March 2012 - sr-users - kamailio.org

Re: [SR-Users] how to configure Kamailio in ECHO mode
by Fomitchev, AndreiX 29 Mar '12

29 Mar '12

Oups, sorry, bad manipulation: I want to configure Kamailio in echo mode, i.e. SIP server repeats the input audio into the output, so I can hear myself with a very small amount of delay... Is it possible? If so, how? Thank you for your help. Best regards, From: Fomitchev, AndreiX Sent: Thursday, March 29, 2012 1:34 PM To: 'sr-users(a)lists.sip-router.org' Subject: how to configure Kamailio in ECHO mode Hi, For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. SIP server --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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how to configure Kamailio in ECHO mode
by Fomitchev, AndreiX 29 Mar '12

29 Mar '12

Hi, For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. SIP server --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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branches usage
by Uri Shacked 29 Mar '12

29 Mar '12

Hi, Thanks, this was very helpful for understanding. Still let’s see if i got it right: I get the INVITE. Do whatever I do on the main route. Then do: “append_branch();” “changes….changes…..;” Now before the t_relay do t_on_branch(name of branch). Get the reply (301 in my case) And from here I am back to the original request? maybe a stupid question now but - I must use the append_branch() right :-)? BR, Uri Hi, On 03/27/2012 04:46 PM, Uri Shacked wrote: >* In my case i need to change the header and then send it. *>* *>* Here the case works if i make the changes after i sent the invite and *>* got the reply. As i know the t_on_branch will work after the invite is *>* sent... am i wrong? *>* *>* There are many questions to ask about the branches... when do they work *>* in a serial manner, when parallel... where can i find it documented best? * Actually it's quite simple. If a request comes in and you modify it in a normal route (e.g. calling rtpproxy_offer(), uac_replace_from(), adding additional headers and stuff like that) and you send it out using e.g. t_relay(), then in a serial scenario when you re-enter via a failure-route, these changes will still be there (which is not always what you want). On the other hand, if you register a branch route via t_on_branch() and do your changes there instead of prior to calling t_relay(), then these changes from branch route will be rolled back, and you can do your changes easily again in the same or another branch route starting with your original message. This is important for example in serial call forward scenarios, where the first destination might require an rtpproxy, but the second doesn't, or when the first destination requires different custom headers compare to the second. Basically the same applies to parallel scenarios. If you modify your message before calling lookup() and you've multiple contacts registered for your destination, then your changes will apply for all branches which are created by that. However if you move your modifications to a branch route, then you can work on your messages individually, depending on the branch. Hope this clarifies it a bit. Andreas

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AVPs in timer-route
by Klaus Darilion 29 Mar '12

29 Mar '12

Hi! I wonder what happens to AVPs defined in a route which is executed by rtimer? It seems (tests on Kamailio 1.5) that the AVPs, once they are defined, they life forever. This would be fine for my application. Are there any issues when using AVPs in timer routes? Thanks Klaus

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ITSPA 2012 Award for Open Source VoIP Projects
by Daniel-Constantin Mierla 28 Mar '12

28 Mar '12

Hello, ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry: * http://www.itspaawards.org.uk/ Open Source VoIP Projects won a special category this year, Members' Pick, for providing a real value to VoIP Industry. I had the chance to attend the event in London and I have been selected to pick up the award. I made a news on kamailio.org website with more details: * http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projec… As you would expect, a complete voip platform usually involves several open source projects, for components such as load balancers, registrar, proxy or media servers, thus the decision of ITSPA for awarding to the group. I am reiterating the thanks to everyone in the community, developers or users, for their dedication in improving the project. This time extended credits to our British community, very active in both development and usage, for their support in this awards ceremony and overall advertising on the UK markets. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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TCP memory failures...
by Krishna Kurapati 28 Mar '12

28 Mar '12

I keep getting these... I am using master branch... Mar 26 06:46:34 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:1063]: ERROR: tcpconn_new: mem. allocation failure Mar 26 06:46:34 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:4135]: ERROR: handle_new_connect: tcpconn_new failed, closing socket Mar 26 06:46:48 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:1063]: ERROR: tcpconn_new: mem. allocation failure Mar 26 06:46:48 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:4135]: ERROR: handle_new_connect: tcpconn_new failed, closing socket Mar 26 06:47:51 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:1063]: ERROR: tcpconn_new: mem. allocation failure Mar 26 06:47:51 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:4135]: ERROR: handle_new_connect: tcpconn_new failed, closing socket Mar 26 06:48:45 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:1063]: ERROR: tcpconn_new: mem. allocation failure Mar 26 06:48:45 siptest /usr/sbin/kamailio[5062]: ERROR: <core> [tcp_main.c:4135]: ERROR: handle_new_connect: tcpconn_new failed, closing socket I have never seen this before... what could cause this? Is there something I could do to fix this issue? Krish Kura

