daemonize.o flags.c Makefile pass_fd.c route.h tcp_init.h usr_avp.h data_lump.c flags.d Makefile.defs pass_fd.d route.o tcp_main.c usr_avp.o data_lump.d flags.h Makefile.modules pass_fd.h route_struct.c tcp_main.d ut.h data_lump.h flags.o Makefile.radius pass_fd.o route_struct.d tcp_main.o utils data_lump.o forward.c Makefile.rules profile route_struct.h tcp_read.c version.h micadeyeye@asmicom:/usr/src/sip_router$ ser The program 'ser' is currently not installed. You can install it by typing: sudo apt-get install ser Make sure you have the 'universe' component enabled bash: ser: command not found * *

2 1

[Serusers] ./ser command gives this
by Adeyeye O. Michael 31 Aug '07

31 Aug '07

micadeyeye@asmicom:/usr/src/sip_router$ ./ser ERROR: loading config file(/usr/local/etc/ser/ser.cfg): No such file or directory micadeyeye@asmicom:/usr/src/sip_router$ cd /usr/local/etc/ser Truly, the folder doesn't exist on my laptop but does on my desktop. Am lost!!! Mike

2 1

[OpenSER-Users] cvs co -r rel_0_9_ sip_router < Installation failure>
by Adeyeye O. Michael 31 Aug '07

31 Aug '07

Hi, Please see below for my terminal output. First, I installed SER using sudo apt-get install ser, it indicated that it was working but no call was accepted by it. I saw SIP register via wireshark. THen I removed it and tried this second way. Now I tried installing from source also but appears futile. Some files are missing like deamonize.d???? ANY WAY OUT PLEASE?? *TERMINAL SCREEN OUTPUT* micadeyeye@asmicom:/usr/src/sip_router$ ls action.c data_lump_rpl.h gentoo md5global.h proxy.h scripts timer.h action.h db globals.h md5.h pt.h ser.8 TODO AUTHORS debian hash_func.c md5utils.c qvalue.c ser.cfg.5 trim.h ccopts.sh doc hash_func.h md5utils.h qvalue.h socket_info.c tsend.c ccver.sh dprint.c INSTALL mem README socket_info.h tsend.h cfg.lex dprint.h ip_addr.c mod_fix.h README.cfg solaris types.h cfg.y dset.c ip_addr.h modparam.c README-MODULES sr_module.c udp_server.c ChangeLog dset.h ISSUES modparam.h re.c sr_module.h udp_server.h comp_defs.h error.c lock_alloc.h modules receive.c stats.c unixsock_server.c config.h error.h locking.h msg_translator.c receive.h stats.h unixsock_server.h COPYING etc lock_ops.h msg_translator.h re.h str.h usr_avp.c crc.c examples lump_struct.h name_alias.h resolve.c tags.h usr_avp.h crc.h fastlock.h main.c netbsd resolve.h tcp_conn.h ut.h CREDITS fifo_server.c Makefile NEWS route.c tcp_init.h utils CVS fifo_server.h Makefile.defs openbsd route.h tcp_main.c version.h daemonize.c flags.c Makefile.modules parser route_struct.c tcp_read.c daemonize.h flags.h Makefile.radius pass_fd.c route_struct.h tcp_server.h data_lump.c forward.c Makefile.rules pass_fd.h rpm test data_lump.h forward.h Makefile.sources profile script_cb.c test.c.bak data_lump_rpl.c freebsd md5.c proxy.c script_cb.h timer.c micadeyeye@asmicom:/usr/src/sip_router$ make all Makefile.rules:88: action.d: No such file or directory Makefile.rules:88: crc.d: No such file or directory Makefile.rules:88: daemonize.d: No such file or directory Makefile.rules:88: data_lump.d: No such file or directory Makefile.rules:88: data_lump_rpl.d: No such file or directory Makefile.rules:88: dprint.d: No such file or directory Makefile.rules:88: dset.d: No such file or directory Makefile.rules:88: error.d: No such file or directory Makefile.rules:88: fifo_server.d: No such file or directory Makefile.rules:88: flags.d: No such file or directory Makefile.rules:88: forward.d: No such file or directory Makefile.rules:88: hash_func.d: No such file or directory Makefile.rules:88: ip_addr.d: No such file or directory Makefile.rules:88: main.d: No such file or directory Makefile.rules:88: md5.d: No such file or directory Makefile.rules:88: md5utils.d: No such file or directory Makefile.rules:88: modparam.d: No such file or directory Makefile.rules:88: msg_translator.d: No such file or directory Makefile.rules:88: pass_fd.d: No such file or directory Makefile.rules:88: proxy.d: No such file or directory Makefile.rules:88: qvalue.d: No such file or directory Makefile.rules:88: re.d: No such file or directory Makefile.rules:88: receive.d: No such file or directory Makefile.rules:88: resolve.d: No such file or directory Makefile.rules:88: route.d: No such file or directory Makefile.rules:88: route_struct.d: No such file or directory Makefile.rules:88: script_cb.d: No such file or directory Makefile.rules:88: socket_info.d: No such file or directory Makefile.rules:88: sr_module.d: No such file or directory Makefile.rules:88: stats.d: No such file or directory Makefile.rules:88: tcp_main.d: No such file or directory Makefile.rules:88: tcp_read.d: No such file or directory Makefile.rules:88: timer.d: No such file or directory Makefile.rules:88: tsend.d: No such file or directory Makefile.rules:88: udp_server.d: No such file or directory Makefile.rules:88: unixsock_server.d: No such file or directory Makefile.rules:88: usr_avp.d: No such file or directory Makefile.rules:88: mem/f_malloc.d: No such file or directory Makefile.rules:88: mem/mem.d: No such file or directory Makefile.rules:88: mem/memtest.d: No such file or directory Makefile.rules:88: mem/q_malloc.d: No such file or directory Makefile.rules:88: mem/shm_mem.d: No such file or directory Makefile.rules:88: mem/vq_malloc.d: No such file or directory Makefile.rules:88: parser/hf.d: No such file or directory Makefile.rules:88: parser/msg_parser.d: No such file or directory Makefile.rules:88: parser/parse_allow.d: No such file or directory Makefile.rules:88: parser/parse_content.d: No such file or directory Makefile.rules:88: parser/parse_cseq.d: No such file or directory Makefile.rules:88: parser/parse_disposition.d: No such file or directory Makefile.rules:88: parser/parse_diversion.d: No such file or directory Makefile.rules:88: parser/parse_event.d: No such file or directory Makefile.rules:88: parser/parse_expires.d: No such file or directory Makefile.rules:88: parser/parse_fline.d: No such file or directory Makefile.rules:88: parser/parse_from.d: No such file or directory Makefile.rules:88: parser/parse_hname2.d: No such file or directory Makefile.rules:88: parser/parse_hostport.d: No such file or directory Makefile.rules:88: parser/parse_methods.d: No such file or directory Makefile.rules:88: parser/parse_nameaddr.d: No such file or directory Makefile.rules:88: parser/parse_param.d: No such file or directory Makefile.rules:88: parser/parser_f.d: No such file or directory Makefile.rules:88: parser/parse_rpid.d: No such file or directory Makefile.rules:88: parser/parse_rr.d: No such file or directory Makefile.rules:88: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:88: parser/parse_to.d: No such file or directory Makefile.rules:88: parser/parse_uri.d: No such file or directory Makefile.rules:88: parser/parse_via.d: No such file or directory Makefile.rules:88: parser/digest/digest.d: No such file or directory Makefile.rules:88: parser/digest/digest_parser.d: No such file or directory Makefile.rules:88: parser/digest/param_parser.d: No such file or directory Makefile.rules:88: parser/contact/contact.d: No such file or directory Makefile.rules:88: parser/contact/parse_contact.d: No such file or directory Makefile.rules:88: db/db.d: No such file or directory Makefile.rules:88: db/db_fifo.d: No such file or directory Makefile.rules:88: lex.yy.d: No such file or directory Makefile.rules:88: cfg.tab.d: No such file or directory bison -d -b cfg cfg.y make: bison: Command not found make: *** [cfg.tab.c] Error 127 micadeyeye@asmicom:/usr/src/sip_router$ cd ..

