- sr-users - kamailio.org

kemi equivalent config
by mohsen khashei 11 Feb '23

11 Feb '23

2 1

KEMI Equivalent of script
by mohsen khashei 10 Feb '23

10 Feb '23

Hi I have an existing kamailio config and I want to migrate to kemi. I want to know what is the equivalent of this code in kemi javascript ? $avp(s:contact) = $hdr(Contact); $avp(s:contact_fix) = $hdr(Contact); avp_subst("$avp(s:contact_fix)", "/@([0-9]+).([0-9]+).([0-9]+).([0-9]+)/@$sel(cfg_get.kamailio.ss_ip)/gi"); remove_hf("Contact"); append_hf("Contact: $avp(s:contact_fix)\r\n"); t_on_reply("REG_FIX"); thanks for your attention.

1 0

Return value from lua/python/js/bash/etc... function
by Dmytro M 10 Feb '23

10 Feb '23

What is the simplest way to return values from external application into Kamailio config and assign it to a variable? I'm looking for the most light weight mechanism to do it and to offload some logic to external app. Something like this would be ideal: $var(myVar), $var(err) = lua_run("lua_func1", "$rU"); OR $var(encString) = exec_cmd("echo 'abc $rU' | base64"); There is this method: http_client_query("http://...", "", "", "$var(result)"); but there is an overhead of networking and HTTP protocol. Please advise. Thanks, Dmytro

3 4

ACK on Reverse Invite is send to NATed Internal port instead of public port
by Angelo Sipper 09 Feb '23

09 Feb '23

Hi, I am having an issue on some outbound calls from nated UAC. On these calls our SBC receive reverse invites from the carrier side within the call. When this happens, my SBC(Kamailio1) sends the invite to UAC and the UAC sends 200 OK (SDP). Unfortunately my SBC(Kamailio1) sends back the ACK not to the correct UAC public port but instead to UAC internal port making the call to crash! My setup has: (TLS)UAC -> (UDP)Kamailio1(UDP) -> (UDP)Kamailio2(TLS) -> Carrier(UDP) Bellow you can find the trace and you can see the ACK send to wrong port on 2023/01/26 09:48:31.254591 Your help will be much appreciated! 2023/01/26 09:48:23.225326 77.1.1.1:2314 -> 17.17.17.211:5060 INVITE sip:300987654321@17.17.17.211:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport Max-Forwards: 70 Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060> From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Remote-Party-ID: <sip:301234567890>;party=called Content-Length: 283 v=0 o=3cxPS 792019937001472 2393157908037633 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:23.240310 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314;received=77.1.1.1 To: <sip:300987654321@17.17.17.211:5060> From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:23.301454 17.17.17.211:5060 -> 17.17.17.181:5060 INVITE sip:300987654321@sip.mydomain.com:5060 SIP/2.0 Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com> From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 300 P-Asserted-Identity: <sip:301234567890@sip.mydomain.com> v=0 o=3cxPS 792019937001472 2393157908037633 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:23.302899 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0;rport=5060 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 To: <sip:300987654321@sip.mydomain.com> From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:24.077941 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 183 Session Progress From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:24.138348 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 183 Session Progress From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:24.139050 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 183 Session Progress From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:24.200575 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 183 Session Progress From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.115324 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.336507 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.432457 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 194.154.142.193 s=SIP Media Capabilities c=IN IP4 194.154.142.196 t=0 0 m=audio 10764 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2023/01/26 09:48:30.433017 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 194.154.142.193 s=SIP Media Capabilities c=IN IP4 194.154.142.196 t=0 0 m=audio 10764 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2023/01/26 09:48:30.457268 77.1.1.1:2314 -> 17.17.17.211:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.481470 17.17.17.211:5060 -> 17.17.17.181:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.44ee18452b274c8de283aa3c27549f4e.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.555470 77.1.1.1:2314 -> 17.17.17.211:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.557824 17.17.17.211:5060 -> 17.17.17.181:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.44ee18452b274c8de283aa3c27549f4e.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.530836 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.531333 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.609667 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6cff4a8159c123f434b422c73e17a08c.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.612312 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.63edf375f6b899dfa9d2d9a2dcee4dfd.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6cff4a8159c123f434b422c73e17a08c.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:32.536512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:32.536839 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:32.623207 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48a5467fc248682a755e70a672268f79.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:32.641928 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.745a55762e847bc3cf56d27926f7e62d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48a5467fc248682a755e70a672268f79.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:34.540735 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:34.541310 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:34.590013 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6c3274e7b34c3b780cfbe9180a4d5eb1.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:34.595763 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.6c6654fa1c2f4b1991d5b6d792f2f081.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6c3274e7b34c3b780cfbe9180a4d5eb1.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:38.561756 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:38.562039 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:38.599703 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.ef7afaa26d39ee76ce9000b1203574e7.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:38.603466 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e022b4e31f8266525e62f213dc4dd8ae.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.ef7afaa26d39ee76ce9000b1203574e7.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:42.584025 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:42.584257 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:42.622809 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.03eec221779917d79f0daa500f7a0cc6.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:42.626545 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.0e211693731e67705c85dd4844e42ce8.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.03eec221779917d79f0daa500f7a0cc6.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:46.629501 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:46.629961 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:46.676968 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.701d02748c09dee63b88ca1e96a4e371.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:46.682458 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.cc80042ad9e1838346f4cf6d66a4c13f.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.701d02748c09dee63b88ca1e96a4e371.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:50.636998 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:50.637287 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:50.675272 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.041bbf55bf3941bfbd16df0f3ca92cf1.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:50.681861 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.be45a1f76f0fcd5ce85292397ff68acc.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.041bbf55bf3941bfbd16df0f3ca92cf1.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:54.677049 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:54.677633 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:54.721399 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6ce35eaf9a5805bf47c868cd80c09478.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:54.729950 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.365607f0beb6e7e5dcb047c1ad6fcf5f.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6ce35eaf9a5805bf47c868cd80c09478.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:58.681010 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:58.681307 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:58.723730 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.abb70cc570d323c867c112765ec50432.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:58.731143 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.12bbae879e42215b50512c04c4fc0214.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.abb70cc570d323c867c112765ec50432.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:02.684483 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:49:02.684985 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:49:02.727999 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.c2b81601667ffb2a688deba47abd85ef.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:02.732652 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.fb4ff819be250670479341711417a554.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.c2b81601667ffb2a688deba47abd85ef.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:03.028200 77.1.1.1:2314 -> 17.17.17.211:5060 BYE sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---7c113301b578bb42;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:49:03.090194 17.17.17.211:5060 -> 17.17.17.181:5060 BYE sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK92d8.6e9f4e9b390a591c1c54d79389ba087d.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:49:03.261453 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE Content-Length: 0 Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bK92d8.6e9f4e9b390a591c1c54d79389ba087d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:03.261941 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE Content-Length: 0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Thank you! Regards, Angelo

2 4

DB_CLUSTER failure with carrierroute (and others?)
by Kaufman 08 Feb '23

08 Feb '23

All, I've been trying to chase down an issue with db_cluster and permissions in combination with something else, where with my full configuration using the same cluster in multiple modules has failures. I've been trying to create a simplified configuration to consistently reproduce the problem without much success, however I did stumble upon THIS particular problem. The configuration here works fine as is - avpops uses the cluster URL, and carrierroute uses the direct SQLite URL. If I define WITH_DBCLUSTER, then carrierroute fails fetching rows using the cluster connection. Note that AVPOPS is only used as a general test to see if the cluster config works. If I remove references to avpops , carrierroute still fails loading from the cluster URL. Thoughts? #!KAMAILIO loadmodule "db_sqlite" loadmodule "pv" loadmodule "db_cluster" loadmodule "carrierroute" loadmodule "avpops" modparam("db_cluster", "connection", "db1=>sqlite:////etc/db.sqlite") modparam("db_cluster", "cluster", "cl1=>db1=9s9s") modparam("db_cluster", "inactive_interval", 30) #!ifdef WITH_DBCLUSTER modparam("carrierroute", "db_url", "cluster://cl1") #!else modparam("carrierroute", "db_url", "sqlite:////etc/db.sqlite") #!endif modparam("carrierroute", "config_source", "db") modparam("avpops", "db_url", "cluster://cl1") modparam("avpops", "avp_table", "usr_preferences") request_route { exit; } Failure log: 0(1) ERROR: carrierroute [cr_db.c:345]: load_route_data_db(): Fetching rows failed 0(1) ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data 0(1) ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data 0(1) ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module carrierroute (/usr/lib/kamailio/modules/carrierroute.so) Kaufman

3 5

Drops on async_route with delay
by Alex Balashov 07 Feb '23

07 Feb '23

Hi, Some percentage of requests processed with async_route("REQ_PROCESS", "5") seem to end up with a resumed transaction, though most do. Not sure what the exact percentage is. The ones that don't Is this due to excessive requests? Is there a limit on internal IPC queue depth? Is it conceptually similar to a generic shared blocking queue internally, along the lines of 'mqueue'? Is there any reasonable way to troubleshoot this? Thanks! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

3 3

Dispatcher + ENUM
by Jawaid Bazyar 07 Feb '23

07 Feb '23

Hi, Can DISPATCHER be used in conjunction with ENUM? E.g., to avoid sending requests to SIP destinations that are returned by ENUM, but, which DISPATCHER knows are down? Thanks, Jawaid

2 1

nating private uas to public ip addr
by Jack Murphy 06 Feb '23

06 Feb '23

I have a multi-homed kam with a uas behind one private interface and a public interface for the uac+sip provider. When the uac initiates a call, kam sends the invite to the private uas which sets up a b-leg call back to kam with a Contact header that has the uas' private ip address. I want to change that private ip address to be Kamailio's advertised/public ip address before it gets forwarded to the sip provider. I am rewriting Contact in a route from request_route like this: > if ($rP != $null) { > $var(ct) = ";transport=" + $rP; > }else{ > $var(ct) = ""; > } > $var(cr) = "<sip:" + $fU + "@" + $RAi + ":" + $RAp + $var(ct) + ">"; > > remove_hf("Contact"); > append_hf("Contact: $var(cr)\r\n"); This seems to work fine for the INVITE going to the provider. There's also the 200 OK that the uas sends for the uac (a-leg) with a private ip address in Contact -- I want to rewrite that before sending it to the UAC, too. Using the same method as above (but inside onreply_route), what happens is the UAC receives the modified OK and sends back an ACK with the modified Contact (kam's public ip addr) in the ACK r-uri, and kam gets in a loop of sending an ACK from its private ip address to its public ip address and replying to itself. call flows: https://imgur.com/a/Yj2DJpn Is there a right way to do this?

