sr-users December 2017

sr-users@lists.kamailio.org

73 participants
100 discussions

by Karsten Horsmann

Hello List, I thought about some kind of Kamailio stats source (like registered users, calls active and some other things) to collect them into influx dB and draw them with grafana. How do you solved that? Timer based routes or statsd or whatever? Kind regards Karsten Horsmann

6 years, 8 months

is_audio_on_hold() doesn't support rfc3264

by Julia Boudniatsky

FYI, After some tests found, that function "*is_audio_on_hold()*" (textops) *supports only rfc 254*3, where "hold" is indicated by setting the "c" destination addresses for the media streams to zero (0.0.0.0). The newest *rfc3264*, where "hold" is indicated by a=sendonly/inactive, *is not supported*. Kamailio version 4.4.6. Best regards, Julia

6 years, 9 months

rtjson + uac + event - OK, ACK and BYE display name not updated

by Diego Nadares

Hi Guys, We are modifying "from" and "to" headers with rtjson module via a json struct. It works fine in INVITE, TRYING, RINGING and OK but not in the following messages. Kamailio It's changing ONLY "from" and "to" URI BUT NOT the DISPLAY NAME. I think it's the tm module but I can't figure out how to confirm this. Maybe I need to configure something else in my cfg or could it be a bug? My json struct that is handled by rtjson module response.routing = "serial"; //response.routing = "parallel"; response.routes = []; response.routes[0] = {}; response.routes[0].uri = "sip:1111111111@172.16.213.31:5060"; response.routes[0].headers = { // Headers to be modified "from": { "display": "1111111111", "uri": "sip:111111111@172.16.213.38" }, "to": { "display": "2222222222", "uri": "sip:2222222222@172.16.213.38" }, }; response.routes[0].headers.extra = {};//"X-Hdr-A: abc\r\nX-Hdr-B: bcd\r\n"; This is part of my cfg .... modparam("dialog","dlg_flag", FLD_START) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) .... # account only INVITEs if (is_method("INVITE") && !has_totag()) { setflag(FLT_ACC); # do accounting * setflag(FLD_START); # do accounting* setflag(FLT_ACCMISSED); setflag(FLT_ACCFAILED); } ..... >From source/Aleg: *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* To dest/Bleg: *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Incorrect display name in BYE and ACK: *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* interface: any filter: ( port 5060 ) and (ip or ip6) # U 2016/12/06 15:52:13.638731 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:5555555555@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: "SIPP" <sip:sipp@172.16.213.21:5060>. Max-Forwards: 70. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Remote-Party-ID: <sip:sipp@172.16.213.21 >:5060;party=calling;id-type=subscriber;privacy=off;screen=no>. Content-Length: 549. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.734159 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. . ................ # U 2016/12/06 15:52:13.878600 172.16.213.38:5060 -> 172.16.213.31:5060 INVITE sip:1111111111@172.16.213.31:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Max-Forwards: 69. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Content-Length: 788. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40236 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. a=sendrecv. a=rtcp:40237. a=ice-ufrag:QNCzXhUr. a=ice-pwd:Ci90BRpdZH51Hx5BaPgGMrGCeh. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40236 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40237 typ host. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.879441 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. ................ # U 2016/12/06 15:52:13.936565 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 180 Ringing. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 316. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:20.882165 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 213. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ................ # U 2016/12/06 15:52:21.173798 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 380. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:21.174390 172.16.213.21:5060 -> 172.16.213.38:5060 ACK sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-4. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:21.248172 172.16.213.38:5060 -> 172.16.213.31:5060 ACK sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.98116b5795d4c30f52156187a72059c8.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.178826 172.16.213.21:5060 -> 172.16.213.38:5060 BYE sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.468639 172.16.213.38:5060 -> 172.16.213.31:5060 BYE sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.469153 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.519423 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Content-Length: 0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . ................ exit 13 received, 0 dropped Sorry for my english. Thanks in advance! Diego.

6 years, 9 months

t_relay dying ?