3 2

[OpenSER-Users] Problem in XMPP module - Open SER xmpp gateway
by Aditya Chaturvedi 31 Aug '07

31 Aug '07

I have openser and jabberd running on the same server. After doing configurations described in documentation of XMPP module (http://www.openser.org/docs/modules/1.2.x/xmpp.html ) I am able to send message from SIP client to XMPP client but not able to send it backwards address of XMPP client : username*xmpp_host@gateway_domain address of SIP client : sip_user_name*openser_domain@xmpp_domain I have seen in log and it seems that message is coming to xmpp module(when sending from xmpp client) but it is unable to send to SIP client. Please have a look at attached log and please help : LOG dump ----------------------------------------------- Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: xmpp: server read [<message xmlns='jabber:client' type='chat' xml:lang='en' id='sd10' to='aditya*192.168.2.14(a)sipxmpp.catch22' from='nitin(a)xmpp.catch22/Pandion'><body>3333333333333333333333333333333</body><html xmlns='http://jabber.org/protocol/xhtml-im'><body xmlns=' http://www.w3.org/1999/xhtml'><span style='font-weight: normal; font-size: 9pt; color: #004200; font-style: normal; font-family: arial'>3333333333333333333333333333333</span></body></html><x xmlns='jisp:x:jep-0038'><name>shinyicons</name></x><active xmlns=' http://jabber.org/protocol/chatstates'/></message>] Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: xmpp: stream callback: 1: message Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: DEBUG:tm:t_uac: next_hop=< sip:aditya@192.168.2.14> Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: DEBUG: mk_proxy: doing DNS lookup... Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: DEBUG:destroy_avp_list: destroying list (nil) Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: DEBUG: dlg2hash: 35729 Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: print_request_uri: sip:aditya@192.168.2.14 Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: DEBUG:tm:set_timer: relative timeout is 500000 Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: DEBUG: add_to_tail_of_timer[4]: 0xb612897c (66815700000) Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: DEBUG:tm:set_timer: relative timeout is 30 Aug 31 10:47:26 catch22 /usr/sbin/openser[17071]: DEBUG: add_to_tail_of_timer[0]: 0xb6128998 (66845) Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: SIP Request: Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: method: <MESSAGE> Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: uri: < sip:aditya@192.168.2.14> Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: version: <SIP/2.0> Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: parse_headers: flags=2 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: Found param type 232, <branch> = <z9hG4bK19b8.405ac901.0>; state=16 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: end of header reached, state=5 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: parse_headers: Via found, flags=2 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: parse_headers: this is the first via Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: After parse_msg... Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: preparing to run routing scripts... Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: parse_headers: flags=100 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:parse_to:end of header reached, state=9 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DBUG:parse_to: display={}, ruri={sip:aditya@192.168.2.14} Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: get_hdr_field: <To> [25]; uri=[sip:aditya@192.168.2.14] Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: to body [ sip:aditya@192.168.2.14^M ] Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: get_hdr_field: cseq <CSeq>: <10> <MESSAGE> Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: get_hdr_body : content_length=31 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: found end of header Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: is_maxfwd_present: max_forwards header not found! Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: add_param: tag=d536f7aa2bf73a9942d6233ad6d8ee06-0364 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:parse_to:end of header reached, state=29 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DBUG:parse_to: display={}, ruri={sip:nitin*xmpp.catch22@sipxmpp.catch22} Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: parse_headers: flags=200 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: find_first_route: No Route headers found Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: loose_route: There is no Route HF Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: grep_sock_info - checking if host==us: 12==12 && [192.168.2.14] == [192.168.2.14] Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: grep_sock_info - checking if port 5060 matches port 5060 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: grep_sock_info - checking if host==us: 12==12 && [192.168.2.14] == [192.168.2.14] Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: grep_sock_info - checking if port 5060 matches port 5060 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:avpops:check_avp: check <192.168.2.14> against <sipxmpp.catch22> as str /1 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:avpops:check_avp: no match Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: lookup(): 'aditya' Not found in usrloc Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: grep_sock_info - checking if host==us: 12==12 && [192.168.2.14] == [192.168.2.14] Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: grep_sock_info - checking if port 5060 matches port 5060 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: t_newtran: T on entrance=0xffffffff Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: parse_headers: flags=ffffffffffffffff Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: parse_headers: flags=78 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: t_lookup_request: start searching: hash=35729, isACK=0 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: RFC3261 transaction matching failed Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: t_lookup_request: no transaction found Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: mk_proxy: doing DNS lookup... Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: check_via_address( 192.168.2.14, 192.168.2.14, 0) Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DBG:check_against_rule_list: using list dns Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:tm:set_timer: relative timeout is 500000 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: add_to_tail_of_timer[4]: 0xb613508c (66815700000) Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:tm:set_timer: relative timeout is 30 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG: add_to_tail_of_timer[0]: 0xb61350a8 (66845) Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:tm:t_relay_to: new transaction fwd'ed Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:tm:UNREF_UNSAFE: after is 0 Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: DEBUG:destroy_avp_list: destroying list (nil) Aug 31 10:47:26 catch22 /usr/sbin/openser[17056]: receive_msg: cleaning up Thanks in advance ~Aditya

2 1

[OpenSER-Users] PDT Module - R-URI translation
by Noel R. Morais 31 Aug '07

31 Aug '07

Hi all, I am trying to use PDT module to translate an URI user@domain to user@IP. But it doesn't accept regex for the domain and prefix. Does somebody know a workaround or another module that works for this purpose? What I am trying to do is something like that: source domain | prefix | destination domain --------------------------------------------------------------------------------- example.com voicemail XXX.XXX.XXX.XXX example.com * YYY.YYY.YYY.YYY example2.com 5511[2-9]{4}[0-9]{4} XXX.XXX.XXX.XXX * voicemail XXX.XXX.XXX.XXX * * YYY.YYY.YYY.YYY thanks, Noel

2 1

[OpenSER-Users] Openser 1.3 with Asterisk 1.4 as mediaserver - any experience on call park/pickup and music on hold
by FRITTELLA FABIO 30 Aug '07

30 Aug '07

Hello, can anyone has esperience on configure asterisk 1.4 with openser 1.3 to allow call park/pickup and MOH ? Thanks. Fabio Frittella

1 0

[OpenSER-Users] can a branch route block be called from another branch route block?
by jh＠tutpro.com 30 Aug '07

30 Aug '07

lets say that two branch route blocks have a common piece of code. can duplicating of this code be avoided by including the common code in a third branch route block, which is then called from the other two branch route blocks or is there some other means to avoid the duplication? i tried to find this out from the core cookbook, but the issue was not discussed. -- juha

2 3

[OpenSER-Users] Problem with Media for the proxies making calls
by Kaeman Chris 30 Aug '07

30 Aug '07

Hello, I am having problem with Media - for one scenario; This scenario works for signaling and media both; softphone --> openser --> proprietory softswitch Caller Via Called Status Duration Codec Type Traffic -------------------------------------------------------------------------------------------- -------- softphoneIP:port - openserIP:port - softswitchIP:port active 0'08" g711 Audio 0/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.9.0 This scenario give PROBLEMS with MEDIA whereas signaling works OK: softphone --> proprietory softswitch --> openser --> proprietory softswitch Below is the output of /usr/local/mediaproxy/sessions.py which shows the called media ip is not seen by openser which results into no audio.. Caller Via Called Status Duration Codec Type Traffic -------------------------------------------------------------------------------------------- -------- spftswitchIP:11920 - openserIP:60008 - ?.?.?.? inactive 0'05" Unknown Audio 0/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.9.0 Here the above called party media shows ? which is the problem. Where i should look as the problem? Please help me. thanks! Kchris --------------------------------- Once upon a time there was 1 GB storage in your inbox. Click here for happy ending.

1 0

[OpenSER-Users] openser 1.2.2 segmentation fault
by Marco 30 Aug '07

30 Aug '07

Hi all, I'm trying to run openser 1.2.2 on Linux, but I obtain alway a segfault while loading module. The gdb shows: Program terminated with signal 11, Segmentation fault. #0 pg_free_row (_row=0x785bc8) at db_res.c:575 575 switch (VAL_TYPE(_val)) { I've a complex configuration, but the problem arise early so I don't think it's a configuration issue. The last messages I got in the log are: openser[31824]: PG[free_query]: PQclear(0x863a70) result set openser[31824]: PG[free_columns]: Freeing RES_NAMES(0x785b20)[0] -> free(0x785b90) 'table_version' openser[31824]: PG[free_columns]: 0x785b90=pkg_free() RES_NAMES[0] openser[31824]: PG[free_columns]: 0x785b60=pkg_free() RES_NAMES openser[31824]: PG[free_columns]: 0x785b78=pkg_free() RES_TYPES openser[31824]: PG[free_rows]: Freeing 1 rows openser[31824]: PG[free_rows]: Row[0]=0x785bc8 The last query I could see in the database is "select table_version from version where table_name='domain'" (it returns 1) I use postgres 8.2.4 on Linux 2.6.18 with xen, and gcc 4.1.3. Could someone suggest how I can investigate further? TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain http://people.equars.com/blogs/marco