2 2

htable: autoexpire vs updatedelete
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/htable#htable.p.htable says for modparam('htable', 'htable', '…autoexpire … updateexpire…') autoexpire -time in seconds to delete an item from a hash table if no update was done to it. If is missing or set to 0, the items won't expire. updateexpire - if set to 1 (default), the time until expiration of an item is reset when that item is updated. Certain uses of htable may dictate that updates should not reset the expiration timeout, however, in which case this attribute can be set to 0. My reading is that, when a value is updated and autoexpire is positive, autoexpire will reset the timer, but updateexpire may or may not reset the same timer. That is: modparam('htable', 'htable', 'a=>autoexpire=5;updatedelete=1') modparam('htable', 'htable', 'b=>autoexpire=5;updatedelete=0') If b was not updated for 5s, will it be removed, as the documentation for autoxpire suggests, irrespective of updatedelete? Greetings Dilyan

2 2

Depending on topoh.mask_key Kamailio either cannot parse RURIs or reads invalid data
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, when I use dns_try_naptr=on corelog=-1 debug=-1 enable_tls=yes use_dns_cache=off dns_cache_init=off modparam("topoh", "mask_key", "TEAI32l)- eauiDEUIA!?()") and run Kamailio under valgrind, Kamailio logs: 20(21) ERROR: kemix [kemix_mod.c:229]: ki_kx_get_ruri_attr(): failed to parse the R-URI 20(21) ERROR: rr [loose.c:1011]: loose_route_mode(): failed to parse Request URI The workflow is: I start Kamailio. A UDP-client registers. Then a websocket client calls the UDP-client. Finally the websocket client hangs up. Valgrind does not report anything suspicious. When I use instead modparam("topoh", "mask_key", "TEAI32l") for the same workflow, with the same configuration, valgrind logs: ==14== Invalid read of size 4 ==14== at 0x4867E5: atomic_cmpxchg_int (atomic_x86.h:224) ==14== by 0x486830: futex_get (futexlock.h:99) ==14== by 0x490CE4: dns_hash_get (dns_cache.c:673) ==14== by 0x4972A6: dns_get_entry (dns_cache.c:2001) ==14== by 0x499288: dns_srv_get_he (dns_cache.c:2455) ==14== by 0x597AAD: no_naptr_srv_sip_resolvehost (resolve.c:1599) ==14== by 0x598332: naptr_sip_resolvehost (resolve.c:1675) ==14== by 0x5983C5: _sip_resolvehost (resolve.c:1707) ==14== by 0x49943B: dns_srv_sip_resolvehost (dns_cache.c:2516) ==14== by 0x49B3DD: dns_sip_resolvehost (dns_cache.c:2738) ==14== by 0x59846A: sip_hostport2su (resolve.c:1727) ==14== by 0x4CB951: forward_request (forward.c:515) ==14== by 0x99993CD: t_relay_to (t_funcs.c:300) ==14== by 0x99EBA32: _w_t_relay_to (tm.c:1764) ==14== by 0x99F4DEB: ki_t_relay (tm.c:2917) ==14== by 0xA80A94B: sr_kemi_lua_exec_func_ex (app_lua_api.c:1022) ==14== by 0xA81237D: sr_kemi_lua_exec_func (app_lua_api.c:1706) ==14== by 0xA81B93F: sr_kemi_lua_exec_func_209 (app_lua_kemi_export.c:1717) ==14== by 0xA8383C0: luaD_precall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA84C4CA: luaV_execute (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA838C90: luaD_callnoyield (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA837029: luaD_rawrunprotected (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8391DF: luaD_pcall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8333DE: lua_pcallk (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA806F0C: app_lua_run_ex (app_lua_api.c:773) ==14== by 0xA825F06: sr_kemi_config_engine_lua (app_lua_mod.c:119) ==14== by 0x5048BE: sr_kemi_route (kemi.c:3784) ==14== by 0x588909: receive_msg (receive.c:502) ==14== by 0xA609ADF: ws_frame_receive (ws_frame.c:644) ==14== by 0x4BCAAB: sr_event_exec (events.c:299) ==14== Address 0x0 is not stack'd, malloc'd or (recently) free'd ==14== ==14== ==14== Process terminating with default action of signal 11 (SIGSEGV): dumping core ==14== Access not within mapped region at address 0x0 ==14== at 0x4867E5: atomic_cmpxchg_int (atomic_x86.h:224) ==14== by 0x486830: futex_get (futexlock.h:99) ==14== by 0x490CE4: dns_hash_get (dns_cache.c:673) ==14== by 0x4972A6: dns_get_entry (dns_cache.c:2001) ==14== by 0x499288: dns_srv_get_he (dns_cache.c:2455) ==14== by 0x597AAD: no_naptr_srv_sip_resolvehost (resolve.c:1599) ==14== by 0x598332: naptr_sip_resolvehost (resolve.c:1675) ==14== by 0x5983C5: _sip_resolvehost (resolve.c:1707) ==14== by 0x49943B: dns_srv_sip_resolvehost (dns_cache.c:2516) ==14== by 0x49B3DD: dns_sip_resolvehost (dns_cache.c:2738) ==14== by 0x59846A: sip_hostport2su (resolve.c:1727) ==14== by 0x4CB951: forward_request (forward.c:515) ==14== by 0x99993CD: t_relay_to (t_funcs.c:300) ==14== by 0x99EBA32: _w_t_relay_to (tm.c:1764) ==14== by 0x99F4DEB: ki_t_relay (tm.c:2917) ==14== by 0xA80A94B: sr_kemi_lua_exec_func_ex (app_lua_api.c:1022) ==14== by 0xA81237D: sr_kemi_lua_exec_func (app_lua_api.c:1706) ==14== by 0xA81B93F: sr_kemi_lua_exec_func_209 (app_lua_kemi_export.c:1717) ==14== by 0xA8383C0: luaD_precall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA84C4CA: luaV_execute (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA838C90: luaD_callnoyield (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA837029: luaD_rawrunprotected (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8391DF: luaD_pcall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8333DE: lua_pcallk (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA806F0C: app_lua_run_ex (app_lua_api.c:773) ==14== by 0xA825F06: sr_kemi_config_engine_lua (app_lua_mod.c:119) ==14== by 0x5048BE: sr_kemi_route (kemi.c:3784) ==14== by 0x588909: receive_msg (receive.c:502) ==14== by 0xA609ADF: ws_frame_receive (ws_frame.c:644) ==14== by 0x4BCAAB: sr_event_exec (events.c:299) The stacktrace with the values of the variables is available at https://github.com/kamailio/kamailio/issues/3350 . Or at least these things seem very similar to me. Any idea? I can share the OCI-image and the full configuration. Greetings Дилян

