- sr-users - kamailio.org

Normal behaviour with loose_routing
by Loic Chabert 30 Jan '18

30 Jan '18

Hi list, I have a setup configurered has diagram below: UAC -> Proxy LB (proxy lb) -> Proxy Router/registrar (proxy B) -> Asterisk SBC My UAC is SIPP and sending a classic call scenario: INVITE - ACK - BYE. But i have some trouble to understand which queries should be loose routed and which queries should be dispatched by dispatcher module. My first invite request has been dispatched to proxy B by proxy LB. After that, ACK and BYE request should be loose_routed or should be … [View More]

3 5

rtjson + uac + event - OK, ACK and BYE display name not updated
by Diego Nadares 30 Jan '18

30 Jan '18

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

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Session Timers
by Duarte Rocha 30 Jan '18

30 Jan '18

Hey guys, I have two issues about Session Timers that need some clarification. 1) I have my Kamailio between my clients CPE's and my GW to the PSTN. I want to configure it in a way that i always try to make the CPE do the refreshing so I can save resources on my kamailio and in the GW. If the CPE doesn't support it, the refresh should be done by the GW. Is it possible to do this test and configuration in every call ? 2) SIP has the MIN-SE mechanic and is well documented, but is it possible … [View More]

2 1

Re: [SR-Users] Problem to install Kameilio from Ubuntu packages
by Victor Seva 30 Jan '18

30 Jan '18

2018-01-26 17:46 GMT+01:00 Alain Bonnefoy - XOOL <a.bonnefoy(a)xool.fr>: > > > I tried to install Kameilio on Ubuntu 16.04 from what explained in the > wiki <https://www.kamailio.org/wiki/packages/debs>. > Wiki updated. Thanks

1 0

Re: [SR-Users] Problem to install Kameilio from Ubuntu packages
by Bastian Triller 30 Jan '18

30 Jan '18

There another command on http://deb.kamailio.org for people behind firewalls which block port 11371: % wget -O- http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - On Jan 26, 2018 5:46 PM, "Alain Bonnefoy - XOOL" <a.bonnefoy(a)xool.fr> wrote: > Hello, > > > > I tried to install Kameilio on Ubuntu 16.04 from what explained in the > wiki <https://www.kamailio.org/wiki/packages/debs>. > > > > But when I execute the command to add your GPG … [View More]

1 0

kamailio tls passthrough
by pavel＠eremina.net 30 Jan '18

30 Jan '18

Hi, All i am trying to load balancing Kamailio with TLS in passthrough mode. is it possible to do it in stateless mode?

2 1

whats happened to iptel.org
by voipspace voipspace 29 Jan '18

29 Jan '18

Anyone know whats happened to iptel.org The webiste is now a wix page iptel.org the sip uri music(a)iptel.org was a useful resource. Now the song has changed and the service is down alot Has it it changed hands or been sold?

1 0

i can not receive BYE after websocket has been closed and reconnected during the call
by Kenan Kocaerkek 29 Jan '18

29 Jan '18

Hi, I have a WebRTC softphone based on JsSIP and kamailio 4.4.7. Generally it works well but sometimes websocket connections interrupt and reconnect immediately (especially running behind proxies). If the interruption occurs during an active call, kamailio tries to send BYE message to softphone over closed websocket. gruu_enabled flag is active on registrar module and JsSIP sends proper sip.instance value. So after ws reconnects, registered contact details do not change on kamailio except … [View More]

1 0

Ann: Registration to Kamailio World 2018 is open!
by Daniel-Constantin Mierla 29 Jan '18

29 Jan '18

Hello, a quick note to the community forums to announce that the registration to next Kamailio World Conference (May 14-17, 2018, in Berlin, Germany) is now open! More details are available at: * https://www.kamailioworld.com/k06/registration/ In a matter of several days, we expect to publish the details about the first group of accepted speakers. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.… [View More]

1 0

New module – embedded MSRP relay
by kossivi HOLONOU 28 Jan '18

28 Jan '18

Hello,I would like to develop a project that will allow me to: - voice, video, screen sharing, etc. sessions with content communication via RTP - end to end presence – this is purely SIP routing - SIMPLE-based presence (aka, presence server or presence agent model) via presence* and pua* modules — user presence, dialog states notification (aka, blinking lamps), resource lists service (including OMA/RCS extensions), user location states notification and replication, audio/video … [View More]

1 0

Kamailio with asterisk on private lan
by Mark Boyce 28 Jan '18

28 Jan '18

Hi all I’m trying to create a relatively simple setup with Kamailio dual homed on public/private ip and asterisk on private ip only. The idea is load balance / fail over asterisk boxes. Following the real-time tutorial I have clients registering with Kamailio, Kamailio registering on clients behalf with asterisk as well as invites going through. However what I’m seeing is that when an invite occurs asterisk offers media on its private ip, as it would. However this is making its way through … [View More]

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SEMS with sbc application
by Mojtaba 27 Jan '18

27 Jan '18

Hello, I have a problem today, It's strange for me. Suppose we have this senario: uac1------->SEMS(mo profile)------->Kamailio-------->SEMS(mt profile)---------->uac2 In above topology, we have two interfaces(intern,extern) for SEMS, and just used as SBC (sbc application). if i used port=5060 as external port, every things is right and log file is like this: [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd msg via UDP from 89.165.117.125:42411 vv --++-- … [View More]

