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37519 discussions

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

15 Jul '15

We wish to use SIP-Router 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 SIP-Router 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> http://www.vectramind.com/

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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> http://www.vectramind.com/

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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 with Yealink? If i could get traces in anyway to help let me know. BR Alberto INVITE sip:212@10.0.1.34:15060 SIP/2.0. Record-Route: <sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes>. Record-Route: <sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes>. Via: SIP/2.0/UDP x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1. Via: SIP/2.0/TLS 10.0.1.111:11891 ;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360. From: "214" <sip:214@1x.x.x.x:8001>;tag=1072578853. To: <sip:212@x.x.x.x:8001>. Call-ID: 0_1310998066(a)10.0.1.111. CSeq: 2 INVITE. Contact: <sip:214@83.x.x.x:11891;transport=TLS>. Content-Type: application/sdp. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. Max-Forwards: 69. User-Agent: Yealink SIP-T21P_E2 52.80.0.3. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 553. . v=0. o=- 20143 20143 IN IP4 x.x.x.x. s=SDP data. c=IN IP4 x.x.x.x t=0 0. m=audio 8530 RTP/AVP 0 8 18 101. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3. a=crypto:3 F8_128_HMAC_SHA1_80 inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=nortpproxy:yes

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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 and chanspy applied so it can be listened to. The problem I have is when I send an INVITE from Asterisk Server, and then kamailio sends an INVITE back to the same Asterisk Server I get the following failure in asterisk console; Not accepting call completion offers from call-forward recipient I appreciate its potentially more an Asterisk question but I wondered if anyone had experienced the same issue? I have setup a completely load balanced environment so dont want to use Primary/Secondary Server routing. Thanks Jon

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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=1478444456. To: <sip:2@x.x.x.x:8002>. Call-ID: 244257786-52768-56(a)IA.CG.BIE.BCH. CSeq: 550 INVITE. Contact: "Anonymous" <sip:212@80.26.x.x:52768>. X-Grandstream-PBX: true. Max-Forwards: 70. User-Agent: Grandstream GXP2140 1.0.4.23. Privacy: id. P-Preferred-Identity: <sip:212@x.x.x.x:8002>. Supported: replaces, path, timer. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 335. . v=0. o=212 8000 8000 IN IP4 80.26.x.x. s=SIP Call. c=IN IP4 80.26.x.x. t=0 0. m=audio 55422 RTP/AVP 0 8 18 9 2 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:9 G722/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19)) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }

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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 restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17170]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking international restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17171]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking service restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17171]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking international restrictions As you can see this shows different threads handling the exact same callid (printed by $ci). And both run the same checks in our routing plan. The main problem is, I'm not able to reproduce this on the testplatform, but only see it happen on the main systems. Also, it doesn't happen that often, in this case there are about maybe 10 similar calls like this over the run of a few weeks. This case show the call being handled by 2 threads, but we've even seen with one call it being handled by 4 threads, all with the same $ci and the same messages, just different PIDs. I've checked and made sure the routing isn't recursively calling itself. But even if that was true, I would imagine the same PID for all the log entries. Running production in debug mode isn't an option, as it will interfere too much with the day-to-day business. So I'm hoping to get a few pointers here maybe, I'm happy to look at the source code more in depth, I just need a little help in the right direction to not get lost in other code. Any help is much appreciated. Cheers, Dirk

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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 defines and libs need to be added to internal libraries -- see the Makefiles inside the folders of lib/ directory in kamailio source tree. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

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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 interface if a packet arrives at the internal interface. Because the SIP server has a fix IP, it's easy to route to this server, but the way back is more complicated and I think my algorithm is not working. But the first problem is the routing behavior from the private server if I try to make a call from intern to extern. The server sends the request to the external IP address from the client and the IP address from the pass through Kamailio proxy is not considered. 8(2720) ERROR: <core> [udp_server.c:576]: udp_send(): sendto(sock,0x7f4724dd9238,1328,0,192.168.1.20:5060,16): Network is unreachable(101) 8(2720) ERROR: tm [../../forward.h:208]: msg_send(): udp_send failed 8(2720) WARNING: tm [t_fwd.c:1610]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed 8(2720) exec: *** cfgtrace:request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=553 a=24 n=sl_reply_error 8(2720) ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) 8(2720) exec: *** cfgtrace:request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=555 a=2 n=exit 7(2718) exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=474 a=5 n=route To change the IP address which is used from the private server as the destination address, I added the following lines to the Kamailio.cfg. modparam("registrar", "use_path",1) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received",1) modparam("path", "use_received", 1) But the routing behavior has not changed. (perhaps this configure was not the right) Is there a way to configure the private server that the last IP address from the record-route header is used? Cheers, Kai

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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")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } route[ADDCHECKSUM] { if($avp(dst_accountcode) || $avp(src_accountcode)) { #some magic remove_hf("X-rand"); remove_hf("X-csum"); append_hf("X-rand: $var(rand)\r\n"); append_hf("X-csum: $var(checksum)\r\n"); } return; } This works well unless the destination is a redirecting sip server. failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } if (t_check_status("3[0-9][0-9]")) { #... route(RELAY); exit; } -user calls destination. -dispatcher sets destination is send to redirectserver and routes to RELAY, addchecksums add some headers called from relay. -redirectserver replies 302. -failure route triggers and ends with a route to RELAY. Any existing old values should be removed and the newer ones added. But end the result is that I have two sets op X-rand/csum headers for the INVITE to the redirection contact. How do I get rid of every X-csum/rand header except the last added ones? Version tested: 4.2.5.

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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; } } -- Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

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