3 4

topoh: th_mask_decode (th_mask_encode (x)) ≠ x
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, after receiving INVITE sip:test-gnome-calls@bapha.be SIP/2.0 Via: SIP/2.0/WSS bapha.be;branch=z9hG4bK974100 Max-Forwards: 70 To: <sip:test-gnome-calls@bapha.be> From: "Online https://sip.bapha.be" <sip:online@bapha.be>;tag=12q2efc423 Call-ID: fkhe8faq0fh80glmq59i CSeq: 5622 INVITE Contact: <sip:4q5ne5no@bapha.be;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.7.8 Content-Type: application/sdp Content-Length: 1964 v=0 o=mozilla...THIS_IS_SDPARTA-99.0 8263261229339246488 0 IN IP4 0.0.0.0 s=- … kamailio calls th_mask_encode( char *in = "sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60722~2>;+sip.instance=\"<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs=\"conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime\n"\ "Content-Type: application/sdp and so on", int ilen = 107, str prefix = { .len = 23, .s ="sip:127.3.4.84;line=sr-" }, int *olen). It returns olen=167, and the string "sip:127.3.4.84;line=sr- if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p-1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWO9-eo9*". Then Kamailio forwards (sends): Contact: <sip:127.3.4.84;line=sr-if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p- 1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWO9-eo9*>;+sip.instance="<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs="conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime" which at some point leads to ACK sip:127.3.4.84;line=sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0 and BYE sip:127.3.4.84;line=sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0 Below you can find th_test.c . The functions th_mask_encode() and th_mask_decode() there are identical to the same functions in src/modules/topoh/th_mask.c . main() provides some tests: Passing «sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60472~2>;+sip.instance="<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs="conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime» to th_mask_encode() returns sip:127.3.4.84;line=sr-if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p- 1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWXi-eo9* (this is what Kamailio does send out). Passing the last string to th_mask_decode() returns sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50-740fb9bdc562;alias=87.118.1 the ;alias= is incomplete, compared to the origin. The above ACK leads to passing "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p-1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0\n"\ "Route: <sip:Q+L06HX94DMPtQWQTI5OE8SQTI5O4MY=@144.76.142.78;transport=ws;r2=on;lr;nat=yes>\n"\ "Route: <sip:127.3.4.84;line=sr-if7s17IB5XEfVzn0WIgWwqggCTzwV5C2g5R5wCg4WwrQ5CuZ6XIQWvH3WUHkWX9tWOnyMq4RbmWsb34Q2YgoiXP-coTAboPlioPtpY6Z8dCO>\n"\ to th_mask_decode() and the result is callub.p.b;rn:s.d2lQg#cfa=bA07beQ7R409bcQ62;alac=87718*1P.15#~60R22~2 with the dots being unprintable characters. (result is garbage). I have seen that callub.p.b in the unparsable R-URI reports from Kamailio. That is an indication that th_mask_decode() does not decode the Contact encoded by th_mask_encode() correctly. Valgrind does not report improper memory usage. Greetings Dilyan gcc -g -o th-test ./th-test.c && ./th-test th-test.c #include <stdio.h> #include <stdlib.h> #include <string.h> #define TH_EB64I \ "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789.-" char _th_EB64[65] = "EFvXIzGq94xKcW126gV5wCdYpNSbiM8enRUou7LmhajBlPtAsk-OQTf3H0DyrZ.J"; int _th_DB64[256]; char *_th_PD64 = "*"; struct str_ { char* s; /**< Pointer to the first character of the string */ int len; /**< Length of the string */ }; typedef struct str_ str; str prefix = { .len = 23, .s ="sip:127.3.4.84;line=sr-" }; char* th_mask_encode(char *in, int ilen, const str *prefix, int *olen); char* th_mask_decode(char *in, int ilen, const str *prefix, int extra, int *olen); char* th_mask_encode(char *in, int ilen, const str *prefix, int *olen) { char *out; int left; int idx; int i; int r; char *p; int block; *olen = (((ilen+2)/3)<<2) + ((prefix!=NULL&&prefix->len>0)?prefix->len:0); out = (char*)malloc((*olen+1)*sizeof(char)); if(out==NULL) { fprintf(stderr, "malloc error\n"); *olen = 0; return NULL; } memset(out, 0, (*olen+1)*sizeof(char)); if(prefix!=NULL&&prefix->len>0) memcpy(out, prefix->s, prefix->len); p = out + (int)((prefix!=NULL&&prefix->len>0)?prefix->len:0); for(idx=0; idx<ilen; idx+=3) { left = ilen - idx - 1 ; left = (left>1)?2:left; block = 0; for(i=0, r=16; i<=left; i++, r-=8) block += ((unsigned char)in[idx+i]) << r; *(p++) = _th_EB64[(block >> 18) & 0x3f]; *(p++) = _th_EB64[(block >> 12) & 0x3f]; *(p++) = (left>0)?_th_EB64[(block >> 6) & 0x3f]:_th_PD64[0]; *(p++) = (left>1)?_th_EB64[block & 0x3f]:_th_PD64[0]; } return out; } char* th_mask_decode(char *in, int ilen, const str *prefix, int extra, int *olen) { char *out; int n; int block; int idx; int i; int j; int end; char c; for(n=0,i=ilen-1; in[i]==_th_PD64[0]; i--) n++; *olen = (((ilen-((prefix!=NULL&&prefix->len>0)?prefix->len:0)) * 6) >> 3) - n; if (*olen<=0) { return NULL; } out = (char*)malloc((*olen+1+extra)*sizeof(char)); if(out==NULL) { fprintf(stderr, "malloc error\n"); *olen = 0; return NULL; } memset(out, 0, (*olen+1+extra)*sizeof(char)); end = ilen - n; i = (prefix!=NULL&&prefix->len>0)?prefix->len:0; for(idx=0; i<end; idx+=3) { block = 0; for(j=0; j<4 && i<end ; j++) { c = _th_DB64[(int)in[i++]]; if(c<0) { free(out); *olen = 0; return NULL; } block += c << (18 - 6*j); } for(j=0, n=16; j<3 && idx+j< *olen; j++, n-=8) out[idx+j] = (char)((block >> n) & 0xff); } return out; } int main() { //this is from th_mask_init for(int i=0; i<256; i++) _th_DB64[i] = -1; for(int i=0; i<64; i++) _th_DB64[(int)_th_EB64[i]] = i; int olen, o2; char *end; char *input = "sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60472~2>;+sip.instance=\"<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>\";+org.linphone.specs=\"conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime\""; printf("input=[%s]\n", input); char* middle = th_mask_encode (input, 107, &prefix, &olen); printf("olen = %i, middle=[%s]\n", olen, middle); end = th_mask_decode(middle, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); char * b1 = "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9*"; o2 = 0; end = th_mask_decode(b1, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); char* b2 = "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p-1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0\n"\ "Route: <sip:Q+L06HX94DMPtQWQTI5OE8SQTI5O4MY=@144.76.142.78;transport=ws;r2=on;lr;nat=yes>\n"\ "Route: <sip:127.3.4.84;line=sr-if7s17IB5XEfVzn0WIgWwqggCTzwV5C2g5R5wCg4WwrQ5CuZ6XIQWvH3WUHkWX9tWOnyMq4RbmWsb34Q2YgoiXP-coTAboPlioPtpY6Z8dCO>\n"\ "Via: SIP/2.0/WSS bapha.be;branch=z9hG4bK9069155\n"\ "Max-Forwards: 70\n"\ "To: <sip:test-gnome-calls@bapha.be>;tag=Pn4npAj\n"\ "From: \"Online https://sip.bapha.be\" <sip:online@bapha.be>;tag=12q2efc423\n"\ "Call-ID: fkhe8faq0fh80glmq59i\n"\ "CSeq: 5623 ACK\n"\ "Supported: outbound\n"\ "User-Agent: SIP.js/0.7.8\n"\ "Content-Length: 0\n"\ "\n"; o2 = 0; end = th_mask_decode(b2, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); return 0; }

2 1

no autocomplete for kamcmd for latest debian packages
by Ovidiu Sas 04 Feb '23

04 Feb '23

Hello, The latest debian packages are built without autocomplete for kamcmd. Do we have the libreadline-dev package installed on the build machine? I know that we have kamctl rpc and kamcli, but if we are still shipping kamcmd, it should have the autocomplete feature enabled. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

3 2

Sip messages responses counters per interface
by Patrick Karton 02 Feb '23

02 Feb '23

2 1

rtpengine - SRTP <> RTP
by kdrewicz+kamailio＠cludo.pl 01 Feb '23

01 Feb '23

I try to workout if - currently it would work, or - where and how to debug more: I face - 2 interfacec - public internet (so, TLS + sRTP) is desired and private - old infrastructure - i mus only use plain RTP 172.23.9.70 - private ip - from this endpoint of kamailio and rtpengine should send only basic RTP 172.23.210.75:5060 private - target for kamailio 1.2.3.24 obfuscated public IP (TLS + sRTP required) kamailio 5.4.4 (x86_64/linux) rtpengine -v Version: 11.1.1.4-1~bpo11+1 all i do is: if (proto==TLS) { rtpengine_manage("RTP/AVP ICE=remove replace-session-connection replace-origin pad-crypto ptime=20 codec-transcode-PCMA record-call=on allow-transcoding direction=external direction=internal record-call=on"); } else if ($ru =~ "transport=tls") { rtpengine_manage("DTLS=on SRTP AVPF ICE=remove replace-session-connection replace-origin pad-crypto ptime=20 codec-transcode-PCMA record-call=on allow-transcoding direction=internal direction=external record-call=on media-address=1.2.3.24"); } # 1.2.3.24 obfuscated public IP 172.23.210.75:5060 is in dispatch.cfg, as '11' route[SBC_CORE] { append_hf("X-My-SRTP: removed31337\r\n"); ### i see this text in invtes from kamailio 172.23.9.70 towards 172.23.210.75:5060 ### i see only RTP, so as expected if (!ds_select_dst("11", "0")) { xwarn("I:$var(i) DROP(DOWN!) FWD:$rm [$fU->$tU] [SBCVIP] to $du\n"); sl_send_reply("503", "Destination down"); exit; } what i did: certificate is a paid one (the public party needs it) TLS works i deleted - entries in (not kamailo) cryptosuite that caused this: 13:08:05 localhost rtpengine[15140]: ERR: [51ad8758-b64d-4d2f-9fd0-41d03a38f74d]: [core] Failed to parse a=crypto attribute, ignoring: unknown crypto suite Tried to search for any "ready" examples for this - only found old threads and - that this should be possible, but - no example for woking config. what i see: Jan 19 19:00:57 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000038 sec Jan 19 19:00:58 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000036 sec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Closing call due to timeout Jan 19 19:00:59 localhost rtpengine[17301]: DEBUG: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Redis delete_async=0 Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Final packet stats: Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- Tag 'JVR5LTs', created 60:00 ago for branch '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscribed to '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscription for '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] ------ Media #1 (audio over RTP/SAVP) using unknown codec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 1.2.3.24:30136 <> 52.129.106.28:17030, SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 1.2.3.24:30137 <> 52.129.106.28:17031 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- Tag '', created 60:00 ago for branch '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscribed to 'JVR5LTs' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscription for 'JVR5LTs' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 172.23.9.70:30014 <> :0 , SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 172.23.9.70:30015 <> :0 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Moved metadata file "/var/spool/rtpengine/tmp/rtpengine-meta-c17bab16-5eea-492e-b1c4-ac9387f3e265-7003946f152c6c8d.tmp" to "/var/spool/rtpengine/metadata" Jan 19 19:00:59 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000828 sec Jan 19 19:01:00 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000053 sec route(SBC_CORE); maybe any hint or - someone has working exmaple of kamailio config + rtpengine settings ? i use only userspace daemon rtp forwarding (this is a test, dont need any performance here) Thanks,

