sr-users May 2017

sr-users@lists.kamailio.org

91 participants
117 discussions

Siremis upgrade and features
by javier falbo 21 May '17

21 May '17

Hi, I notice Siremis is not usable for me as it is very limited in the user management interface. I will suggest the developpers to include a USER management GUI visual interfase, as nowadays i use command line to do the following by putty tool: a) "Create Users" kamctl add <user> <password> b) "Delete Users" kamctl rm <user> c) "Show users in database" kamctl db show subscriber d) "Show online users" kamctl online Adding this on Siremis will help a lot for daily use as a Manager Account. Someone knows the Siremis author to add this suggestion? or someone has already coded a VISUAL GUI interfase for the above and could share it? Regards, Javier

4 7

Siremis searching from htable
by Olli Attila 21 May '17

21 May '17

Hello, Is there any way to make partial search from all htable entries with for example: %partofkeyname%? I tried to put that on search box but I get no results. If I use the exact full name of the key in the search i get the result. The search is here: Sip admin services - > server services -> htable list -> search I would like that the search function could return all of the keys that match the searched pattern but I have not succeed in making that kind of search. Cheers, Olli

2 2

How to debug Kamailio
by Remsik,Robert 19 May '17

19 May '17

Hello! I've got a base install of Kamailio and I'm trying to have it forward packets generated by sipsak to a different proxy (a migration of an existing ruleset by opensips). I can get the Kamailio process started but I can't figure out why Kamailio isn't forwarding the packet and I haven't been able to get debugging working to determine what's going on inside Kamailio to fix it. Can anyone please help me figure out how to debug this? Thank you in advance, Robert Robert Remsik ACNS Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

3 4

Repeat of packets
by Jean Cérien 18 May '17

18 May '17

Hello I am getting started with Kamailio (or restarted, used it briefly years ago), with the final objective to do load balancing. For the time being, I am just trying to have one asterisk and one kamailio, on the same box. I have setup a box with an asterisk 11.3, and kamailio 4.4. I've taken the config file from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . My idea is that asterisk runs on port 5080, while kamailio on port 5060. Client interacts with Kamailio on port 5060. It almost works... Registration is fine, but when I send an invite, it is properly acknowledged (by asterisk 100 trying then 200 OK) - but the OK message gets repeated multiple times and asterisk issues its infamous 'Retransmission timeout reached ...' - as if Kamailio wasnt processing it. See below ngrep traces between asterisk and kamailio Any ideas where to look ? Thanks J. # U +18.289105 192.168.2.228:5060 -> 192.168.2.228:5080 INVITE sip:102@192.168.2.228 SIP/2.0..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922 4198ef593.0..Via: SIP/2.0/UDP 192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102@ 192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Contact: <sip:iper@(null)>..Content-Type: application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY , MESSAGE, SUBSCRIBE, INFO..Max-Forwards: 69..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Subject: Phone call..Content-Length: 437....v=0..o=199 2799 2990 IN IP4 192. 168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0 0..m=audio 7078 RTP/AVP 124 111 110 0 8 101..a=rtpmap:124 opus/48000..a=fmtp:124 useinbandfec=1; usedtx=1..a=rtpmap:111 s peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110 speex/8000..a=fmtp:110 vbr=on..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..m=video 9078 RTP/AVP 103 99..a=rtpmap:103 VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99 profile-level-id=3.. # U +0.000683 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as57c98c4b..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.001035 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as30d9cfb4..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.000407 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;rec eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip :102@192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H..Supported: replaces, timer..Contact: <sip:102@192.168.2.228:5080>..Content-Length: 0.... # U +0.003961 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.087859 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as57c98c4b..To: < sip:199@192.168.2.228:506 0>;tag=524348182..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.000213 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as30d9cfb4..To: < sip:199@192.168.2.228:506 0>;tag=939659485..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.011138 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.200291 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.400478 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. #

3 5

IMS: Including contents in the body of third-party REGIISTER request.
by Haggerty, Paul (phaggert) 18 May '17

18 May '17

It looks like the Kamailio IMS modules don't support including the original REGISTER request in third party registrations as defined in 3GPP TS 24.229. Can someone confirm this? Should this be a tracked defect or enhancement? In the meantime, is there any workaround available? Our application server needs to get the contents from the original REGISTER message from the UE. Thanks, Paul

2 4

Publish/notify for group kamailio
by Nguyen Tran Nhan 18 May '17

18 May '17

Hi, Is there any way to send Publish/Subscribe to group and Notify for all group (conference)? I have tried to send Subscribe to conference URI (SEMS) but I got method not implemented error. Any idea? Thanks, Nhan

1 0

kamcmd cli autocomplete fetaure stopped working after update from 4.3.4 to 4, 4.4 and 4.4.5
by Nuno Reis 18 May '17