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uac with more then one target?
by Karsten Horsmann 28 Mar '12

28 Mar '12

Hello, my kamailio proxy handles calls between public sip-clients and internal ivr systems. Some customer give us calls via sip-trunks to our proxy and this goes to the ivr too. Now some ask if we can act as UAC to get calls from them, for example authenticate to sipgate and feed the landline calls to our ivr system via kamailio. AFAIK the module UAC provides only one pair of user/password credentials. Is that right? Have i here a chance to do the job with kamailio or must i use an second voip system for that? -- Mit freundlichen Grüßen *Karsten Horsmann*

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Re: [SR-Users] Search all dialogs - Kamailiio 3.0.3
by Daniel-Constantin Mierla 28 Mar '12

28 Mar '12

Hello, On 3/23/12 2:00 PM, Charles Chance wrote: > > Hi all, > > I was wondering if anyone could offer any advice please. > > We are using the dialog module and need to be able to find out whether > another dialog exists within a particular profile and having a > specific value, so that we can send the new call to the same gateway. > As far as I am aware, the only way to see other dialogs from the > request route is to query the database (having set db mode to > realtime). This is ok, but the profile/value is not stored in the > database table. Is there a way of fetching the information that I am > missing, or should we just add some customisation to the module so > that it also writes this information to db? > in the last version you can attach variables to dialog and they are stored in database -- but that requires you to upgrade. Also, devel version saves the profiles in database, in a json format. What you can do with 3.0.x: - use sqlops to write to db table the info you need -- you would have to update db based on call events, such as initial invite: insert record to db; 200ok to initial invite, confirm the dialog in the record; bye: delete the db record; you need to delete also for the case of negative reply to invite. You can use rtimer to delete old records (those that don't get bye). This is pretty much what dialog module does, but hides it in the c code -- all doable in the config - use htable module to store the details in memory instead of a db table In summary: upgrading to 3.2.x makes your life much easier. Next major release comes with a better dialog storage, you may want to play with it as well. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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siptrace and duplicate_uri - not forwarding the packets
by Mino Haluz 28 Mar '12

28 Mar '12

Hi, I have siptrace working but when I set duplicate_uri, no packet is sent to the destination. Was there some bug fix or anything I have to set further? There is nothing in syslog, siptrace locally is stored fine. modparam("siptrace", "db_url", "mysql://localhost/kamailio") # Database URL # modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) # Flag is used to mark messages to trace modparam("siptrace", "trace_sl_acks", 0) # Do not trace ACKs separately. They are traced during normal sip_trace() modparam("siptrace", "duplicate_uri", "sip:IP1:5060") kamailio 3.2.0 Thanks Mino

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kamailio 3.2.2 undefined symbol: boot_OpenSER while loading module perl
by Karsten Horsmann 28 Mar '12

28 Mar '12

Hi, i am using Centos 6 (32bit) and kamailio 3.2.2 build from the tar.gz. I installed all dependencies for the modules and the make all, make install seems to be fine. But something is wrong with my kamailio - how can i fix this? If i try to load the perl module i get this error messages: ERROR: <core> [sr_module.c:565]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/perl.so>: /usr/local/lib/kamailio/modules_k/perl.so: undefined symbol: boot_OpenSER <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 214, column 12-20: failed to load module ls -la /usr/local/lib/kamailio/modules_k/perl.so -rwxr-xr-x 1 root root 107985 28. Mär 09:38 /usr/local/lib/kamailio/modules_k/perl.so ldd /usr/local/lib/kamailio/modules_k/perl.so linux-gate.so.1 => (0x008fe000) libperl.so => /usr/lib/perl5/CORE/libperl.so (0x00dd8000) libresolv.so.2 => /lib/libresolv.so.2 (0x00343000) libnsl.so.1 => /lib/libnsl.so.1 (0x00cdf000) libdl.so.2 => /lib/libdl.so.2 (0x0024f000) libm.so.6 => /lib/libm.so.6 (0x00fab000) libcrypt.so.1 => /lib/libcrypt.so.1 (0x002c3000) libutil.so.1 => /lib/libutil.so.1 (0x00d21000) libpthread.so.0 => /lib/libpthread.so.0 (0x00110000) libc.so.6 => /lib/libc.so.6 (0x0035c000) libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00aa1000) /lib/ld-linux.so.2 (0x004f7000) libfreebl3.so => /lib/libfreebl3.so (0x00632000) -- Kind Regards Mit freundlichen Grüßen *Karsten Horsmann*

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