5 9

transport=ws causing kamailio to use wrong listen socket
by nathan.bruning＠talksome.com 31 Jan '23

31 Jan '23

Hi all, We're hitting an issue while integrating secure websockets in our existing SIP infrastructure using Kamailio. When the registration comes in, it causes an entry in our AOR table, with ";transport=ws" appended. When we want to send a message to this client (using t_relay), kamailio seems to take 'ws' as being *unsecure* websockets. In turn, this makes Kamailio try to send out the message using a TCP listener - while it should have picked the TLS listener. There are some remarks in the sources about ws vs. wss, so i'm struggling to figure out where things go wrong. I've also created github issue #3340 with more details. Any help would be appreciated. If this turns out to be a Kamailio bug, i'm happy to provide a patch.

6 7

How can I setup Kamailio for failover on LCR ?
by scramatte 31 Jan '23

31 Jan '23

Hello, How can I setup Kamailio for failover on LCR ? Currently my Kamailio 4.x works with lcr but doesn’t care about gateway status. I mean that if a gateway match rule , it sends the call through without check if the gateway is available an never try the next one if the first fail. I’ve tried to implement inactivate_gw() But I received error that lcr_id_avp is not defined. I’ve added the parameters to lcr module for Ping and enable defunc but I’m unable to get it working. By the way I ‘ve trad that defunct is not a good practice as hacker sending malformed sip header can throw down the gateway. So what is the good and working way to setup outbound call failover ? Regards Sebastien

1 0

Evapi Dispatcher 100% cpu.
by Sebastian Sastre 30 Jan '23

30 Jan '23

Hello We are currently using EVAPI to push messages into kamailio from a go app. For the most part ift works without problems, messages get delivered into the event route and we use it to update presence status. Recently we started noticing delays on the evapi processing. After 500 requests x min the pid corresponding to the EVAPI dispatcher gets pegged at 100% CPU and processing of requests starts slowing down until in some cases the connection just drops and we have to reconnect. After that pid drops, the dispatcher goes back to working. Unfortunately the error we get back is very generic and is not telling us much logger.go:39: ERR Handler returned error (write tcp 127.0.0.1:48448->127.0.0.1:8228: write: broken pipe) This only happens on prod servers and with high load which makes it hard to debug. There is no other slowness or increase of traffic or drop SIP traffic, only evapi dispatcher at 100% and the EVAPI workers on idle. Curious if anyone has run into this issue. We've tried different versions (5.4 , 5.5 and 5.6) and it happens on all of them. Any feedback appreciated ! Thanks !

2 1

Is preloading openssl_mutex_shared necessary for Kamailio 5.6 or is it not necessary?
by Дилян Палаузов 30 Jan '23

30 Jan '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/tls says: For OpenSSL (libssl) v1.1.x, it is required to preload 'openssl_mutex_shared' library shipped by Kamailio. For more details see 'src/modules/tls/utils/openssl_mutex_shared/README.md'. 'src/modules/tls/utils/openssl_mutex_shared/README.md says: **IMPORTANT: the workaround of using this preloaded shared library is no longer needed starting with Kamailio v5.3.0-pre1 (git master branch after September 14, 2019). The code of this shared library has been included in the core of Kamailio and the same behaviour is now achieved by default.** This is a shared library required as a short term workaround for using Kamailio with OpenSSL (libssl) v1.1. It has to be pre-loaded before starting Kamailio. In v1.1, libssl does not allow setting custom locking functions, using internally pthread mutexes and rwlocks, but they are not initialized with process shared option (PTHREAD_PROCESS_SHARED), which can result in blocking Kamailio worker processes. Is preloading openssl_mutex_shared necessary for Kamailio 5.6 or is it not necessary? If it is not necessary, why isn’t it removed from the source code? Greetings Дилян

1 0

Websocket-answers sent by K. log unparseable errors
by Дилян Палаузов 29 Jan '23

29 Jan '23

Hello, when I connect to Kamailio over websocket, according to the websocket protocol K. sends “HTTP/1.1 101 Switching Protocols” and K. logs: 17(18) DEBUG: websocket [ws_handshake.c:179]: ws_handle_handshake(): found Upgrade: websocket 17(18) DEBUG: websocket [ws_conn.c:195]: wsconn_add(): connection id [14] 17(18) DEBUG: websocket [ws_conn.c:213]: wsconn_add(): new wsc => [0x7fcad6e10160], ref => [0] 17(18) DEBUG: websocket [ws_conn.c:233]: wsconn_add(): added to conn_table wsc => [0x7fcad6e10160], ref => [1] 17(18) DEBUG: sl [sl.c:310]: send_reply(): reply in stateless mode (sl) 17(18) DEBUG: <core> [core/msg_translator.c:162]: check_via_address(): (22.222.222.222, 22.222.222.222, 0) 17(18) DEBUG: <core> [core/tcp_main.c:1644]: _tcpconn_find(): found connection by id: 14 17(18) DEBUG: <core> [core/tcp_main.c:2528]: tcpconn_send_put(): send from reader (18 (17)), reusing fd 17(18) DEBUG: <core> [core/tcp_main.c:2763]: tcpconn_do_send(): sending... 17(18) DEBUG: <core> [core/tcp_main.c:2796]: tcpconn_do_send(): after real write: c= 0x7fcad6da7b38 n=238 fd=9 17(18) DEBUG: <core> [core/tcp_main.c:2797]: tcpconn_do_send(): buf= 17(18) DEBUG: <core> [core/parser/parse_fline.c:255]: parse_first_line(): bad request first line 17(18) DEBUG: <core> [core/parser/parse_fline.c:258]: parse_first_line(): at line 0 char 22: 17(18) DEBUG: <core> [core/parser/parse_fline.c:264]: parse_first_line(): parsed so far: HTTP/1.1 101 Switching 17(18) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 22) 17(18) DEBUG: <core> [core/parser/msg_parser.c:675]: parse_msg(): invalid message 17(18) ERROR: <core> [core/parser/msg_parser.c:749]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols Sia: SIP/2.0/TLS 22.222.222.222:33776 Sec-WebSocket-Protocol: sip Upgrade: websocket Connection: upgrade Sec-WebSocket-Accept: NeswoJX5ZQp7ER8hTKM5B3a3HDQ= Content-Length: 0 > 17(18) ERROR: <core> [core/msg_translator.c:3256]: build_sip_msg_from_buf(): parsing failed 17(18) DEBUG: app_lua [app_lua_api.c:1924]: sr_kemi_lua_exit(): script exit call 17(18) DEBUG: app_lua [app_lua_api.c:789]: app_lua_run_ex(): ksr error call from Lua: ~~ksr~exit~~ 17(18) DEBUG: app_lua [app_lua_mod.c:164]: sr_kemi_config_engine_lua(): execution of route type 513 with name [ksr_xhttp_event] returned 1 22.222.222.222 is my public address (the address of my router). It is not the IP-address of Kamailio. While the returned address is not parseable in terms of SIP, it is a valid Websocket-answer message. This valid condition triggers logging an error, but does not represent an error condition. How can I avoid that the above non-error is logged as an error? Thanks for your feedback Дилян