3 3

Transcoding
by Wilkins, Steve 26 Jan '18

26 Jan '18

Hello All, I am currently using Kamailio and Asterisk on Centos 7 servers and trying to enable WebRTC jsSIP clients to be able to do Audio/Video calls with Provider Phones (Purple, Z, Sorenson, etc....), however, the providers do not have vp8 codecs (which is what the WebRTC clients use for Audio) so I believe I will need a media proxy server to resolve the video issues. My question is, can rtpproxy or rtpengine perform this transcoding? If so, and if rtpengine is the way to go, should I use … [View More]

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Problem to install Kameilio from Ubuntu packages
by Alain Bonnefoy - XOOL 26 Jan '18

26 Jan '18

Hello, I tried to install Kameilio on Ubuntu 16.04 from what explained in the wiki. But when I execute the command to add your GPG key, I just get replied that the timeout occured. What to do ? Alain Provenance : Courrier pour Windows 10

1 0

Kamailio log file empty
by Ashutosh chaubey 26 Jan '18

26 Jan '18

Hi, My kamailio.log file is empty, following is my configuration: In Kamailio.cfg- log_facility=LOG_LOCAL0 #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=3 log_stderror=no #!endif Inside rsyslog.conf # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log User has all the privileges to write in … [View More]

3 2

EOL of geoip module.
by Sergey Okhapkin 26 Jan '18

26 Jan '18

Maxmind just announced EOL of legacy geolite databases used by geoip module, no updates starting this April. See more at https://blog.maxmind.com/ 2018/01/02/discontinuation-of-the-geolite-legacy-databases/ If you use kamailio's geoip module, I suggest to switch to geoip2 module which uses Geolite2 databases and is a plug in replacement for geoip module.

2 1

Behavior of UAC-Redirect module
by Richard Skidmore 26 Jan '18

26 Jan '18

2 2

SIP reply reason and code in dialog
by voipspace voipspace 26 Jan '18

26 Jan '18

within event_route[dialog:end] How do I find out SIP reply reason phrase and SIP reply code $rr $rs and $T_rpl($rs) $T_rpl($rr) are null

3 4

Three core dump bug fixes IMS kamailio 5.1.0 -> Pull Requests will follow
by Valentin Christoph 25 Jan '18

25 Jan '18

Hi all, When switching from kamailio 4.4.5 to kamailio 5.1.0 with the IMS installation in our lab, we detected three bugs, that lead to core dump. We will issue three pull requests in the course of this afternoon. KR ____ Christoph VALENTIN Software Developer | R&D Core / MGW + HLR + MSC P +43 50 811 3785 | M +43 664 628 3785 christoph.valentin(a)kapsch.net<mailto:christoph.valentin@kapsch.net> Kapsch CarrierCom AG | Lehrbachgasse 11 | 1120 Vienna | Austria Company Register at: … [View More]

2 2

Modify Reply Created
by Duarte Rocha 25 Jan '18

25 Jan '18

Hey everyone. Right now i'm doing a redirect server and i'm trying to sinalize the redirection using a 3xx reply. In order to do that, according the RFC I need to add a Contact and Diversion header. How do i do that ? i'm creating the reply with send_reply() but i can't seem to find a way to add these parameters to the reply. Thanks for the help. Cheers

3 2

kamctl domain reload
by David Villasmil 25 Jan '18

25 Jan '18

Hello guys, This is weird, using kamctl I am able to do things from like: kamctl trusted show +----+--------------+-------+--------------+--------------+------+----------+ | id | src_ip | proto | from_pattern | ruri_pattern | tag | priority | +----+--------------+-------+--------------+--------------+------+----------+ | 1 | 1.2.3.4 | udp | | NULL | | 0 | +----+--------------+-------+--------------+--------------+------+----------+ but … [View More]

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Re: [SR-Users] Modify Reply Created
by Duarte Rocha 25 Jan '18

25 Jan '18

Thanks for the reply, that works :) What about if I want to delete or overwrite an header ? I need to change to To Header. If i use append_reply() will it overwrite ? Cheers

1 0

Allow Methods and Allow events SIP Headers are missing on SIP Registrtaion
by Suresh Talasaniya 25 Jan '18

25 Jan '18

Hi All, I am facing issue while SIP REGISTER. Allow methods and Allow Event SIP Headers are the mission of 401 Unauthorized and 200 OK on Successful SIP Registered. Please find the attached document in which I mention both working and nonworking SIP Message. It would be appreciable if anyone help figures out this issue on Kamailio server. Thank you. -- Regards, Suresh Talasaniya. Contact : +91-9724264776 Skype : suresh.talsaniya

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Purge htable after dialog termination
by Jean Cérien 24 Jan '18

24 Jan '18

Hello I am following up on the improper sip dialog that has been fixed by storing contact details into a htable. I am trying to delete the key at the end of the call, so the table doesnt grow exponentially. I've added a event_route[dialog:end] {} block, but I see it is processed BEFORE the BYE is processed in the request_route(). Therefore, when I delete the data in the htable, I cant relay this BYE correctly. Is there another event (or technique) I could use to purge the htable ? doing … [View More]

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Question about [SIP Transaction] Handle Cancel message
by Jacek S 23 Jan '18

23 Jan '18

Hi How you finally solved the problem with "481 Call transaction does not exist"? I have the same problem. Best regards, Jacek Szyszka

1 0

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