3 5

RE: [OpenSER-Users] nat pinging question
by Papadopoulos Georgios 30 Aug '07

30 Aug '07

you are right, thanks a lot! > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Thursday, August 30, 2007 2:53 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] nat pinging question > > maybe you are loading mediaproxy module too which also sends > keep alives? > > regards > klaus > > Papadopoulos Georgios schrieb: > > Hello all, > > > > In nathelper the documentation mentions two pinging types > (upd 4 byte > > packets and stateless SIP request). Is there a way to > choose between > > one or the other or are they both activated by default? > > I have configured nathelped to ping with OPTIONS every 15 seconds. > > modparam("nathelper", "natping_interval", 15) modparam("nathelper", > > "sipping_method", "OPTIONS") In addition to the OPTIONS > packets that > > are sent every 15 seconds, there is a 4 byte packet that is > sent every > > minute. Why does this happen and is there a way to turn it off? > > > > This is what the ping packet sequence looks like: > > time ping > > 0 ---> 4 byte pkt > > 0 ---> OPTIONS > > 15 ---> OPTIONS > > 30 ---> OPTIONS > > 45 ---> OPTIONS > > 60 ---> 4 byte pkt > > 60 ---> OPTIONS > > > > thank you > > > > George > > > > > > > > > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > addressee(s). > > If you are not the intended recipient, or person responsible for > > delivering this information to the intended recipient, > please notify > > the sender immediately. Unless you are the intended recipient or > > his/her representative you are not authorized to, and must > not, read, > > copy, distribute, use or retain this message or any part of > it. E-mail > > transmission cannot be guaranteed to be secure or error-free as > > information could be intercepted, corrupted, lost, > destroyed, arrive > > late or incomplete, or contain viruses. > > > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] nat pinging question
by Papadopoulos Georgios 30 Aug '07

30 Aug '07

Hello all, In nathelper the documentation mentions two pinging types (upd 4 byte packets and stateless SIP request). Is there a way to choose between one or the other or are they both activated by default? I have configured nathelped to ping with OPTIONS every 15 seconds. modparam("nathelper", "natping_interval", 15) modparam("nathelper", "sipping_method", "OPTIONS") In addition to the OPTIONS packets that are sent every 15 seconds, there is a 4 byte packet that is sent every minute. Why does this happen and is there a way to turn it off? This is what the ping packet sequence looks like: time ping 0 ---> 4 byte pkt 0 ---> OPTIONS 15 ---> OPTIONS 30 ---> OPTIONS 45 ---> OPTIONS 60 ---> 4 byte pkt 60 ---> OPTIONS thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[OpenSER-Users] New module - carrierroute - in OpenSER
by Henning Westerholt 30 Aug '07

30 Aug '07

Hello everybody, i'm happy to announce the addition of a new module for OpenSER called "carrierroute". This module provides routing, balancing and blacklisting capabilities, from a config file or a database source. It can uses one routing tree, or if needed for every user a different routing tree for number prefix based routing. It supports several route tree levels, e.g. for failback routes. This modules scales up to more than a million users, and is able to handle more than 10k routing table entries. It should be able to handle more, but this is untested at the moment. Routing tables can be reloaded and edited (in config file mode) with the MI interface, the config file is updated according the changes. This is not implemented for the db interface, because its easier to do the changes directly on the db. But the reload and dump functions works of course here too. Basically the module could be used as an replacement for the lcr and the dispatcher module, if you have certain performance, flexibility and/or integration requirements that these modules don't handle properly. But for small installations it probably make more sense to use the lcr and dispatcher modules. Please refer to the documentation at http://www.openser.org/docs/modules/devel/carrierroute.html for further informations, the source code of the modules is available in the svn trunk. Best regards, Henning Westerholt -- Henning Westerholt - Unix system development 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[OpenSER-Users] proxy1(client)-->Openser-->proxy2(supplier)
by Kaeman Chris 30 Aug '07

30 Aug '07

Hello, I have a problem while handling media with openser when the call arrives from a proxy as shown in the following scenario; proxy1(client) --> Openser (No Audio on the call)--> proxy2(supplier) Where as the same scenario with the call originating from the user from the local domain/realm works perfect with this scenario. user@openserdomain --> openser --> proxy2(supplier) After spending the whole day with googling the stuff on the openser sites and others, i am in a vision that Openser with the mediaproxy module can handle only user on the local domain for media. Please help me to understand is the first scenario possible to have audio? thanks!! Hope to hear soon.Thanks!! Regards, KChris --------------------------------- Download prohibited? No problem. CHAT from any browser, without download.

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[OpenSER-Users] OpenSER Admin Training Session at VoN Europe 2007 Conference, Rome, Italy, September 26
by Alicia Codrington 30 Aug '07

30 Aug '07

Good morning I wish to register for the above and need to pay via wire transfer. Please forward the relevant details in order that all relevant registrations for staff can be expedited. Thank you so much for your speedy response. Cordially ________________________________ your internet phone company www.ctpcom.net <http://www.ctpcom.net/> www.calltheplanet.com <http://www.calltheplanet.com/> Alicia Codrington Office Administrator [ p ] +1-246-228-2159 [ f ] +1-246-430-1418 [ @ ] alicia(a)calltheplanet.com ________________________________

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[OpenSER-Users] Re: LCR not running
by jh＠tutpro.com 29 Aug '07

29 Aug '07

martin, were you able to solve your "lcr not running" problem? i tried to understand how your gw and lcr tables looked like, but due to line breaks was not able to figure it out. if you still have the "no gateways" found problem, please email to the list your table contents in readable form (also telling null and empty string from apart). -- juha

1 0

[OpenSER-Users] To URI
by Marc LEURENT 29 Aug '07

29 Aug '07

Hello everybody, I have booked fights, hotel and tickets for the Rome VON meeting!! I hope most of you will be there! I have a little problem... My OpenSER server has the IP 88.191.45.91 My SIP/PSTN gateway has the IP 212.129.6.65 I do not require authentication to send INVITE from my openser IP. How is it possible to change the To: <sip:0677832974@sd-7501.dedibox.fr:5060;user=phone>. to To: <sip:0677832974@212.129.6.65:5060;user=phone>. ?? Thanks!! See you soon in Rome # U 88.191.45.91:5060 -> 212.129.6.65:5060 INVITE sip:0677832974@212.129.6.65:5060 SIP/2.0. Record-Route: <sip:88.191.45.91;lr=on;ftag=c0a80101-1ca265a>. Via: SIP/2.0/UDP 88.191.45.91;branch=z9hG4bKd06b.6af66bc6.0. Via: SIP/2.0/UDP 192.168.95.81:5060;rport=62704;received=81.57.0.22;branch=z9hG4bK4296575315214209209. From: "Marc LEURENT"<sip:mleurent@sd-7501.dedibox.fr:5060;user=phone>;tag=c0a80101-1ca265a. To: <sip:0677832974@sd-7501.dedibox.fr:5060;user=phone>. Call-ID: 7b7f245-c0a80101-0-20(a)192.168.95.81. CSeq: 1 INVITE. Max-Forwards: 50. Supported: timer, replaces. Session-Expires: 1800. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO. Contact: <sip:mleurent@81.57.0.22:62704;user=phone;nat=yes>. User-Agent: THOMSON ST2030 hw0 fw1.56 00-0E-50-4E-AF-C4. Allow-Events: refer,dialog,message-summary,check-sync,talk,hold. Content-Type: application/sdp. Content-Length: 295. . v=0. o=mleurent 30025307 30025307 IN IP4 192.168.95.81. s=-. c=IN IP4 88.191.45.91. t=0 0. m=audio 35154 RTP/AVP 8 0 18 4 97. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:97 telephone-event/8000. a=fmtp:97 0-15. a=sendrecv. a=nortpproxy:yes.

2 3

[Serusers] why combine ser with asterisk
by Nhadie 29 Aug '07

29 Aug '07

Hi All, What's the advantage of combining ser with asterisk? I always see comments like using ser with asterisk is a very good solution etc. etc. the thing i liked with ser is that it does not do codec translation, which saves me cpu usage and also bandwidth. if i combine it with asterisk, would it not use codec translation? i also read that there is also a problem when ser and asterisk is run on the same machine, why is it so? if use prepaid billing solution for asterisk like astcc, would i then be able to provide prepaid service? soryy for asking too much, i'd just like to really understand it. Thank You in advanced. Regards, Nhadie

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[Serusers] Calls Per Second
by Jesús Terán 29 Aug '07

29 Aug '07

Hello, I want to know How many simultaneous calls can i have with the SER? I believe this depends with the Fisical Memory and Processor, i did a test with a traffic generator with a SER installed in a PC with 150 MB of RAM and Pentium 3 with 650 MHZ, and with 15 calls per second the CPU goes 100% and goes down. What kind of server do you recomend us to get a lot of calls per second? Att: Jesus Teran

3 2

[OpenSER-Users] I've questions about OpenSER
by mario gu 29 Aug '07

29 Aug '07

Dear All, I'm newbody on OpenSER, and i've some questiones about OpenSER as follow, 1. Can OpenSER run over Linux Redhat 9.0? If yes, how do I do? 2. From overview of OpenSER, I know that OpenSER can act as SIP dispatcher server, can you share more detail about it? Appreciate you very much in advance. Mario Gu mariogu2004(a)hotmail.com _________________________________________________________________ 与联机的朋友进行交流，请使用 MSN Messenger: http://messenger.msn.com/cn

2 1

[OpenSER-Users] avp_check("$avp(some_avp)", "re/$avp(some_regex_avp)/g"), use AVP as Regex?
by Brandon Armstead 29 Aug '07

29 Aug '07

Hello, Is it possible to use an AVP as a regex string? Everytime I get 'regexp operation requires string value'. I'm looking for a way to compare a value against a list of regex strings... i.e. avp_check("$avp(some_avp)", "re/$avp(some_regex_avp)/g"); when $avp(some_regex_avp[0]) is equal to a regex string, $avp(some_regex_avp[1]) is equal to some regex string, and so on and so forth, any help would be appreciated, thank you!