1 0

Locally generated BYE sent to private IP
by James Lipski 27 Jan '23

27 Jan '23

Hi all, I'm doing some experimenting with kamailio -- currently looking for a replacement of our existing (aging) SIP infrastructure. One of the experiments that I'm doing is testing the dialog module's dlg_set_timeout function. The issue that I'm having is while I do see that kamailio is generating a BYE, and sends it out to the B-leg of the call properly, the BYE goes to a private address on the A-leg. To Note: - kamailio is behind a NAT (on AWS) - both UA endpoints are remote and are both behind a NAT - Not sure if this matters, but I do have the topos module enabled, though I see the same behavior regardless if it's enabled or not. - Have set nathelpers (fix_nated_register, fix_nated_contact,etc...) which helps calls between endpoints and the endpoints themselves are able to send/receive BYEs when they generate them; however doesn't seem to help if kamailio locally generates the BYE. The only thing I do see within my logs is when the BYE is being generated by kamailio, it uses the private IP of my UA and not public for the caller. Callee, the public address is referenced. If more information is needed, please let me know (just didn't want to spam my script/ logs). Thanks. Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:399]: send_bye(): sending BYE to caller Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:458]: t_uac_prepare(): next_hop=<sip:1000@10.0.0.47:38606;line=9k4wsgc9> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:158]: dlg2hash(): hashid 40484 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 1 (request) flags 1 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:679]: parse_msg(): SIP Request: Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:680]: parse_msg(): method: <BYE> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:682]: parse_msg(): uri: <sip:1000@10.0.0.47:38606;line=9k4wsgc9> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:684]: parse_msg(): version: <SIP/2.0> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Via] type 1 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_via.c:1300]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK42e9.4a962087000000000000000000000000.0>; state=16 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:555]: parse_headers(): Via found, flags=2 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:557]: parse_headers(): this is the first via Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [To] type 3 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=ueisi99vbf Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=29 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): <To> [61]; uri=[sip:1000@fakedomain.com] Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (44)[<sip:1000@fakedomain.com>], to tag (10)[ueisi99vbf] Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [From] type 4 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [CSeq] type 5 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <BYE> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Call-ID] type 6 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Content-Length] type 12 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:187]: get_hdr_field(): content_length=0 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Max-Forwards] type 8 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1132060721 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=29 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:1013]: tps_request_sent(): handling outgoing request (1, 1) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:417]: tps_pack_message(): compacted headers - x_via1: [SIP/2.0/UDP <KAMAILIO PUBLIC IP>:5060;branch=z9hG4bK42e9.4a962087000000000000000000000000.0](85) - x_via2: [](0) - x_vbranch1: [z9hG4bK42e9.4a962087000000000000000000000000.0](46) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:539]: tps_pack_message(): compacted headers - a_rr: [](0) - b_rr: [](0) - s_rr: [](0) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:544]: tps_pack_message(): compacted headers - as_contact: [](0) - bs_contact: [](0) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:1027]: tps_request_sent(): no x-uuid header - nothing to do Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/msg_translator.c:1799]: check_boundaries(): no multi-part body Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [topos_mod.c:567]: tps_msg_sent(): new outbound buffer generated Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:686]: send_prepared_request_impl(): uac: 0x7f22f94448e8 branch: 0 to 10.0.0.47:38606 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/onsend.c:50]: run_onsend(): required parameters are not available - ignoring Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:432]: send_bye(): BYE sent to caller Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:399]: send_bye(): sending BYE to callee Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:458]: t_uac_prepare(): next_hop=<sip:1001@<UA PUBLIC IP>:5066>Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:158]: dlg2hash(): hashid 40482

3 3

Kamailio not responding after stress test
by Igor Olhovskiy 27 Jan '23

27 Jan '23

Hello, I'm continuing investigations on Kamailio and stress test. Got it again in the state where it's not accepting any new TCP/TLS connections (UDP still works though), but all looks good from lsof/netnstat part, like system is not reporting any zombie connections. Restart of Kamailio process helps This time I got output of kamctl trap Put it here: https://pastebin.com/iYrNZ8U9 kamailio --version version: kamailio 5.6.2 (x86_64/linux) 54a9c1 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_BLOCKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 54a9c1 compiled on 14:01:01 Oct 18 2022 with gcc 4.8.5 It's statically linked with tlsa pointing on openssl-1.1.1q Settings related to TLS are: fork=yes children=4 tcp_children=12 enable_tls=yes tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=10 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 Can you please help to read gdb output and understand where I missed in config? Thanks in advance! -- Best regards, Ihor (Igor)

3 7

Kamailio 5.6 on AlmaLinux 8 TLS crash
by Ihor Olkhovskyi 27 Jan '23

27 Jan '23

Hello! Currently testing Kamailio on Almalinux 8 (stated to be RHEL 8 binary compatible) and got this while trying to get TLS working: /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: ca_path='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:361]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:365]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=1 /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:372]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:379]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:383]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=1 /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:386]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:389]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 /usr/sbin/kamailio[32858]: INFO: tls [tls_domain.c:736]: set_verification(): TLSc<default>: Server MUST present valid certificate /usr/sbin/kamailio[32862]: ERROR: rtpengine [rtpengine.c:3070]: send_rtpp_command(): can't send command "ping" to RTPEngine <udp: 127.0.0.1:19999> /usr/sbin/kamailio[32862]: ERROR: rtpengine [rtpengine.c:2919]: rtpp_test(): proxy did not respond to ping /usr/sbin/kamailio[32883]: INFO: snmpstats [snmpstats.c:519]: spawn_sysUpTime_child(): A snmpVersion parameter was not provided. Defaulting to 3 /usr/sbin/kamailio[32883]: ERROR: snmpstats [snmpstats.c:561]: spawn_sysUpTime_child(): snmpget failed to run. Did you supply the snmpstats module with a proper snmpgetPath parameter? The kamailioSIPServiceStartTime is defaulting to zero /usr/sbin/kamailio[32881]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) /usr/sbin/kamailio[32870]: CRITICAL: <core> [core/mem/q_malloc.c:145]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten(0, abcdefed)[0x7fe8375d3820:0x7fe8375d3858]! Memory allocator was called from tls: tls_init.c:293. Fragment marked by tls: tls_init.c:293. Exec from core/mem/q_malloc.c:391. /usr/sbin/kamailio[32870]: CRITICAL: <core> [core/mem/q_malloc.c:153]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten [0x7fe8375d3790:0x7fe8375d37c8] - fragment marked by tls: tls_init.c:293 systemd[1]: Started Process Core Dump (PID 32923/UID 0). systemd-coredump[32924]: Process 32870 (kamailio) of user 0 dumped core.#012#012Stack trace of thread 32870:#012#0 0x00007fe841786aff raise (libc.so.6)#012#1 0x00007fe841759ea5 abort (libc.so.6)#012#2 0x00000000006eec2f qm_debug_check_frag (kamailio)#012#3 0x00000000006f09a6 qm_malloc (kamailio)#012#4 0x00000000006fc930 qm_shm_malloc (kamailio)#012#5 0x00007fe83b5b9c31 ser_malloc (tls.so)#012#6 0x00007fe83fa4090d CRYPTO_zalloc (libcrypto.so.1.1)#012#7 0x00007fe8364b8163 sc_hash_type (libnetsnmp.so.35)#012#8 0x00007fe8364b49df hash_engineID (libnetsnmp.so.35)#012#9 0x00007fe8364b4b11 search_enginetime_list (libnetsnmp.so.35)#012#10 0x00007fe8364b4f8e set_enginetime (libnetsnmp.so.35)#012#11 0x00007fe8364b4885 init_snmpv3_post_config (libnetsnmp.so.35)#012#12 0x00007fe8364b9c92 snmp_call_callbacks (libnetsnmp.so.35)#012#13 0x00007fe836f3601b register_with_master_agent (snmpstats.so)#012#14 0x00007fe836f37f14 run_alarm_check (snmpstats.so)#012#15 0x0000000000531cde compat_old_handler (kamailio)#012#16 0x000000000053284e slow_timer_main (kamailio)#012#17 0x000000000042e3e1 main_loop (kamailio)#012#18 0x0000000000438d14 main (kamailio)#012#19 0x00007fe841772d85 __libc_start_main (libc.so.6)#012#20 0x000000000041cfce _start (kamailio) /usr/sbin/kamailio[32915]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 24 /usr/sbin/kamailio[32858]: ALERT: <core> [main.c:774]: handle_sigs(): child process 32870 exited by a signal 6 ... version: kamailio 5.6.3 (x86_64/linux) fe0c4d flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: fe0c4d compiled on 11:02:26 Jan 19 2023 with gcc 8.5.0 Same config is starting without issues on CentOS 7. OpenSSL version is 1.1.1k Kamailio is installed from official RPM repos. Is core dump would be useful here or best to compile from source and redo? -- Best regards, Ihor (Igor)

3 6

topos vs topoh, any reason for putting a private IP in the Contact header?
by Benoit Panizzon 27 Jan '23

27 Jan '23

Hi I'm trying to work around Route-Header and Via Issues with the two topology hiding modules topoh and topos and trying to figure out, which one works better for our environment. My conclusion so far: topos creates very clean header, but needs a database or redis. I'm always reluctant in adding more components which could fail or cause load. topoh also works, but it worries me a bit that according to the manual: https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask… a private IP is used to mask the contact header. I have come across a CPE which, as far as I see, is misbehaving by ALWAYS sending a PRACK to the Hostname or IP found in the Contact header, and ignoring Route and Via Header. So why using a private IP as 'mask' and not the IP or Hostname of the actual kamailio instance? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 6

Re: [sr-dev] Is there any Function or parameter to check Initial Invite Response on Kamailio
by Henning Westerholt 25 Jan '23

25 Jan '23

(removed sr-dev from CC) Hello, you can do that with the reply_route and/or onreply_route, for an example check out example cfg or this: https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_on_reply Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: HimaBindu G <himabindu.garadareddy(a)gmail.com> Sent: Tuesday, January 24, 2023 3:16 PM To: sr-users(a)lists.kamailio.org; sr-dev(a)lists.kamailio.org Subject: [sr-dev] Is there any Function or parameter to check Initial Invite Response on Kamailio Hi All, Can you please suggest any function or parameter to check Initial Invite Response from the backend. Thanks & Regards, Hima Bindu.