by Jean Cérien

Hello I am using kamailio 5.0.2, on a debian 9 system. Everything was running fine, until one of our voip provider changed his switch. Our kamailio is relaying between several voip providers and several asterisk (only the signalisation, no rtp). When we get an invite from this new switch, we select an asterisk and relay it correctly to this box. However, the OK is relayed back. But when the voip providers sends an ACK for this OK, the t_relay function does not return at all, it just dies with no action and no error. Here is the snippet from route(relay) xlog("L_INFO","route(relay) @@ $rm - Source: $si:$sp, fu:$fu, tu:$tu\n" ); $var(restrelay)=t_relay(); xlog("L_INFO","route(relay) @@ $rm - t_relay result: $var(restrelay)" ); if (!$var(restrelay)) { When processing the initial invite, I do get both INFO messages. When the ACK is processed, I only get one INFO message, and no ACK is relayed - so it seems execution dies in the t_relay What could be wrong ??? J. Details of the ACK: ACK sip:dialednumber@kamailioIP:5060 SIP/2.0 Via: SIP/2.0/UDP VOIPPROVIDERIP:5060;branch=z9hG4bKjq16fi00d85uee181ic0.1 To: <sip:kamailioIP:5060>;tag=as7f10ed48 From: <sip:sourcenumber@orange-multimedia.fr >;tag=SD8u3ob01-7dd8efde-0002-00d5-0000-0000 Call-ID: SD8u3ob01-688d41f5e2164ceeafe76f40b82b3f97-v300g00030 CSeq: 1 ACK Max-Forwards: 66 Content-Length: 0 Route: <sip:kamailioIP;lr=on;ftag=SD8u3ob01-7dd8efde-0002-00d5-0000-0000;did=a0b.49d> (should you have enough width; the diagram will look ok) VOIP PROVIDER KAMAILIO ASTERISK 21:38:54.685149 │ ──────────────────────────> │ │ ▒ +0.000451 │ 100 trying -- your call is │ │ ▒ 21:38:54.685600 │ <────────────────────────── │ │ ▒ +0.000084 │ │ INVITE (SDP) │ ▒ 21:38:54.685684 │ │ ──────────────────────────> │ ▒ +0.000831 │ │ 100 Trying │ ▒ 21:38:54.686515 │ │ <────────────────────────── │ ▒ +0.000471 │ │ 200 OK (SDP) │ ▒ 21:38:54.686986 │ │ <────────────────────────── │ ▒ +0.000394 │ 200 OK (SDP) │ │ ▒ 21:38:54.687380 │ <────────────────────────── │ │ ▒ +0.038694 │ ACK │ │ ▒ 21:38:54.726074 │ ──────────────────────────> │ │ ▒ +0.060155 │ │ 200 OK (SDP) │ ▒ 21:38:54.786229 │ │ <<<──────────────────────── │ ▒ +0.000138 │ 200 OK (SDP) │ │ ▒ 21:38:54.786367 │ <<<──────────────────────── │ │ ▒ +0.005721 │ ACK │ │

6 years, 10 months

WebRTC proxy for legacy systems

by paul＠kristianpaul.org

Hi, I'm looking to use kamailio as a webrtc proxy for legacy sip system that doesnt have this capability, is there a example or blueprint i can follow to get started with this? I'm RTFMing the docs but still need a while to understand kamailio internals :-)

6 years, 10 months

Kamailio World 2018 - Call for Presentations

by Daniel-Constantin Mierla

Hello, I would like to announce that Call for Presentations at Kamailio World 2018 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k06/call-for-speakers/ The 6th edition of the event takes place again in Berlin, Germany, during May 14-16, 2018. Expect over 150 participants, developers and community members as well as representatives from other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