2 1

[OpenSER-Users] Interop with PSTN gateway sometimes fails - openser1.2.1
by Naija Man 29 Aug '07

29 Aug '07

Hi all, I am a beginner in openser and I really appreciate the help I have been getting from this group. I am setting up an interop with a PSTN gateway with intermittent success. I will try to explain my setup below. redirect server / / 2 / / 3 1 / / 4 5 asterisk -----> openser (v1.2.1) -----> pstnGateway ------> PSTN (cell phone) Calls are generated from asterisk (step 1) with a 3-digit prefix (say, 999) on the ANI to identify a particular group of calls, and sent to openser ( v1.2.1). Openser checks the from_uri; detects the 999 prefix, strips the leading 3 digits and forwards the INVITE to the redirect server (step 2). The redirect server responds with a 301, containing the location of the pstnGateway (step 3). Openser sends an INVITE to address in the 301 to the pstnGateway (step 4) and the call gets completed (step 5) This setup works intermittently. I sometimes have to restart openser before it works again. When it works, I get a 100 trying message back from the PSTN gateway, but when it fails, I see a number of invites sent from openser to the gateway without getting a 100 trying message back. And the invite times out with a 408 message. After a few seconds, the PSTN number actually rings, but I get dead air. My configuration is as below: ********************************************************** loadmodule "uac_redirect.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/etc/openser_fifo") modparam("rr", "enable_full_lr", 1) modparam("tm", "wt_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "fr_timer", 2) modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("uac","from_restore_mode","auto") # main routing logic route { if (!mf_process_maxfwd_header("16")) { xlog("L_INFO", " Reply: 483 - Too Many Hops\n"); sl_send_reply("483","Too Many Hops"); exit; }; if ( msg:len > max_len ) { xlog("L_INFO", " Reply: 513 - Message Too Big\n"); sl_send_reply("513", "Message too big"); exit; }; if (loose_route()) { record_route(); t_relay(); exit; }; record_route(); # Outgoing route route(1); route(1); } # ROUTE 1 - OUTGOING CALLS route[1] { # Check prefix - if the ANI of the tenant starts with 999 # and the total number of digits is more than 12 if ((from_uri=~"^sip:999.*")&&$(fu{s.len}) > 12) { # replace both display and uri # Strip out the leading 999 digits before sending it to redirect server uac_replace_from("$(fU{s.substr,3,0})","sip:$(fU{s.substr ,3,0})@$fd"); # Check for re-directs t_on_failure("4"); xlog("L_INFO", "RequestUri:[$ru] FromUri:[$fu] forwarding to redirect server...\n"); ds_select_dst("6", "0"); # forward to redirect server t_relay(); exit; } } } ############################################################################### # Process Replies ############################################################################### onreply_route[1] { if( status =~ "18[0-9]" ) { # Reset the flag t_on_failure("0"); } } #============================================= # Default redirect handler - PSTN gateway inter-op #============================================= failure_route[4] { get_redirects("*"); t_relay(); } ********************************************************************************************* Below is a snippet of my logs when the call fails. -----Call 1 Aug 28 19:53:27 [12103]: [INVITE] [from: asterisk] [calling: 15552121212] [caller: 99915553131313] Aug 28 19:53:27 [12103]: RequestUri:[sip:15552121212@openser] FromUri:[ sip:99915553131313@asterisk] forwarding to redirect server... Aug 28 19:53:35 [12103]: [ACK] [from: asterisk] [calling: 15552121212] [caller: 99915553131313] Aug 28 19:53:35 [12103]: ERROR:uac:replace_from: decline FROM replacing in sequential request in auto mode (has TO tag) Aug 28 19:53:35 [12103]: RequestUri:[sip:15552121212@openser] FromUri:[ sip:99915553131313@asterisk] forwarding to redirect server... Aug 28 19:53:41 [12107]: [BYE] [from: redirectServer] [calling: 99915553131313] [caller: 15552121212] -----Call 2 Aug 28 19:53:53 [12105]: [INVITE] [from: asterisk] [calling: 15552121212] [caller: 99915553131313] Aug 28 19:53:53 [12105]: RequestUri:[sip:15552121212@openser] FromUri:[ sip:99915553131313@asterisk] forwarding to redirect server... Aug 28 19:53:56 [12105]: ERROR:tm:w_t_relay: t_forward_nonack failed Thanks and many regards, Tolu

1 0

[OpenSER-Users] Broken "BYE" returned from Asterisk on TLS implementation ?
by David Loh 29 Aug '07

29 Aug '07

Hi All, Greeting. I've been struggle with OpenSER TLS implementation for more than a week, since I've ported from UDP to TLS, everything work fine except the "BYE" request from Asterisk (loose route), my implementation was something like below: [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] My OpenSER.cfg already configured to listen on two port which is :- "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or even voicemail) having no problem, but when the callee disconnect the call, caller will never get hang up :( I've attached my ethereal trace/ngrep to pastebin, http://pastebin.ca/673392 Wondering if anyone can help me with the broken "BYE" that returned from Asterisk ? Line #131, supposedly this line should have contain 2 Via header, one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", but somehow the TLS via header was gone !! (compare to previous ACK (Line #117) /INVITE (Line #51). Due to the missing TLS via header, OpenSER log file was complaining "protocol/port mis-match". The last BYE request (Line #256) is actually firing from Client, which contain the "TLS" via. I've even tried "force_send_socket" to port 5061 (instead of 5060) from loose route, but it complaining TLS certificate error, since Asterisk doesn't support TLS natively, I've no clue why is the ACK/INVITE/CANCEL work but not BYE. if (loose_route) { .... if(is_method("BYE")) { force_send_socket(IP:5061); } } Has any one gone through of this kinda OpenSER over TLS + Asterisk setup, I'm really appreciate if you can share your experience with me, or pin point what's the mistakes I made here. Thanks in advance. Regards, David Loh

2 2

[OpenSER-Users] Need help in configuring Open SER - XMPP module with Jabberd
by Aditya Chaturvedi 28 Aug '07

28 Aug '07

Need some help in configuring Open SER xmpp module (I tried configuring using FPSDEM 2007 presentation pdf) I have one single server on which Open SER and jabberd both are running. I tried configuring xmpp module on Open SER installation by changing the openser.cfg file. I added following things in openser.cfg file: loadmodule "xmpp.so" modparam("xmpp","domain _separator","*") modparam("xmpp","gateway_domain"," sipxmpp.catch22") modparam("xmpp","xmpp_domain","xmppsip.catch22") modparam("xmpp","xmpp_host","xmpp.catch22") modparam("xmpp","backend","component") modparam("xmpp","xmpp_password","0915fc59edc8bd9425ddff892d89c22b2f3f0030") if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; 'sipxmpp.catch22', 'xmppsip.catch22', 'xmpp.catch22' , all are alias names of same machine. So using this I am able to send Message from SIP client to XMPP Client but not in opposite direction. Also message is getting delivered some times with some difficulty and some time more than one time. SIP client registered user - user1@openser_domain XMPP client registered user - user2@xmpp_host I sent message from SIP client to user2*xmpp_host@openser_domain(xmpp_host = xmpp.catch22 in my case) and it got delivered but from opposite side XMPP client is trying to send message to user1*openser_domain@xmpp_domain(xmpp_domain = xmppsip.catch22 in my case) which is not getting delivered. So at one place xmpp_host is working and at other xmpp_domain is used which is not working. I am getting confused with the settings of domains i.e. gateway_domain and xmpp_domain in openser.cfg file, Please give clarity on that. probably i am doing mistake there.One more question , how gateway_domain is different from openser_domain?? is there some more documentation from where i can take reference? Thanks, Aditya

1 0

[Serusers] TCP sessions
by Stefan Ljung (TN/EAB) 28 Aug '07

28 Aug '07

Is it possible to make the SER 0.9 (or 2.0) to reuse the listening TCP port , usually 5060, when it relays requests to the next SIP element in the network.