1 0

RTPengine multiple setID sockets
by John Hardiman 24 Jan '23

24 Jan '23

Hi All, Hope you are all well? I am hoping you can help with a bit of syntax I am struggling to get working in the intended way. (We are using Kamailio 5.5.3 and using Kemi/python.) The scenario is in AWS where we have separate Kamailio and RTPengine instances, with a cluster of FreeSWITCH EC2's on private addresses only. In the current setup, all RTP routes via the NAT gateway and back to the private IP of the selected FreeSWITCH (and vice versa). This works in the current config, but I would want this traffic not going via the NAT gateway and use the private address and route internally on the VPC. (These EC'2 have only 1 network interface and that cannot be changed for this work.) So anything between FreeSWITCH and Kamailio/RTPengine uses the private IP's for media, the rest use public IP's. It is good to note that the RTPengine's are the same in each set (3 total RTPengine's), just using the public/private IP's based on set ID. My first aim is to use KSR.permissions.allow_source_address() to check if the source address is from the FreeSWITCH cluster and set the rtpengine setID based on this. I have used this method for different parts of the config and it works. I am failing at this first step it seems, as no matter what I use in the function (below) it defaults and uses the public IP's in SDP. (No matter if I use the same setID in each set) kamailio.cfg snippet: # ----- rtpengine params ----- modparam("rtpengine", "setid_avp", "$avp()") # Public IP's ID 1: modparam("rtpengine", "rtpengine_sock", "1 == udp:5.5.5.5:2223 udp:6.6.6.6:2223 udp:7.7.7.7:2223") # Internal IP's ID 2: modparam("rtpengine", "rtpengine_sock", "2 == udp:10.0.0.1:2223 udp:10.0.0.2:2223 udp:10.0.0.3:2223") Kemi file snippet: # RTPengine Options for different SetID def ksr_route_rtpengine(self, msg): if KSR.textops.has_body_type("application/sdp") > 0 : # Source address is private subnet - use RTPengine set 2 (private IP's) if KSR.permissions.allow_source_address(400) == 1: KSR.rtpengine.set_rtpengine_set(2) KSR.rtpengine.rtpengine_manage( "RTP/AVP replace-session-connection replace-origin port-latching ICE=remove") else: # Source address is NOT private - use RTPengine set 1 (public) KSR.rtpengine.set_rtpengine_set(1) KSR.rtpengine.rtpengine_manage( "RTP/AVP replace-session-connection replace-origin port-latching ICE=remove") KSR.xlog.xlog("L_INFO", "RTPengine handling Block") return 1 Group 400 is the private subnet of the FreeSWITCH clusters we have behind on private addresses and the list contains the subnet of private IP's. Using the above, the Kamailio starts fine and understands/loads each set of RTPengine's and calls work with audio. but regardless of the set number used (even only using set 2 in both) it defaults to public IP's in the SDP. (This is the only place in the config where this function is defined) My SDP from FreeSWITCH egress to Kamailio/RTPengine has private IP's in the SDP. The only thing I do not fully understand is the setid_avp - specifically $avp() block and if this needs to stay empty, or be duplicated with the set ID in each one. Any help would be greatly appreciated in understanding where I am going wrong here. Thanks, John.

3 2

[SR-Users]: Port number in To: field
by Olivier 24 Jan '23

24 Jan '23

Hello, While trunking with an Oracle SBC, I was lately required to include a port number in INVITE To: field. Would you qualify this as a "legal requirement" (according to SIP Standards) ? Best regards

4 3

dialog: db_mode => 1, enable_dmq => 1 but entries not replicated to 2nd node DB
by Benoit Panizzon 24 Jan '23

24 Jan '23

Hi I am struggling with DMQ and dialog DB storage leading to orphan or duplicate entries in database and not loading all dialog information upon restart. usrloc on the other side, also using local db storage and dmq, works flawlessly even when restarting one node. All location information is replicated between the databases via DMQ. So trying to find what is different. usrloc module: modparam("usrloc", "db_url", DBLOCAL) modparam("usrloc", "db_mode", 2) modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) dialog module: modparam("dialog", "db_url", DBLOCAL ) modparam("dialog", "db_mode", 1 ) modparam("dialog", "db_skip_load", 0) modparam("dialog", "enable_dmq", 1) via kamcmd dlg.list I see that dialog information is being replicated between the two nodes. But I can only find the dialog in the local database of the node which is handling the call. Is this the expected behavior? I also just found how to create duplicate entries in the database... * Run a call on Node A, verify the dialog information is store in the local database and replicated to node B via DMQ. * Restart Kamailio on Node A. There I suspect, node A is restoring Dialog Information via DMQ from Node B and loading the same information from database, leaving duplicates in dialog_vars. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Kamailio World 2023 - Call For Presentations
by Daniel-Constantin Mierla 23 Jan '23

23 Jan '23

Hello, I would like to announce that Call for Presentations at Kamailio World 2023 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k2023/call-for-speakers/ The 11th edition of the conference returns to Berlin, Germany, during June 5-7, 2023, as an in-person event. Expect a large range of participants, developers and community members as well as representatives of 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.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 0

Performances issue when freeing shared memory in custom module (Kamailio 5.5.2)
by Chaigneau, Nicolas 23 Jan '23

23 Jan '23

Hello, I'm encountering performance issues with Kamailio (5.5.2). I'm using a custom Kamailio module that loads routing data in memory, using Kamailio shared memory. This routing data is very large. It can be fully reloaded through a Kamailio RPC command (which is done once each day). When reloading, two sets of data are maintained, one "loading" and another "current" (the latter being used to handle SIP requests). When loading of the new data is finished, it is swapped to "current". Then, memory of the old (now unused) data is freed. I've noticed that when Kamailio is freeing the old data, there is a very significant performance impact on SIP requests. This is surprising to me, because the SIP requests do not use this old data. This is not a CPU issue, idle CPU% is at about 99% at that moment. I'm using the following functions : - shm_mallocxz - shm_free From what I understand, shm_free is actually "qm_shm_free" defined in "src\core\mem\q_malloc.c" (the default shared memory manager being "qm"). I've noticed that there is also a variant shm_free_unsafe ("qm_free"), which does not perform locking. I'm wondering if the lock could be the cause of my performances issues ? (But I'm not sure how this could be possible, because although the SIP requests need to access the shared memory allocated, they do not use directly the functions from the share memory manager.) If the performances issues are causes by the lock, could I use the unsafe version "safely" ? (considering that it is guaranteed that the old data cannot be used by anyone else) Thanks for your help. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 6

dialog.dlg_filter_mode has ambiguous definition with double negation
by Дилян Палаузов 23 Jan '23

23 Jan '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/dialog#dialog.p.dlg_fil… says for dialog.dlg_filter_mode (int) Set dialog fitering mode, which can specify what dialogs are processed. Its value can be a combination (the sum) of following flags: 1 - do not send keepalives and do not execute timeout function if dialog is not local (if the associated bind address is not a local socket). Default value is “0”. As you can see, the definition of 1 is a double negation and there is no definition for 0. What means 0: - do not send keepalives and do not execute timeout function if dialog is local - send keepalives and execute timeout function if dialog is not local - send keepalives and execute timeout function if dialog is local

1 0

Does tm.t_relay() create a transaction?
by Дилян Палаузов 20 Jan '23

20 Jan '23

Hello, the documentation for tm.t_newtran() says „This is the only way a script can add a new transaction in an atomic way.“ Moreover tm.t_send_reply() “creates the transaction if it does not exist (executing internally t_newtran()) and sends a stateful reply (executing internally t_reply())”. This means that only t_newtrans() and t_send_reply() can create transactions in an atomic way. The documentation of tm.t_relay() does not say whether the function creates a transaction. It also does not say what happens if t_reply() is not called after t_newtrans(). But for tm.t_set_retr() and tm.t_set_retr() is written “If the transaction is already created (e.g called after t_relay() or in an onreply_route) …” This suggests that tm.t_relay() does create atomically or not-atomically a transaction. - What does it mean to create a transaction in a non-atomic way? - Does tm.t_relay() create a transaction? Greetings Dilyan

2 2

sslmode in db_postgres
by Centelles Martin, Joel 20 Jan '23

20 Jan '23

Hello everybody, I'm working on a kamailio instance as a basic registrar service and I thought to use db_postgres to connect a postgres service only allowing SSl connections. I see that that module doesn't provide any parameters to select sslmode and I was wondering if this module actually supports TLS connections. Does anybody know if this is supported and, if so, how to enable it? or should that support be added into db_postgres module code? Thank you very much in advance. BR, Joel Centelles RTC Lead Engineer | Care Communications R&D joel_centellesmartin(a)baxter.com<mailto:joel_centellesmartin@baxter.com> | baxter.com<https://baxter.com/> [cid:9ee6bcbd-9c30-45ec-bb43-8f439f19f724]<https://baxter.com>

1 0

[core/tcp_main.c:617]: _wbufq_add(): (1244714 bytes): write queue full or timeout
by Benoît Panizzon 20 Jan '23

20 Jan '23

Hi I'm trying to get dlg.list on a busy system via jsonrpc and xhttp. I already increased: tcp_conn_wq_max=2048 tcp_wq_max=20480 modparam("ctl", "binrpc_max_body_size", 2048) modparam("ctl", "binrpc_struct_max_body_size", 2048) But it looks like the tcp write queue overflows up at shy over 1MB reply no matter what I change. Any ideas what else to toggle? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 9

