sr-users March 2017

sr-users@lists.kamailio.org

87 participants
111 discussions

rtpproxy doesn't record after start_recording
by Marko Tirs 11 Mar '17

11 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

1 0

Dispatcher Module - Hava limit?
by Rodrigo Moreira 11 Mar '17

11 Mar '17

Hello, I am experiencing the following problem: my dispatcher module is losing many calls when the scenario for balancing is about 175 CPS or more. Is there a limit that the module can do? Where do I find this information? Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069

3 6

Multiple parallel branches, one invalid
by Nathan Ward 10 Mar '17

10 Mar '17

Hi, I have a scenario I’m not sure of the best way to solve. I use alias_db and have multiple destinations for a single number, and have append_branches set to 1. I then use lookup_branches to get details from the location table for each branch. I then do t_load_contacts and t_next_contacts, then t_relay to parallel fork the calls to the destination, and all the phones ring at once. This all works great if all the destinations from the dbaliases table are online. However, if one of them is offline, lookup_branches says "Not found in usrloc”. This branch is still there, and the call is attempted. In my environment, this then does a DNS lookup, can’t resolve the domain, and the branch fails. This is the first branch that is tried, and when this branch has an error like this, it causes t_relay() to fail - I check for the output of t_relay() and reply with an error - per the default config that Kamailio ships with. I guess what I need to figure out, is if there’s a way to have t_relay fail only if all branches fail? Or, is there a more elegant way to handle this? I should add, I only allow a single registration per subscriber - so, lookup_branches only returns one branch per alias from alias_db_lookup. -- Nathan Ward

2 1

404 not here for BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/ Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95: F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+ rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. 3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487;branch=z9hG4bK-524287-1--- 1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060;rport=5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0.

2 1

sip6 BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95:F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;branch=z9hG4bK-524287-1---1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060 ;rport=5060;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0.

3 2

Bounty for topos fix
by Trevor Peirce 09 Mar '17

09 Mar '17

Hello, I would like to offer a $100 USD bounty if a developer is able to provide a fix for the two topos problems I have experienced. I will add another $50 if this can be solved within 48 hours of the time stamp of this message. The bug report is at: https://github.com/kamailio/kamailio/issues/1005 You may find the most relevant information in my last few updates. Please refer to mar1-nat.txt and mar1-reinvite.txt for the most recent SIP captures, and the post below that for example database records. Steps to reproduce the NAT problem can be found at https://github.com/kamailio/kamailio/issues/1005#issuecomment-283255720 Some fine print to clarify the offer: Payment is contingent on resolving both described use-cases. Payment will be sent via your choice of PayPal, cheque drawn on a US bank, or cheque drawn on a Canadian bank. Payment will be issued once the patch is accepted by the Kamailio project, although the bonus for a quick resolution simply requires a suitable patch to be submitted to the bug tracker within 48 hours from now. Thanks! -- Trevor Peirce AcroVoice Solutions Inc

1 1

Changes in LCR Target List doesn't takes place immediately
by Nitesh Lohchab 09 Mar '17

09 Mar '17

Hello, I believe LCR Target list is stored in the database and any changes made here should affect the behaviour of the calls imedialtely. I have changed one of my Target List Rule , where I changed my gateway Id to another gateway. But the calls are still being routed via previous gateway. Is there something I can do to make the changes in production hours. Sometimes one of carriers delivers bad call quality I would like to switch the carriers. -- Regards, Nitesh Lohchab

2 1

rtpengine sending rtp to wrong endpoint after reinvite
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, One of our customers is using a SEMS box to place two outbound calls using our sip trunk. Once the first call is connected a second call is placed and when the second call answers their server sends a re-invite to switch audio ports so the rtp traffic doesn't flow through their server anymore but is routed inside our platform. Basically, they just switch SDP's of both calls. It seems like a random issue, and is not really reproducible, except for placing multiple calls and sometimes both parties can hear each other, other times they can't, because rtpengine fails (I think) to update the endpoint and keeps sending rtp back to their server for one of the call legs. We tried to reproduce the case using a freeswitch box and it worked every time. After the reinvite, the rtp remained within our platform. The signaling in both cases still goes through the freeswitch or sems for call control. Does anyone have experience with this case? Or seen the issue before where rtpengine keeps sending rtp to the original endpoint? Regards, Grant Bagdasarian CM

2 3

Access avp between sip transactions
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, Is it possible to access an avp variable from a different sip transaction? On an INVITE is set an avp, but on ReINVITE I would like to access this avp again and get the value. Currently the value is null, which makes sense. Not sure if this is possible and if it violates any rules inside the Kamailio engine? I guess I could use a htable, but I'm just wondering if there is a built-in way for doing this. Regards, Grant Bagdasarian CM

2 1

Kamailio Presence help required.
by SamyGo 08 Mar '17

08 Mar '17

Hi, I'm in need of making/tweak an existing Kamailio Presence setup which is giving some tough time. *Whats already working:* BLF dialog states changes are already sent across the users. SCA is working as well. *What isn't working:* When a User comes online then it sends SUBSCRIBE with *Event: dialog* and don't get notified of its subscribers state right then unless the monitored extensions make a call (BLF works) *Why is it not working:* As evident from wireshark traces, the user IP phones (Test sets: Polycoms, Yealink, CISCO, Grandstream) don't send our *Event: presence* rather only *Event:dialog* and Kamailio do not send NOTIFY out to everyone. Though yes there is an internally generated PUBLISH seen and handled properly upon registration state changes. Jitsi has been tested and Kamailio send out these registration state change info to jitsi, somce jitsi sends *Event:presence *in SUBSCRIBE. I need dialog based NOTIFY to be sent out on registration state-change. Need pointers and help on the topic, looking forward to some feedback. Regards, Sammy

4 8

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sr-users March 2017