sr-users July 2015

sr-users@lists.kamailio.org

86 participants
151 discussions

License for iptel SIP server when used as part of a commercial product development
by Mahesh Kumaraguru 15 Jul '15

15 Jul '15

We wish to use iptel SIP server for our commercial product development. We plan to develop SIP clients for Android, iOS, Windows phone and PC browser to have voice and video calls. Please let us know the licensing terms under which iptel SIP server is available, especially in the context of us using it for our commercial product. Many Thanks and Best Regards, Mahesh Kumaraguru +(91)-942-417-4471 +(91)-835-782-9220 mahesh.kumaraguru(a)vectramind.com<mailto:mahesh.kumaraguru@vectramind.com… [View More]

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Issues with Yealink SDP and sdp_with_transport()
by Alberto Sagredo 14 Jul '15

14 Jul '15

Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is send SAVP or not to divert to and rtp proxy or rtpengine, as you know rtpproxy supports recording and rtpengine does not yet. So when using if(sdp_with_transport("RTP/SAVP")) with Grandstream Phones all worked fine, but when configuring Optional or Compulsory SRTP in Yealink it seems to do not detect i have seen that crypto lines are not in the final SDP but do not know if thats the reason Did you have a similar issue … [View More]

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Kamailio Load balancing Across Multiple Asterisk Servers -Chanspy
by Jonathan Hunter 14 Jul '15

14 Jul '15

Hi guys, I have a kamailio load balancing requests across 2 Asterisk Servers, and all is working well. Only problem now is when utilizing feature codes to look to listen into calls using the chanspy application I am having issues. I am passing calls to *2222 into Asterisk, where an IVR answers the call, then sends an INVITE back to kamailio to check to see if the extension I want to listen into is in an active call (hash table), and the call will then be forwarded to the Asterisk in question … [View More]

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Nat Detect does not work with GS Phones
by Alberto Sagredo 14 Jul '15

14 Jul '15

I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on this phones Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind nat so NATMANAGE is not called Here its some trace. ANy clue how to handle this phones if they activate STUN for example U 80.26.x.x:52768 -> 192.168.0.170:8002 INVITE sip:2@x.x.x.x:8002 SIP/2.0. Via: SIP/2.0/UDP 80.26.x.x:52768;branch=z9hG4bK358742535;rport. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=… [View More]

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Possible problem with multiple threads handling same SIP message
by Dirk Teurlings - SIGNET B.V. 14 Jul '15

14 Jul '15

Hello, Running kamailio 4.2.5 for a few weeks now, coming from 4.2.3. Since the upgrade we increased the logging substantially in our routing to get more insight into what is going on. We noticed that in a few cases the log showed a SIP message was being threated by multiple forks, each considering if it was a non-existing call. e.g. Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17170]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking service … [View More]

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Re: [SR-Users] code coverage in kamailio using gcov
by Daniel-Constantin Mierla 14 Jul '15

14 Jul '15

Hello, On 09/07/15 12:54, Surendra Pullaiah wrote: > > Hello > > > > I am trying to implement code coverage tool in > kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and > it got compiled successfully. > > My problem is when I am running kamailio it is saying > lib64/kamailio/libkcore.so.1: undefined symbol: __gcov_merge_add. > > > > Can anyone help me on this. > Maybe those … [View More]

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Kamailio as pass through proxy
by kai.ohnacker＠cbc.de 14 Jul '15

14 Jul '15

Hello, I try to setup a SIP infrastructure with Kamailio server in private network and a pass through Kamailio proxy in the DMZ for NAT handling. (-> TEST.jpg) The pass through proxy uses a new routing algorithm and this works fine for external calls to internal calls. (->Kamailio.cfg.txt) The logic behind is if a packet on external interface arrives, the proxy change the URI and forward the packet to the internal SIP server. Otherwise, the proxy shall forward the packet to external … [View More]

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Removing headers in a failure route on 3xx redirect
by Daniel Tryba 13 Jul '15

13 Jul '15

I'm adding and removing headers in the standard route[RELAY], but if there is a redirection involved headers aren't removed. route[RELAY] { #stdstuff route(ADDCHECKSUM); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { … [View More]

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Pipelimit
by Fred Posner 13 Jul '15

13 Jul '15

I know in ratelimit, you can define a method within queues... for example, modparam("ratelimit", "queue", "3:INVITE") to match on INVITE method. With pipelimit, is the method called from the config, such as: if(is_method("INVITE")) { # Checking limit on INVITE if (!pl_check("$au", "traildrop", "$var(limita)")) { pl_drop(); exit; } } if(is_method("REGISTER")) { # Checking limit on REGISTER if (!pl_check("$au", "traildrop", "$var(limitb)")) { pl_drop(); exit; } } --… [View More]

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msilo delete rows
by Evgeny Zhavoronkov 13 Jul '15

13 Jul '15

Hi All, I'm fighting with msilo module now. I recive an offline message, but msilo gets 408 status and think that a message wasn't sent succesfully. According to the log, the error somewhere in t_relay, but I couldn't find where exactly. Therefore, msilo doesn't delete rows from silo table in database and everytime the user reregisters he gets offline messages again and again. -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

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sr-users July 2015