What are extended flags?
by Дилян Палаузов 20 Jan '23

20 Jan '23

Hello, The corex module manages extended flags with functions the setxflag(), isxflagset(), and resetxflag(). tmx.t_flush_xflags() does “flush the extended flags from the current SIP message into the already created transaction. It makes sense only in routing block if the transaction was created via t_newtran() and the extended flags have been altered since. It is not needed to execute this function when using t_relay() (or similar tm relay functions, xflags are synchronized automatically in that case).” The documentation of the tm module does not say anything about extended flags - in particular it does not say that t_relay() synchronizes them. Moreover, while the corex module utilizes the extended flags, the documentation of tm and tmx do not state, that they depend on the corex module. So, what are extended flags? Greetings Дилян

1 0

SIP Trunk with multiple phone numbers for account
by Carlos Escalona 19 Jan '23

19 Jan '23

Hello all! I have a question. In my system we use a series of VoIP providers to receive external calls to our telephone numbers, for the same provider we have several registered numbers and we need our providers to send that information in some way, so far it does not seem that there is a common way to do it (report the number that was dialed by the person who made the call). From what I understand, SIP does not have a place for that information because there are no things like phone numbers in the specification, my question is, is there a correct place to send such information, or is it entirely up to each provider? (Now, the most common is to have in the RURI and To header, the id of the trunk registered in the uacreg table) I have seen a couple of options that seem to me the most “correct”: send the information in the To header or send in a custom header. I really appreciate any help you can provide. Atenciosamente, [image: photo] <https://www.techer.com.br/> *Carlos Escalona* IT - Development 41 3073-0091 | R 1011 www.techer.com.br -- O conteúdo desta mensagem é confidencial. Se você não se encontra na lista de destinatários ou tenha recebido por engano, não a copie, imprima, envie, ou utilize, de qualquer forma, seu conteúdo. Neste caso, destrua a mensagem e, por favor, notifique o remetente.

3 3

Setting socket_workers does not work if using interface name
by Kaufman 19 Jan '23

19 Jan '23

At a quick glance it looks like using socket_workers when the listen parameter uses the interface name does not work. For example: children = 4 socket_workers=2 listen=udp:10.10.10.10:5060 Gives two listeners on udp:10.10.10.10:5060 children = 4 socket_workers=2 listen=udp:eth0:5060 Gives four listeners on the socket. Note this is just a quick cursory observation. Is it by design or a design limitation? Kaufman

3 3

RTC/Kamailio dinner at Fosdem 2023
by Daniel-Constantin Mierla 17 Jan '23

17 Jan '23

Hello, Fosdem 2023 returns to Brussels, Belgium, for a physical event and I'm writing to see if there are many people from the community going to the event and want to organize again the usual RTC/Kamailio dinner on Saturday evening (Feb 4, 2023). Each person should pay for himself, likely to be some price increase comparing with the last one. If there is enough interest, Torrey will help to book a place to accommodate us (first choice could be the same restaurant like in the past, if available). Reply if you want to join the dinner and say how many other people are joining you. Note that RTC Devroom got at this edition a half a day room allocation, on the afternoon of Feb 5, 2023 (Sunday). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

7 8

Re: rtp lost package in 300 concurrent calls and above
by Henning Westerholt 16 Jan '23

16 Jan '23

Hello, please keep the list in CC. Ok, the machine seems to not really using all its cores. Regarding playing a file, this can be also done with rtpengine: https://kamailio.org/docs/modules/5.5.x/modules/rtpengine.html#rtpengine.f.… Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com> Sent: Monday, January 16, 2023 3:35 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] rtp lost package in 300 concurrent calls and above thanks for your reply my kamailio server has 8GB of ram and 4core*2.6 on esxi(dedicated)Ubuntu 22.04.1 LTS. is it low? htop on server shows load average 1.04 at max and I didn't see more than 10% load on any core of the server. I've selected rtp media server because of playback of a voice file over incoming calls in the future usage On Mon, Jan 16, 2023 at 5:45 PM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, do you observe (too) high load on the system with the standard system management tools? If you want to just proxy calls, my recommendation would be to use rtpengine. Here you should be able to achieve higher concurrent calls rates, if you have a few CPUs on a decent machine. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Mohammad Reza Keshavarzianpoor <keshavarzianpoorm(a)gmail.com<mailto:keshavarzianpoorm@gmail.com>> Sent: Monday, January 16, 2023 2:50 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] rtp lost package in 300 concurrent calls and above Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards

4 10

rtp lost package in 300 concurrent calls and above
by Mohammad Reza Keshavarzianpoor 16 Jan '23

16 Jan '23

Hi, I've installed last version of kamailio over ubuntu. I need to broadcast machine calls to destinations. my concurrent call should be around 300(my pstn channels), so I start calls with cps around 20, because my calls contains a less than 30 second and if destination answer the call it will hanged up in less than or equal to 30 seconds. My problem is: to test my kamailio config, I'm using startrinity sip tester over an win server 2019 on local network. and it shows about 20% lost package on rtp and a jitter about 500ms. what should I do? my config is based on rtp media server module. Is there a complete working sample config for kamailio with rtp media server module? I tried rtpproxy and rtp engine in past and the same problem happened. Thanks for any help you can provide Best regards

4 5

Authenticated ACK: $au not set?
by Benoit Panizzon 16 Jan '23

16 Jan '23

Hi Gang I have this code snipplet: if (has_credentials("$fd")) { xlog("L_INFO", "$cfg(route): got $rm with credentials. Validate them!\n"); if ($aU == $null) { xlog("L_INFO", "$cfg(route): no auth user, send challenge\n"); } auth_challenge("$fd", "0"); exit; } } Please don't ask if this makes sense, it's a constructed snipplet to show the situation. If $rm is INVITE, this works fine, no challenge is being sent, $au is populated. If $rm is ACK (some CPE seem to send the Credentials with each message) then $au is $null, auth_challenge is called and fails. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Parallel call forking with call authorization
by Mathias Schneuwly 13 Jan '23

13 Jan '23

Hi As a first notice; I already wrote to the mailing list, but without registration. I read afterwards that I should register first because of spam reasons. If this is a duplicate, then I'm sorry. I'm new to mailing lists... I've an issue with parallel call forking together with call authentication. I hope somebody can give me some hints about what the problem could be and how I could solve this problem. I'm using Kamailio 5.5.4 with the python KEMI binding. Kamailio is in front of an Asterisk, which requires call authorization. RTPEngine is also inplace which does SRTP to RTP bridging. Simple calls from and to Asterisk through Kamailio and RTPEngine work without any issue. So the basic setup seems to work fine. The problem starts when I do parallel call forking where both calls are sent to Asterisk. If I use "KSR.tm.t_on_failure("trunk_authentication")" for authentication, the first branch is authenticating and the device rings, but the second branch does not authenticate. Well, that's not 100% true, if the SIP timeout appears on the first branch, I can see in the log, that the second branch enters the trunk_authentication method, but then, the call is already terminated. Call forking to two devices (not behind Asterisk and without call authentication) works also as expected. I made some trials with "KSR.tm.t_on_branch_failure("trunk_authentication")" but in this case, the behavior is strange. Then the authentication for both branches work and both devices are ringing. If a device hooks up, the second device does not stop ringing. And after a few seconds, the first device cancels the call, because it did not get the ACKs. I can also see that the CANCEL is sent from Kamailio to Asterisk, but Asterisk does not react to it. For trunk authentication, I'm using the implementation from the well known examples plus a few own stuff. Btw, NAT is not enabled...: def trunk_authentication(self, msg): scode = KSR.pv.get("$T_reply_code") self.log('notice', 'trunk_authentication {}'.format(scode)) if WITH_NAT and self.ksr_route_natmanage(msg) == -255: return 1 # Call manage_rtp_engine to destroy RTPEngine if it has been used. call_context = self.get_call_context(msg=msg) callee_object = call_context.callee.get_callee_object() if callee_object.connection_type: call_context.manage_rtp_engine() if KSR.tm.t_is_canceled() > 0: return 1 if scode in [401, 407]: KSR.pv.sets('$avp(auser)', KSR.pv.get('$sel(cfg_get.asterisk.trunk_username)')) KSR.pv.sets('$avp(apass)', KSR.pv.get('$sel(cfg_get.asterisk.trunk_password)')) if KSR.uac.uac_auth(): # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if KSR.is_method_in('IBSU'): if KSR.tm.t_is_set("branch_route") < 0: KSR.tm.t_on_branch("ksr_branch_manage") self.log('notice', 'Authentication Asterisk trunk') if KSR.tm.t_relay() < 0: KSR.sl.sl_reply_error() return 1 Any ideas? Best regards Mathias

3 6

src/modules.lst - difference between skipping a module from compile list and excluding a module from compile list
by Дилян Палаузов 13 Jan '23

13 Jan '23

Hello, the generated src/modules.lst file contains: # the list of modules to skip from compile list skip_modules= … # the list of modules to exclude from compile list exclude_modules= … For me excluding a module from the compile list and skipping the module when compiling is the same. Please describe what is the difference between skipping and excluding, either in modules.lst, at https://www.kamailio.org/wikidocs/devel/makefile-system/ or at https://kamailio.org/docs/tutorials/devel/kamailio-install-guide-git/ . Thanks Дилян

2 1

Newbie question
by Joao Paulo Vaz 12 Jan '23

12 Jan '23

Hello masters, I am a newbie who recently got to know Kamailio. I have a scenario and I would like to know if this tool would suit: a central asterisk server with all user accounts and three remote units with Kamailio acting as a proxy. Is there any module or configuration that if the central asterisk server fails to respond, would the remote proxies act in survival mode, allowing local communication by users until the central asterisk server returns a response? Att, Joao Paulo