18 May '17

Hi guys. I've just noticed that I can't use autocomplete under kamcmd cli anymore, after upgrading from 4.3.4 to latest versions. Should I open a github issue for this? -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/>

2 5

no audio with kamailio & rtpproxy bebind NAT
by 赵国杰 18 May '17

18 May '17

Hi everyone, I'm new to kamailio and rtpproxy and the problem's been bothering me for days. Any help would highly appreciated. My network topology is as follows: kamailio5.0 and rtpproxy2.0 are running on the same server. The server is behind NAT, its private ip is 10.240.0.62, public ip is <PUBLICIP> the server has 5060 tcp&udp, 35000-65000 udp port enabled. There're 2 sip clients(test && test1) running in another LAN(175.100.202.xxx) . Situation: Both clients registered successfully and they can call each other. But neither of them has audio.(If i kill rtpproxy, audio is ok). My kamailio cfg(only posted the part that i change), #!KAMAILIO #!define WITH_MYSQL #!define WITH_DEBUG #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PSTN listen=udp:10.240.0.62:5060 advertise <PUBLICIP>:5060 modparam("rtpproxy", "rtpproxy_sock", "udp:10.240.0.62:64000") rtpproxy cmd: rtpproxy -A <PUBLICIP> -F -l 10.240.0.62 -m 35000 -M 65000 -s udp:*:64000 -d INFO I used ngrep -d any -qt -W byline port 5060 to capture the sip msg, they are as follows: U 2017/05/18 05:56:29.320152 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.321103 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---9e44840a531df765;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="<PUBLICIP>", nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8". Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.384015 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 ACK. Content-Length: 0. . U 2017/05/18 05:56:29.388008 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Proxy-Authorization: Digest username="test",realm="<PUBLICIP>",nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8",uri="sip:test1@<PUBLICIP>",response="e7c42929b9a62de980651327b3a6999c",algorithm=MD5. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.389271 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.389935 10.240.0.62:5060 -> 175.100.202.254:53006 INVITE sip:test1@175.100.202.254:53006;ob SIP/2.0. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 349. . v=0. o=- 1500767650 1 IN IP4 <PUBLICIP>. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 46862 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=nortpproxy:yes. U 2017/05/18 05:56:29.782323 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>. CSeq: 2 INVITE. Content-Length: 0. . U 2017/05/18 05:56:29.846192 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:29.846709 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:32.768386 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 233. . v=0. o=- 3704075788 3704075789 IN IP4 10.60.208.171. s=pjmedia. c=IN IP4 10.60.208.171. t=0 0. m=audio 4008 RTP/AVP 0 101. c=IN IP4 10.60.208.171. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2017/05/18 05:56:32.769284 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 200 OK. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 255. . v=0. o=- 3704075788 3704075789 IN IP4 <PUBLICIP>. s=pjmedia. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 42584 RTP/AVP 0 101. c=IN IP4 <PUBLICIP>. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=nortpproxy:yes. U 2017/05/18 05:56:32.836731 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1 SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 70. Route: <sip:<PUBLICIP>;lr;nat=yes>. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:32.837266 10.240.0.62:5060 -> 175.100.202.254:53006 ACK sip:test1@175.100.202.254:53006;ob SIP/2.0. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.c283a6093e5f4143adec82663a0ff1d1.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:33.025120 10.240.0.62:5060 -> 175.100.202.254:53006 .... U 2017/05/18 05:56:38.025707 10.240.0.62:5060 -> 175.100.202.33:1490 .... U 2017/05/18 05:56:38.025731 10.240.0.62:5060 -> 175.100.202.33:10870 .... -- cheers, Jesse

1 0

Kamailio 4.4.X bug fixes
by José Seabra 17 May '17

17 May '17

Hello there, I apology for the following question but i didn't find this information on the kamailio site. When the development cycle in terms of bug fixes ends on version 4.4.x? Thank you Regards José

2 2

PV Setting Question
by Duarte Rocha 17 May '17

17 May '17

Good Afternoon, I'm having trouble with creation and value setting of some pseudo-variables. In the first cases, I used the pvapic.c function pv_get_sintval() like this : pv_get_sintval(msg, param, res, number_int_value); Using pv_get_sintval() i can assign the value to the PV and print it in the configuration file without any problem. However, if i need to assign a char* to the PV and use pv_get_strzval(msg, param, res, number_char_value);, it doesn't reach the configuration file. I made some debugs inside pv_get_strzval() in order to see the values of "res->rs.s" and res->rs.len and both values are fine. I just can't access them in the configuration file. Thanks in advance for your help! Cheers, Duarte Rocha

2 1

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sr-users May 2017