3 2

[OpenSER-Users] OpenSER and SIPPY b2bua
by TCB 28 Aug '07

28 Aug '07

Anyone out there managed to use sippy b2bua with openser . Any pointers would be most welcome. -- TC

1 0

[OpenSER-Users] Group and directed call pickup
by Zahid Mehmood 28 Aug '07

28 Aug '07

Hi All, I am working on implementing group/directed call pickup and it is my understanding that this requires dialog state information. Can someone describe the best approach to implement this when using multiple proxies? We have multiple proxies that replicate registration information to each other. Is there a way to share the dialog information among multiple proxies? or should all calls be routed through one proxy? Also, does this require presence server? Thanks in advance. -- Zahid

1 0

[OpenSER-Users] sip_msg_cloner: cannot allocate memory
by laurent schweizer 28 Aug '07

28 Aug '07

Hello, I have 2 openser working on 2 diffrents server. The first is doing the full job, Registration, authentifications .... And the second is only used for SEMS so a very simple config file. the first is working perfectly but the second create me this night a memory error. He was working correcly more that 1 month and I don't think that the problem is a overload usage because during the night I have low traffic. ./openser[5429]: ERROR: t_newtran: new_t failed ./openser[5435]: ERROR: sip_msg_cloner: cannot allocate memory ./openser[5435]: ERROR: new_t: out of mem: mpath="/usr/local/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "xlog.so" loadmodule "avpops.so" modparam("tm", "fr_inv_timer", 15000) modparam("tm", "pass_provisional_replies", 1) modparam("rr", "enable_full_lr", 1) children=60 disable_tcp=yes disable_dns_blacklist=yes disable_dns_failover=yes disable_dns_blacklist=true # modparam( "tm", "tw_append", "info_append:hdr[Content-Length];hdr[Content-Type];msg[body]") #modparam( "tm", "tw_append", "info_append:Content-Length=$avp(i:70);Content-Type=$avp(i:71);body=$avp(i:72)") modparam("tm", "tw_append", "billing:billingid=$avp(i:67);accountid=$avp(i:68);Remote-Party-ID=$avp(i:69);Pays=$avp(i:70)") avp_aliases="bid=i:67;acid=i:68;rp=i:69;pays=i:70" # ------------------------- request routing logic ------------------- route{ # xlog("L_ERR","BEGIN: call [$rm] ru[$ru] fu[$fu] tu[$tu] si[$si] ct[$ct]\n"); if (!mf_process_maxfwd_header("5")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return; }; sl_send_reply("100","---- trining"); record_route(); # if (loose_route()) { # xlog("L_ERR","LOOSE_ROUTE: [$rm] ru[$ru] fu[$fu] tu[$tu] si[$si] ct[$ct]\n"); # } if (!t_newtran()){ sl_send_reply("500","could not create transaction"); return; }; if (method=="INVITE"){ $avp(i:67) = $hdr(x-billingId); $avp(i:68) = $hdr(x-accountcode); $avp(i:69) = $hdr(Remote-Party-ID); $avp(i:70) = $hdr(Pays); if (uri =~ "sip:.*@") { if (!t_write_unix("/tmp/sems_sock","billing/billing")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } # if (method=="INFO") { # $avp(i:70) = $cT; # $avp(i:71) = $cl; # $avp(i:72) = $rb; # # if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ # log("could not contact sems\n"); # t_reply("500","could not contact media server"); # } # } } if (method=="BYE" || method=="CANCEL"){ if(!t_write_unix("/tmp/sems_sock","bye")) t_reply("500","error contacting bye"); } return; }

3 3

[OpenSER-Users] Bug in CPL module
by Helmut Kuper 28 Aug '07

28 Aug '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, currently I update openser 1.2.0 to 1.2.2. I found that cpl_run.c has an error arounf line 553: case REASON_ATTR: <<<<<<< .mine reason.len=n; ======= reason.len = n; >>>>>>> .r2629 should be: case REASON_ATTR: reason.len = n; regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGzYHh4tZeNddg3dwRAlCFAKCqgvyB2laQR2zusGt2ux02dAzNRQCdGBST DCAzgAopkYtVxhKS5rqr/DM= =O9Ck -----END PGP SIGNATURE-----

2 1

[Serusers] SIP-based multipoint video conferecing?
by SIP 28 Aug '07

28 Aug '07

Is anyone aware of, or does anyone have any experience with any SIP-based multipoint (i.e. more than 2 users) video conferencing server software? Ideally, of course, I'd like an H263 codec module for SEMS, but barring that, another SIP server that I can easily mesh with SER, forwarding video requests as need be to the video conferencing server to handle just the video mixing. Anyone use anything like that in their environments? N.

3 2

[OpenSER-Users] AVP Iteration
by Brandon Armstead 28 Aug '07

28 Aug '07

Hello, I'm trying to setup some kind of iteration through an AVP list using recursive routing, and use an AVP to create an "index" incrementing counter, i.e. if(is_avp_set("$avp(counter)")){ $avp(counter) = $avp(counter) + 1; } else { $avp(counter) = 0; } then I wish to apply to an avp index likewise: if($avp(some_avp[$avp(counter)]) == null){ # this avp list index does not exist, stop recursive routing } else { # keep doing operations } However this is trivial, there seems not to be any easy way of going about this, also openser.cfg errors out when I try to nest AVP's and use an AVP value as an index value. xl_parse_index: error parsing format [$avp(counter)])] expecting ']' parse error (410,59-89): unknown script variable Any thoughts? All Appreciated, thank you! -- Brandon Armstead

2 1

Re: [OpenSER-Users] About RTP stream
by Dan-Cristian Bogos 28 Aug '07

28 Aug '07

Yin, here you have a proper configuration: http://openser.svn.sourceforge.net/viewvc/openser/branches/1.2/examples/nat… Just make sure you install rtpproxy and start it on the same host with your openser. If properly installed, openser should recognize it in the logs (found RTP proxy, support for it activated) Cheers, DanB On 8/28/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote: > Hi Dan, > > In fact, I have two telephones VOIP and Openser as a SIP Proxy (in the same local network,there's no NAT.), they can register and call each other with no probleme. Now I just need the stream RTP pass by Openser, but no directely between two telephone. I have installed the rtpproxy, but I don't understand how it works exactely. And I don't know if it is nessary to install mediaproxy? This is my openser.cfg, can you give me some advice? > Thank you very much! > > Yin > > > -----Message d'origine----- > De : Dan-Cristian Bogos [mailto:dan.bogos@gmail.com] > Envoyé : mardi 28 août 2007 10:29 > À : zze-WU Yin RD-MAPS-LAN > Cc : users openser.org > Objet : Re: [OpenSER-Users] About RTP stream > > Yin, > > It all depends on what do u want to use it for. Can u explain a bit your scenario? > > DanB > > On 8/28/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote: > > > > Thank you for your help. I don't need NAT, it's just a local network. What I have to use? The Rtpproxy or Mediaproxy? And what diffrence between them? > > > > Yin > > > > -----Message d'origine----- > > De : Dan-Cristian Bogos [mailto:dan.bogos@gmail.com] Envoyé : lundi 27 > > août 2007 18:01 À : zze-WU Yin RD-MAPS-LAN Cc : users(a)openser.org > > Objet : Re: [OpenSER-Users] About RTP stream > > > > Hello Yin, > > > > openser by itself has no RTP capability. If you want NAT bypass solutions, check out the following softwares: > > 1. RTPProxy - http://ftp.iptel.org/pub/rtpproxy/ > > 2. Mediaproxy: - http://ag-projects.com/MediaProxy.html > > > > If you are looking for some RTP related applications, you can check SEMS: > > http://www.iptel.org/sems > > > > Hope this helps. > > > > DanB > > > > On 8/27/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote: > > > > > > > > > > > > Hello everyone, > > > > > > Now my openser is just a Sip proxy which manage the sip > > > requests and I want to manage RTP streams by the openser, that means > > > openser recieve RTP streams and retransmit them. How can I do? > > > > > > Thank you very much! > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > >

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Re: [OpenSER-Users] About RTP stream
by Dan-Cristian Bogos 28 Aug '07