3 4

unsubscribe.pls
by Stanly Levyy 11 Jan '23

11 Jan '23

*** THIS E-MAIL MESSAGE AND ANY DOCUMENTS ATTACHED IS/ARE INTENDED ONLY FOR THE USE OF THE INDIVIDUAL(S) OR ENTITY TO WHICH IT IS ADDRESSED; AND MAY CONTAIN INFORMATION THAT IS LEGALLY PRIVILEGED, CONFIDENTIAL AND/OR EXEMPT FROM DISCLOSURE UNDER APPLICABLE LAW, including the Electronic Communications Privacy Act, 18 U.S.C. Secs 2510-2521 *** On Wednesday, January 11, 2023, 01:36:44 AM PST, <sr-users-request(a)lists.kamailio.org> wrote: Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Performing async_http calls on stateless Loadbalancers (Julius Flohr) 2. Re: Releasing v5.6.3 (Daniel-Constantin Mierla) 3. Performing async_http calls on stateless Load balancers (Lewis Hutchinson) ---------------------------------------------------------------------- Message: 1 Date: Tue, 10 Jan 2023 18:50:44 +0100 From: Julius Flohr <flohr(a)sipgate.de> Subject: [SR-Users] Re: Performing async_http calls on stateless Loadbalancers To: Henning Westerholt <hw(a)gilawa.com> Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Alles wird besser <alleswirdbesser(a)sipgate.de> Message-ID: <34C7487C-0900-4C49-BBB4-49A0291D83E6(a)sipgate.de> Content-Type: multipart/alternative; boundary=Apple-Mail-0AF7C672-1C85-4A98-9DDC-EED95FFB4B7E A message part incompatible with plain text digests has been removed ... Name: not available Type: text/html Size: 7117 bytes Desc: not available ------------------------------ Message: 2 Date: Wed, 11 Jan 2023 08:30:17 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: [SR-Users] Re: Releasing v5.6.3 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, "Kamailio (SER) - Devel Mailing List" <sr-dev(a)lists.kamailio.org> Message-ID: <454ae3b5-a810-f393-4196-e71c4482e9d3(a)gmail.com> Content-Type: text/plain; charset=UTF-8 Hello, the usual note that preparing the release of Kamailio v5.6.3 starts soon, therefore any commits to branch 5.6 should be announced first on sr-dev mailing list and wait for the ack before pushing to repo. Once the announcement is out, commits can go in the branch directly. Cheers, Daniel On 06.01.23 16:44, Daniel-Constantin Mierla wrote: > Hello, > > I am considering to release Kamailio v5.6.3 (out of branch 5.6) next > week (likely on Wednesday, Jan 11, 2023). If anyone is aware of > issues not yet on the bug tracker, report them there asap in order to > have a better chance to be fixed. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com ------------------------------ Message: 3 Date: Wed, 11 Jan 2023 09:20:28 +0000 From: Lewis Hutchinson <lewis.hutchinson(a)missionlabs.co.uk> Subject: [SR-Users] Performing async_http calls on stateless Load balancers To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Cc: "flohr(a)sipgate.de" <flohr(a)sipgate.de>, "alleswirdbesser(a)sipgate.de" <alleswirdbesser(a)sipgate.de> Message-ID: <CWXP265MB2517CB6E3E8188F3E3BB5B2AC5FC9(a)CWXP265MB2517.GBRP 265.PROD.OUTLOOK.COM> Content-Type: text/plain; charset="us-ascii" Hi Julius If I have understood this correctly, I think you can achieve this using KEMI Python leveraging aiohttp & asyncio i.e async with aiohttp.ClientSession() as session: This can be done outside of REQUEST_ROUTE Requires a rewrite from Native to KEMI so might be a bit excessive but would work. Regards Lewis ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list -- sr-users(a)lists.kamailio.org To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org ------------------------------ End of sr-users Digest, Vol 212, Issue 33 *****************************************

1 0

Performing async_http calls on stateless Load balancers
by Lewis Hutchinson 11 Jan '23

11 Jan '23

Hi Julius If I have understood this correctly, I think you can achieve this using KEMI Python leveraging aiohttp & asyncio i.e async with aiohttp.ClientSession() as session: This can be done outside of REQUEST_ROUTE Requires a rewrite from Native to KEMI so might be a bit excessive but would work. Regards Lewis

1 0

Releasing v5.6.3
by Daniel-Constantin Mierla 11 Jan '23

11 Jan '23

Hello, I am considering to release Kamailio v5.6.3 (out of branch 5.6) next week (likely on Wednesday, Jan 11, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 1

Help with rtpengine_delete()
by Jose Figueroa 10 Jan '23

10 Jan '23

Hello everyone! I've been struggling with a kamailio scenario with the rtpengine_delete. My scenario is the following: I've setup a Kamailio multihomed, since some of my vendors has provided us a direct link using a different local subnet in eth2 and I have my public IP in eth1. The kamailio i've been working on is using app_lua module. SIP A <---------> 200.XXX.XXX.14 (eth1) [Kamailio 5.6] (eth2) 10.203.5.8 <------> Vendor 10.203.5.20 100.XXX.XXX.22 Basically, I receive a call from public IP eth1 and according to the logic from the vendor I forward statefully the call to 10.203.5.20 in second branch. I've been reading a lot of issues and documentation such as:[#840 rptengine], [#875 rtpengine] and [#2222 kamailio], but I don't really get working the rtpengine_delete() correctly, since I go to netstat -plan|grep rtpengine and I see some sessions still that are not freed. My second case is the following when I receive a 503 from another GW: SIP A <---------> 200.XXX.XXX.14 (eth1) [Kamailio 5.6] (eth1) 200.XXX.XXX.14 <-----------> Vendor GW 233.XXX.XXX.19 100.XXX.XXX.22 In that case I still see the rtpengine ports, mostly with my public IP 200.XXX.XXX.14 but some of them with the LAN 10.203.5.8 IP. Also, something curious I see on my 2nd use case is the following: https://pastebin.com/raw/B4vCgExX So, it would be great if someone can point me to the right direction, how and when I should use rtpengine_delete in order to finish the rtpengine sessions correctly and the ports can be freed up. Any help will be great! [840] https://github.com/sipwise/rtpengine/issues/840 [875] https://github.com/sipwise/rtpengine/issues/875 [2222] https://github.com/kamailio/kamailio/issues/2222 -- Regards, Jose Figueroa

3 2

Why do htable Keys have two colons?
by Noah Mehl 10 Jan '23

10 Jan '23

Hey everyone, Whenever I read documentation for htable usage, and other examples of other scripts in Kamailio, htable keys are typically named: something::something Examples are: $au::auth_count, $ci::srcip, join::$rU Why are there two colons (as opposed to a | )? Is this just a standard semantic thing in Kamailio? Is there something about that double character that’s unusual and easy to parse with SIP values? Or is it just what people do, and that’s a good choice to continue for all other Kamailio script writers to have understanding? Any insight would be appreciated. Thanks! ~Noah

3 6

Kamailio Crypto module and 3rd party openssl
by Arsenijs Kabihno 10 Jan '23

10 Jan '23

Hello, I am trying to make use of crypto module. The scenario is the following: 1. Encrypt plaintext passwords on kamailio side using crypto_aes_encrypt function of crypto module 2. Decrypt them outside of kamailio using 3rd party tools, such as PHP openssl_decrypt Crypto module parametrs are the following: modparam("crypto", "salt", "") modparam("crypto", "key_derivation", 0) If encryption and decription are handled by kamailio, the initial password matches the decrypted string (initial password -> encrypted string -> decrypted string) For example: sql_query("ds_dburl", "select secret from kamailio_sip_buddies where username='$au'", "secret"); $avp(secret_plain) = $dbr(secret=>[0,0]); //value: 'kamailio' crypto_aes_encrypt("$avp(secret_plain)", "YUZySHVtdlVTYjI0TGJNd3JTeVV0MlRaRWFraFNBRlM=", "$var(encrypted)"); //value: 'qfOqTMoJMgGAherGCqLRRQc4zTlqsEj3MEtcORurFf0=' crypto_aes_decrypt("$avp(secret_plain)", "YUZySHVtdlVTYjI0TGJNd3JTeVV0MlRaRWFraFNBRlM=", "$var(decrypted)"); //value: 'kamailio' But in case I try to decrypt the value of $var(encrypted) using PHP openssl_decrypt or online tool like https://www.devglan.com/online-tools/aes-encryption-decryption, I am getting the result 'kamailio' prepended by 16 random bytes, like: 6“¤›¿Ð+IÊ\ñžïZkamailio or 6��+I\��Zkamailio I believe this 16 bytes are actually random initialization vector in some encoding. Possibly kamailio is trimming them from result here: https://github.com/kamailio/kamailio/blob/master/src/modules/crypto/crypto_… Moreover, if secret was encrypted using crypto_aes_encrypt and random IV, you can use any IV while decrypting and get 16 different bytes+result Please advice if someone have used crypto module in similar way and what are the possible ways to acheive results consistency

2 5

pkg memory leak when acc module cdr_enabled
by Juha Heinanen 10 Jan '23

10 Jan '23

In latest stable K release, we noticed pkg memory leak (pgk memory usage increases by each processed call). It turned out that the leak goes away if acc module cdr_enable is not enabled. Could be a bug in dialog or acc module. Any debug instructions if the bug is not obvious? -- Juha

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