6 years, 10 months

Cnxcc doesn't update amount values in redis

by Donat Zenichev

Hi community. Have a problem with cnxcc call monitoring. __set_max_credit() allocates new call as expected: "DEBUG: cnxcc [cnxcc_mod.c:444]: __dialog_created_callback(): Flag is not set for this message. Ignoring DEBUG: cnxcc [cnxcc_mod.c:1569]: __set_max_credit(): Setting up new call for client [2], max-credit[0.100000], cost-per-sec[0.010000], initial-pulse [6], final-pulse [6], call-id[2f77fd103d0bfa28718d896876b373fe@b2b.user.agent:50600] DEBUG: cnxcc [cnxcc_mod.c:1470]: set_ctrl_flag(): Flag set! DEBUG: cnxcc [cnxcc_redis.c:313]: redis_get_int(): Got INT value: concurrent_calls=0i DEBUG: cnxcc [cnxcc_redis.c:68]: redis_get_or_create_credit_data(): credit_data with ID=[2] DOES NOT exist in the cluster, creating it... DEBUG: cnxcc [cnxcc_redis.c:92]: redis_insert_credit_data(): Inserting credit_data_t using ID [2] DEBUG: cnxcc [cnxcc_mod.c:1125]: __get_or_create_credit_data_entry(): Credit entry didn't exist. Allocated new entry [0x7f17668f6d70] DEBUG: cnxcc [cnxcc_mod.c:1255]: __alloc_new_call_by_money(): New call allocated for client [2]". So that I can see values inside redis: redis.callision.info:6389> HGETALL cnxcc:money:2 1) "concurrent_calls" 2) "0" 3) "consumed_amount" 4) "0.000000" 5) "ended_calls_consumed_amount" 6) "0.000000" 7) "max_amount" 8) "0.000000" 9) "number_of_calls" 10) "1" 11) "type" 12) "1" But cnxcc doesn't update credit amount during the call. modparam("debugger", "mod_level", "cnxcc=4") but there are no rows in kamailio log like: "DEBUG: cnxcc [cnxcc_check.c:94]: check_calls_by_money(): ec=0.000000, ca=0.600000, ca2=0.600000 DEBUG: cnxcc [cnxcc_check.c:107]: check_calls_by_money(): Client [some_client] | Ended-Calls-Credit-Spent: 0.000000 TotalCredit/MaxCredit: 0.600000/3.000000". cnxcc parameters: modparam("cnxcc", "dlg_flag", FLG_DLG) modparam("cnxcc", "redis", "addr=redis.youwillnever.find;port=6389;db=0") modparam("cnxcc", "credit_check_period", 1) function that performs credit setup: if (!cnxcc_set_max_credit("$var(setid)","$var(credit)","$var(cost_per_sec)","$var(i_pulse)","$var(f_pulse)")) { xlog("L_WARN", "----- cnxcc: $fU - Error setting up credit control - R=$ru ID=$ci\n"); sl_send_reply("402", "Payment Required"); } Why is it so? And as see, all values are set zero. Seems to be problem with redis inserts. Any ideas are highly appreciated. Thanks. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua>

6 years, 10 months

NOTIFY generation for REFER

by Konstantin Tumalevich

I try to build custom attended transfer logick with kamailio and asterisk. My backend can do all work (include interconnect between asterisk servers), so I need to fully handle REFER in kamailio. But I can't send back NOTIFY message with the result of a transfer. Is any way to do this logic inside kamailio or I need to write external SIP service? My route looks like: if(!route(FROM_NODE)) { if(is_method("REFER")) { xlog("L_NOTICE", "Got REFER in call ci:$ci for $(hdr(Refer-To){nameaddr.uri})"); sl_send_reply("202", "Accepted [1]"); # HERE I MAKE async request and want to send NOTIFY back exit; } } -- Skype: konstantin.tumalevich

6 years, 10 months

BUG: tls [tls_server.c:1229]: tls_read_f(): SSL_ERROR_WANT_READ but data still in the rbio (0x7ffd354fcd30, 8 bytes at 461)

by Abdoul Osséni

Hello Dear, I use Kamailio 5.1.0 on Linux server: root@sip-africallshop-com:~# kamailio -V version: kamailio 5.1.0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2 I have sometimes the following logs: Dec 14 19:22:16 sd-110402 /usr/local/sbin/kamailio[24197]: BUG: tls [tls_server.c:1229]: tls_read_f(): SSL_ERROR_WANT_READ but data still in the rbio (0x7ffd354fcd30, 8 bytes at 461) Dec 20 10:46:32 sd-110402 /usr/local/sbin/kamailio[24197]: BUG: tls [tls_server.c:1229]: tls_read_f(): SSL_ERROR_WANT_READ but data still in the rbio (0x7ffd354fcd30, 8 bytes at 8) Dec 22 11:02:49 sd-110402 /usr/local/sbin/kamailio[24212]: BUG: tls [tls_server.c:1229]: tls_read_f(): SSL_ERROR_WANT_READ but data still in the rbio (0x7ffd354fcd30, 8 bytes at 3357) Does anybody can explain me these logs? Thanks. Abdoul OSSENI

6 years, 10 months

One way video using Kamailio, WebRTC, and Asterisk

by Wilkins, Steve

First I want to give Denys a huge shout-out for all of the help he has given me. It is wonderful that boards like this exists and people are so willing to help a newbie learn. I am on what I am hoping is my last major issue with WebRTC<=>WebRTC calls (using tryit-jssip Chrome or Firefox). I am using Kamailio 5, and Asterisk 15 (pjsip). I am making calls between two WebRTC clients - Client1, and Client2 (using tryit-jssip) Problem: If Client1 calls Client2, and Client2 'ANSWERS', I only have audio/video on Client1. Client2 gets no audio/video, but is connected. If I switch things up and call Client1 from Client2, the same thing happens (Client2 has audio/video and Client1 does not); I can only get audio/video on the calling laptop; the called laptop has no audio/video, but is connected. I see no errors in any of the logs. I am hoping that someone out there has seen this behavior before and has an idea as to the cause and possible solution. Thank you, -Steve

6 years, 10 months

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sr-users December 2017