28 Aug '07

Yin, It all depends on what do u want to use it for. Can u explain a bit your scenario? DanB On 8/28/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote: > > Thank you for your help. I don't need NAT, it's just a local network. What I have to use? The Rtpproxy or Mediaproxy? And what diffrence between them? > > Yin > > -----Message d'origine----- > De : Dan-Cristian Bogos [mailto:dan.bogos@gmail.com] > Envoyé : lundi 27 août 2007 18:01 > À : zze-WU Yin RD-MAPS-LAN > Cc : users(a)openser.org > Objet : Re: [OpenSER-Users] About RTP stream > > Hello Yin, > > openser by itself has no RTP capability. If you want NAT bypass solutions, check out the following softwares: > 1. RTPProxy - http://ftp.iptel.org/pub/rtpproxy/ > 2. Mediaproxy: - http://ag-projects.com/MediaProxy.html > > If you are looking for some RTP related applications, you can check SEMS: > http://www.iptel.org/sems > > Hope this helps. > > DanB > > On 8/27/07, zze-WU Yin RD-MAPS-LAN <yin.wu(a)orange-ftgroup.com> wrote: > > > > > > > > Hello everyone, > > > > Now my openser is just a Sip proxy which manage the sip > > requests and I want to manage RTP streams by the openser, that means > > openser recieve RTP streams and retransmit them. How can I do? > > > > Thank you very much! > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > >

1 0

[Serusers] ser + radius/mysql
by Nhadie 28 Aug '07

28 Aug '07

Hi, i'm testing ser with radius using mysql. at first without inserting the Sip-Group on radgroupcheck, my x-lite can register. but when i add it i always get unauthorized. is there something wrong on the values i inserted? on my ser.cfg REGISTER part i only have this if (method=="REGISTER") { if (!radius_www_authorize("mydomain.com")) { if (isflagset(8)) {xlog("L_INFO", "Authenticating\n");}; www_challenge("mydomain.com", "0"); break; }; if (!save("location")) { sl_reply_error(); }; break; }; mysql> select * from radcheck; +----+------------------------+-----------+----+---------+ | id | UserName | Attribute | op | Value | +----+------------------------+-----------+----+---------+ | 1 | 1001(a)mydomain.com | Password | == | TEST1001 | +----+------------------------+-----------+----+---------+ 1 row in set (0.00 sec) mysql> select * from radgroupcheck; +----+-----------+-----------+----+----------------------+ | id | GroupName | Attribute | op | Value | +----+-----------+-----------+----+----------------------+ | 1 | intl-pstn | Sip-Group | == | world-pstn | +----+-----------+-----------+----+----------------------+ 2 rows in set (0.00 sec) mysql> select * from usergroup; +----+------------------------+-----------+ | id | UserName | GroupName | +----+------------------------+-----------+ | 1 | 1001(a)mydomain.com | intl-pstn | +----+------------------------+-----------+ 1 row in set (0.00 sec)

2 1

[OpenSER-Users] OpenSER as Asterisk/CallWeaver frontend
by Eugene Prokopiev 28 Aug '07

28 Aug '07

Hi, Can anybody tell some success story about using OpenSER as Asterisk/CallWeaver frontend? I tried to googling first but where are no detailed useful stories :( Now I have CallWeaver (Asterisk fork) installed on Linux server which connected to 2 different providers via one wan interface but 2 ip addresses from one network, for example: eth0 - 10.0.0.2/24 -> provider1(10.0.2.1) - 10.0.0.3/24 -> provider1(10.0.3.1) So, I need send SIP requests to provider1 from address 10.0.0.2/24 and to provider2 from 10.0.0.3/24. This must be done because my internet provider must account trafic on router 10.0.0.1 for 2 VoIP providers separately and it's billing system can do it only by different source ip addresses and can't do it by destination ip addresses. I can separate SIP requests between 2 ip addresses with linux kernel 2.6.18 or higher by using ip_nat_sip iptables module and MARK/SNAT. But this approach can't be used in OpenVZ virtial environment in which CallWeaver must be used. Is it possible to send SIP requests from CallWeaver via OpenSER and modify them with OpenSER to replace source addresses in UDP headers and SIP headers? How can I say CallWeaver to send SIP requests via OpenSER? How can I implement this routing logic in OpenSER? Can anybody give me a simple example? -- Thanks, Eugene Prokopiev

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[OpenSER-Users] Unauthorized Calls - [Openser - X-lite]
by Jeferson Prevedello 27 Aug '07

27 Aug '07

Hello, I implemented an environment using to openser + mysql. The enviroment functions perfectly, however I perceived that users (branches) not registered in mysql are generating called. I installed the X-lite softphone in my computer trying to reproduce the situation. In the properties of configuration of the X-lite, "field Password" I type "trash" as password (wrong password). The display of X-lite showed the following message: "Registration error: 401 - Unauthorized". In the contacts drawer I add a contact (double click on the new contact), and the call was generate without restriction (very bad). Some idea of as I solve this problem? Thanks Regards Jeferson

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Fw: [OpenSER-Users] Unauthorized Calls - [Openser - X-lite]
by Simone Torri 27 Aug '07

27 Aug '07

Hi Guys, Here in my company I have the same problem. The solution that I adopted was apply the proxy_authorize function and restrict all incoming calls. I use openser only for originate calls. Some tip ? Cheers, Torri ----- Original Message ----- From: "Jeferson Prevedello" <jprevedello(a)terra.com.br> To: "Dan-Cristian Bogos" <dan.bogos(a)gmail.com> Cc: <users(a)openser.org> Sent: Monday, August 27, 2007 12:26 PM Subject: Re: [OpenSER-Users] Unauthorized Calls - [Openser - X-lite] Hello DanB, More a problem ! :-( I apply the following configuration in my openser.cfg: if (method=="INVITE") { if (!proxy_authorize("", "subscriber")) { proxy_challenge("","0"); exit; } }; I perceived that with the configuration above 'only' registered users can generate called, however I not receive more called originated through of PSTN or of any branch of PBX. I believe these calls are deny because the source (PSTN - Branches) not are registering in the openser server. Is possible to apply the configuration above only for calls 'originated' from openser ? Thanks ! Regards Jeferson ----- Original Message ----- From: "Dan-Cristian Bogos" <dan.bogos(a)gmail.com> To: "Jeferson Prevedello" <jprevedello(a)terra.com.br> Cc: <users(a)openser.org> Sent: Monday, August 27, 2007 8:35 AM Subject: Re: [OpenSER-Users] Unauthorized Calls - [Openser - X-lite] Hello Jeferson, Your configuration looks a bit messy, if I were OpenSER I would also refuse it. :). I would suggest taking a more standard configuration (u can find many examples on this location: http://openser.svn.sourceforge.net/viewvc/openser/branches/1.2/examples/) and use 1.2 branch of software for start, and experiment with it into some lab environment. It is a bit difficult as a beginner to start directly experimenting on a production configuration, perhaps written by somebody else without understanding it. You will end up having big issues when troubleshooting in production environment. The tip I gave you would be really easy to implement it with a block of few lines, eg: if (is_method("INVITE")){ if (!proxy_authorize("", "subscriber)) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; }; Documentation for you to understand those lines here: http://www.openser.org/docs/modules/1.2.x/auth_db.html#AEN192 Usually, there is a loot of documentation and howtos in openser wiki, so I would suggest you having a glance on some titles which look close to your needs as a beginner. http://www.openser.org/dokuwiki/doku.php Cheers, DanB On 8/27/07, Jeferson Prevedello <jprevedello(a)terra.com.br> wrote: > Hello DanB, > > Thanks! > > As DanB´s suggestion, I tried to implement a mechanism that only allowed > authenticated members make calls, but my configuration didn´t function. > > This is my first project with openser, therefore I do not have much > experience. If someone know how to help me to implement this verification, > I > will be very thankful. > > Below, my openser.cfg file: > > -x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x -x-x-x-x-x-x-x-x > > > # ----------- global configuration parameters ------------------------ > > debug=3 > fork=yes > log_stderror=no > log_facility=LOG_LOCAL7 > > # hostname matching an alias will satisfy the condition uri==myself". > alias=xxx.xxx.xxx.xxx > listen=udp:xxx.xxx.xxx.xxx:5060 > > # check_via - Turn on or off Via host checking when forwarding replies. > # Default is no. arcane. looks for discrepancy between name and > # ip address when forwarding replies. > check_via=yes > > # syn_branch - Shall the server use stateful synonym branches? It is > # faster but not reboot-safe. Default is yes. > syn_branch=yes > > # dns - Uses dns to check if it is necessary to add a "received=" field > # to a via. Default is no. > # rev_dns - Same as dns but use reverse DNS. > dns=no > rev_dns=no > port=5060 > children=4 > > # memlog - Debugging level for final memory statistics report. Default > # is L_DBG -- memory statistics are dumped only if debug is set high. > memlog=3 > > # sip_warning - Should replies include extensive warnings? By default > # yes, it is good for trouble-shooting. > sip_warning=yes > > # fifo - FIFO special file pathname > fifo="/tmp/openser_fifo" > > # reply_to_via - A hint to reply modules whether they should send reply > # to IP advertised in Via. Turned off by default, which means that > # replies are sent to IP address from which requests came. > reply_to_via=no > > # mhomed -- enable calculation of outbound interface; useful on > # multihomed servers. > mhomed=0 > > # ------------------ module loading ---------------------------------- > > # Uncomment this if you want to use SQL database > loadmodule "/usr/lib/openser/modules/mysql.so" > loadmodule "/usr/lib/openser/modules/sl.so" > loadmodule "/usr/lib/openser/modules/tm.so" > loadmodule "/usr/lib/openser/modules/rr.so" > loadmodule "/usr/lib/openser/modules/maxfwd.so" > loadmodule "/usr/lib/openser/modules/usrloc.so" > loadmodule "/usr/lib/openser/modules/registrar.so" > loadmodule "/usr/lib/openser/modules/textops.so" > loadmodule "/usr/lib/openser/modules/nathelper.so" > loadmodule "/usr/lib/openser/modules/acc.so" > loadmodule "/usr/lib/openser/modules/xlog.so" > > # Uncomment this if you want digest authentication > # mysql.so must be loaded ! > loadmodule "/usr/lib/openser/modules/auth.so" > loadmodule "/usr/lib/openser/modules/auth_db.so" > > # ----------------- setting module-specific parameters --------------- > > # ------------- usrloc parameters > > # 2 enables write-back to persistent mysql storage for speed > # disable=0, write-through=1 > modparam("usrloc", "db_mode", 0) > > # minimize write back window - default is 60 seconds > modparam("usrloc", "timer_interval", 30) > > # ------------- auth parameters > > # 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 parameters > > # add value to ;lr param to make some broken UAs happy > modparam("rr", "enable_full_lr", 1) > > # ------------- !! Nathelper > > modparam("registrar", "nat_flag", 6) > modparam("nathelper", "natping_interval", 30) # Ping interval 30 s > modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind > NAT > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # > Nathelper with RTPproxy > > # ------------- tm parameters > > modparam("tm", "fr_timer", 12) > modparam("tm", "fr_inv_timer", 24) > > # ------------- acc parameters > > modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") > modparam("acc", "db_flag", 2) > modparam("acc", "db_missed_flag", 2) > modparam("acc", "log_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "log_level", 2) # Set log_level to 2 > > # Allow no more than 1 contacts per AOR > modparam("registrar", "max_contacts", 3) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > 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; > }; > > # < Acconting > > if (method=="INVITE") > { > log(1, "Generate call - START\n"); > setflag(1); /* set for accounting (the same value as in > log_flag!) */ > setflag(2); > }; > > if (method=="BYE") > { > log (1, "Hung-up \n"); > setflag(1); > }; > > if (method=="CANCEL") > { > log (1, "Lost call \n"); > setflag(1); > } > > if (!method=="REGISTER") > record_route(); > > if (nat_uac_test("3")) > { > # Allow RR-ed requests, as these may indicate that > # a NAT-enabled proxy takes care of it; unless it is > # a REGISTER > > if (method == "REGISTER" || ! search("^Record-Route:")) > { > log(1,"LOG: Someone trying to register from private > IP, > rewriting\n"); > > # This will work only for user agents that support > symmetric > # communication. We tested quite many of them and > majority is > # smart enough to be symmetric. In some phones it > takes > a configuration > # option. With Cisco 7960, it is called > NAT_Enable=Yes, > with kphone it is > # called "symmetric media" and "symmetric signalling". > > fix_nated_contact(); # Rewrite contact with source IP > of > signalling > force_rport(); # Add rport parameter to topmost > Via > setflag(6); # Mark as NATed > }; > }; > # 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"); > 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("xxx.xxx.xxx.xxx", "subscriber")) > { > www_challenge("xxx.xxx.xxx.xxx", "0"); > return; > }; > save("location"); > return; > }; > > lookup("aliases"); > if (!uri==myself) > { > append_hf("P-hint: outbound alias\r\n"); > route(1); > return; > }; > > # Router Cisco if not sip branche > log(1,"LOG: testando se destino-sip e' 418x ...\n"); > > if ( ! ( uri =~ "^sip:418[1-9].*" ) && > ! ( uri =~ "^sip:4397")) > { > log(1,"LOG: destino-sip not is 418x .\n"); > route(2); > > log(1,"LOG: rewriting hostport yyy.yyy.yyy.yyy:5060...\n"); > rewritehostport("yyy.yyy.yyy.yyy:5060"); > log(1,"LOG: t_relay...\n"); > t_relay(); > > log(1,"LOG: break...\n"); > return; > } > log(1,"LOG: destino-sip 418x, continue .\n"); > > # native SIP destinations are handled using our USRLOC DB > if (!lookup("location")) > { > sl_send_reply("404", "Not Found"); > return; > }; > }; > append_hf("P-hint: usrloc applied\r\n"); > route(1); > } > > ####################################### > > route[1] > { > # !! Nathelper > if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && > !search("^Route:")) > { > sl_send_reply("479", "We don't forward to private IP > addresses"); > return; > }; > > # if client or server know to be behind a NAT, enable relay > if (isflagset(6)) > { > force_rtp_proxy(); > t_on_reply("1"); > append_hf("P-Behind-NAT: Yes\r\n"); > }; > > if (!t_relay()) > { > sl_reply_error(); > return; > }; > } > # !! Nathelper > onreply_route[1] > { > # NATed transaction ? > if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") > { > fix_nated_contact(); > force_rtp_proxy(); > } > else if (nat_uac_test("1")) > { > fix_nated_contact(); > }; > } > > ####################################### > > route[2] { > > ### Dial Plan for gateway VoIP ### > > # Sao Paulo 11 > if ( uri =~ "^sip:9911.*" ) > { > log(1,"LOG: destination is 9911x, change prefix..."); > strip(4); > prefix("011"); > return; > } > > # Error (Number inexistent) > sl_reply_error(); > > } > > -x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x -x-x-x-x-x-x-x-x > > Regards > Jeferson > > > > > > ----- Original Message ----- > From: "Dan-Cristian Bogos" <dan.bogos(a)gmail.com> > To: "Jeferson Prevedello" <jprevedello(a)terra.com.br> > Cc: <users(a)openser.org> > Sent: Saturday, August 25, 2007 3:06 PM > Subject: Re: [OpenSER-Users] Unauthorized Calls - [Openser - X-lite] > > > > Hello Jeferson, > > > > it all depends on your openser.cfg. > > If you put in there that all the INVITE-s should be authenticated, your > > users will not be able anymore to call without having a valid user and > > password for your server. Note that by default openser will not do any > > check for you, in order to keep the flexibility of be used in > > different environment setups. > > > > Cheers, > > DanB > > > > On 8/25/07, Jeferson Prevedello <jprevedello(a)terra.com.br> wrote: > >> > >> > >> Hello, > >> > >> I implemented an environment using to openser + mysql. The enviroment > >> functions perfectly, however I perceived that users (branches) not > >> registered in mysql are generating called. > >> > >> I installed the X-lite softphone in my computer trying to reproduce the > >> situation. > > >> In the properties of configuration of the X-lite, "field Password" I > >> type > >> "trash" as password (wrong password). > >> > >> The display of X-lite showed the following message: "Registration > >> error: > >> 401 > >> - Unauthorized". > >> > >> In the contacts drawer I add a contact (double click on the new > >> contact), > >> and the call was generate without restriction (very bad). > >> > >> Some idea of as I solve this problem? > >> > >> Thanks > >> > >> Regards > >> Jeferson > >> > >> _______________________________________________ > >> 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 Flickr agora em português. Você clica, todo mundo vê. http://www.flickr.com.br/

1 0

[OpenSER-Users] About RTP stream
by zze-WU Yin RD-MAPS-LAN 27 Aug '07

27 Aug '07

Hello everyone, Now my openser is just a Sip proxy which manage the sip requests and I want to manage RTP streams by the openser, that means openser recieve RTP streams and retransmit them. How can I do? Thank you very much!

2 1

[Users] understanding SEMS
by Victor Gamov 27 Aug '07

27 Aug '07

hi guys! can somebody explain me how SEMS works with SER/OpenSER. As I conceive proxy sends requests to SEMS via unix socket. Then SEMS needs constructs reply or new request and ask proxy to send it. But where reply come to proxy? into onreply_route[0] block? is it possible to use t_on_reply() function when initial request writing to SEMS to ask proxy to use specific onreply_route[] block? Thanks for advise! -- CU, Victor Gamov

4 5

[OpenSER-Users] Upload to XCAP Server - 405 Method Not Allowed error.
by Kriti Wadhwa 27 Aug '07

27 Aug '07

Hi, I am testing Presence service using Openser, XCAP server... After firing up the UCT IMS Client in the final step, everything goes well except the "Upload to XCAP Server" part... On clicking that I get: <!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML 2.0//EN"> <html><head> <title>405 Method Not Allowed</title> </head><body> <h1>Method Not Allowed</h1> <p>The requested method PUT is not allowed for the URL /xcap-root/pres-rules/user/alice/presence_rules.xml.</p> <hr> <address>Apache/2.0.54 (Fedora) Server at xcap Port 80</address> </body></html> Tried changing permissions to the presence_rules.xml file and also made all necessary changes in httpd.conf file but in vain... Hope someone can help me in this. Thanks, - Kriti Mailo: kriti.wadhwa(a)tcs.com =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

1 0

[OpenSER-Users] Openser + Asterisk
by CLEMENT Frédéric 27 Aug '07

27 Aug '07

3 4

[Serusers] acc_db_request milliseconds
by Sho Aihara 27 Aug '07

27 Aug '07

Hello. Please teach me about acc_db_request for the call charge. For example, when call starts (ACK), my ser.cfg acc_db_request will executed to insert a new row for db. timestamp ------------------- 2007-08-24 23:50:00 And when call ends (BYE), acc_db_request executed. timestamp ------------------- 2007-08-24 23:50:03 These values were considered about milliseconds? For example... timestamp ------------------- 2007-08-24 23:50:00(.000) timestamp ------------------- 2007-08-24 23:50:03(.525) In this case, I think that the interval of these time becomes 4 seconds If I use these values for the call charge (round-up millisecond). Or, has the time recorded with BYE already been rounded up? Thanks, Sahria

1 0

[OpenSER-Users] en.wikipedia
by kgfleischmann＠t-online.de 27 Aug '07

27 Aug '07

The Wikipedia article of OpenSER is subject of a deletion request. The reasons can be found in this text on the discussion page of "Miconda" "Notability of OpenSER A tag has been placed on OpenSER, requesting that it be speedily deleted from Wikipedia. This has been done because the article appears to be about a person, group of people, band, club, company, or web content, but it does not indicate how or why the subject is notable: that is, why an article about that subject should be included in an encyclopedia. Under the criteria for speedy deletion, articles that do not assert the subject's importance or significance may be deleted at any time. Please see the guidelines for what is generally accepted as notable. If you think that you can assert the notability of the subject, you may contest the deletion by adding {{hangon}} to the top of the page (just below the existing speedy deletion or "db" tag), coupled with adding a note on the article's talk page explaining your position, but be aware that once tagged for speedy deletion, if the article meets the criterion it may be deleted without delay. Please do not remove the speedy deletion tag yourself, but don't hesitate to add information to the article that would confirm the subject's notability under Wikipedia guidelines. For guidelines on specific types of articles, you may want to check out our criteria for biographies, for web sites, for bands, or for companies. Feel free to leave a note on my talk page if you have any questions about this. Calltech 11:29, 25 July 2007 (UTC) " If someone could take care of the article and possibly join the deletion discussion (http://en.wikipedia.org/wiki/Wikipedia:Articles_for_deletion/OpenSER) ? /Klaus

3 2

Re: [Serusers] why combine ser with asterisk
by SIP 26 Aug '07

26 Aug '07

Michael Grigoni wrote: > SIP wrote: > > <snip> > >> Take Cray Supercomputers as a prime example. Seymour Cray built this >> fantastic supercomputer architecture, but didn't build any >> peripherals for it or even an operating system, assuming that, if >> people had the raw speed and power available, they'd be eager to use >> it even if they had to build their own hardware/software. > > Yes, Seymour liked to code in machine language right from the console > and the O/Ses for his machines developed 'organically' from the bottom up > in general, but wouldn't you consider SCOPE, KRONOS, NOS etc. operating > systems? And as for hardware, the CDC peripherals and Hyperchannel > offerings seemed to be the best in the industry on Cray CPUs... > > Regards, > > Michael Right, but Cray himself was TOTALLY against their creation. This was one of the reasons he left CDC to form Cray Research (which, mind you, CDC was very much in support of, as they had much faith in Cray's abilities). He argued vehemently that the peripherals were unnecessary products, and eventually, this disagreement caused him to leave and form Cray Research, where he could be free from 'management failures' as he liked to call them. The CDC, however, was extremely supportive of Cray Research, and put money and development time into it, much to Cray's chagrin. Then, again, when Cray Research was faced with the decision of whether to build computers that people would buy or computers for the sake of building computers, they understood that they needed to stay in business, and decided on the path that would lead them to solvency. This caused Cray to leave his own company and form Cray Computers. He vowed that CCC would make fast and powerful computers and that nothing else would be a priority. He assumed that speed and power were the ultimate goal and that everything would fall into place if he achieved that goal. Of course, his shortsightedness caused bankruptcy. Incidentally, having had to use NOS on a regular basis back in the day, I would scarcely call it an operating system. Anything that lets you manually change the bits in memory of a running program isn't an OS, it's just a GUI for the hardware. ;) N.

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[Serusers] xlog module prefix text feature
by Tomasz Zieleniewski 26 Aug '07

26 Aug '07

Hi, I am using xlog module for my ser.cfg debuging. I though that it would be very helpful to have a feature which would prefix each log with a module-parameter defined text. It is useful for instance when there are more then one instances of ser on the same host. Cheers Waiting for your feedback Tomasz

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[Serusers] Host three way conference
by inge 26 Aug '07

26 Aug '07

Hi all, I've some questions about it. First of all, when I use a SPA-941 with SER, I can do a three way conference with two incoming calls. Why I can't if I use an other product like SPA2100 ? I can deduce that this service depends on the functionalities of the product, but I'm not sure. But if I want nevertheless to use this service independently of the product used, how can I do ? Using SEMS ? In this case, can we install it without reinstallating SER ? Moreover, with SEMS I think I will have to determine a trigger ? (ie. something like a prefix in the URI ?) Thank for your answers and your support. Regards, Adrien .L

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[OpenSER-Users] rpm packaging
by Daniel-Constantin Mierla 24 Aug '07

24 Aug '07

Hello, I would like to welcome two new members: Jan Ondrej and Peter Nixon. So far, they did a lot of work to package openser for Fedora and Suse distributions, becoming now the maintainers of the spec files for those distributions in OpenSER repository. Many thanks for the commitment and effort that will lead in installing and maintaining easier OpenSER in these distributions. Cheers, Daniel

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[OpenSER-Users] call forking
by bernard.le＠orange-ftgroup.com 24 Aug '07

24 Aug '07

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[OpenSER-Users] How to avoid the Apple tax?
by Richard Bennett 24 Aug '07

24 Aug '07

Hi, I've been looking for an ultraportable laptop to run Ubuntu Linux on, as a more portable alternative to my current Dell that breaks my back each time I take it with me. Seeings as I don't need Windows I was hoping to avoid the Microsoft tax by buying a barebones laptop that doesn't come preloaded with Windows, i found something good here: http://www.rkcomputer.net/store/index.php/action/item/id/181/ (System76 have the same one, but they don't ship to Europe) But by the time I loaded the options I need, and shipped it to Belgium, paying the VAT and postage it all adds up and takes weeks. It turns out I can get a Macbook 13" delivered free in 3 days with a US keyboard (requirement), and it looks a lot nicer too. But of course, even though I just want to run Ubuntu on it, it comes preloaded with OSX. Has anyone heard of a way of avoiding the Mac tax, like the way people request a Microsoft rebate by not accepting the Ms License? I don't want to migrate to OSX because I run all my servers on Linux and by using it daily I learn a lot about the OS, and I have become addicted to the Opens source way of helping and getting helped. I'm not really worried about the financial cost of a bundeled Mac OSX, but am looking for a way to show Apple that the fact that they bundle OSX on all their hardware is no better than the way MS get Windows preloaded on all consumer PCs. I'll start by sending this question to their customer service and see what they say... Richard

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[Serusers] 100 Trying response generation question
by tzieleniewski 24 Aug '07

24 Aug '07

Hi, I have to SERs working serialy. When first one receives INVITE it doesn't generate the 100 Trying response. First one passed the INVITE message to the second proxy which generates the 100 response. How is the 100 Response generation controlled in SER, is it configurable?? Cheers Thanks in advance Tomasz

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[Serusers] Authentication configuration problem
by Sean 24 Aug '07

24 Aug '07

Greetings, I'm trying to configure authentication for the UA, I uncommented some sentences in the original ser.cfg for authentication as the comment in the ser.cfg guided. but I found that the following sentence will cause the ser crash: modparam("usrloc", "db_mode", 2) of course I've marked the previous sentence which will set the parmaeter to 0. can anyone help me on this problem? thanks a lot. Sean ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

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