sr-users April 2014

sr-users@lists.kamailio.org

100 participants
170 discussions

One-way Relay through RTPproxy
by Nandini madhu 07 Apr '14

07 Apr '14

Dear All, I am running Kamailio (4.0) and RTPproxy server (1.2.1) server in my set-up. When i call from Client A to B , the RTP packets are proxying trhrough RTPproxy service to other end client, but no flow from client B to client A. With this calls Are unsuccessful. I traced a SIP capture (please find in the attachment) and analysed it, as in there only one client 's (A) IP address is changed to RTPproxy IP address in SDP body. What can be do to resolve this issue ? Please anybody help me And also find the attachment for my kamailio config file, and suggest me or point me about any misplacement of NAT traversal function call in the script. Please consider this and help me out as i am newbie in scripting. Regards, Nandini

2 2

Show Domain Name in Siremis
by aawaise 07 Apr '14

07 Apr '14

Hello, I have a Kamailio-3.1.1 running with its DB on MySQL running on a seperate machine. Siremis is installed for interaction with DB. In web interface of Siremis, domain name of users is not shown. How can I show that ?? Like my domain name is "defg" and user is "abc". It shows online users as "abc@HostNameOfUser" instead of "abc(a)defg.com". Secondly does Siremis read information from Kamailio server or directly from DB server ?? Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp12653… Sent from the Users mailing list archive at Nabble.com.

1 0

Restricting Kamailio to single login
by MrIhaveAnOpinionOnEverything 05 Apr '14

05 Apr '14

Hi Everyone: I am a newbie in kamailio. Basically I am looking for a way to restrict the registration or login in kamailio per user to just one. The experience I want to setup is to unregister an older register/login to kamailio by a new login/register after the register command has already passed the authentication leg. Is there an example for this? Thanks. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com

3 2

Kamailio Freeswitch
by MrIhaveAnOpinionOnEverything 05 Apr '14

05 Apr '14

To whom it may concern: We are configuring a SIP platform with Kamailio and Freeswitch with this setup: UAC 1 ==> Kamailio ==> Freeswitch UAC 2 <== Kamailio <== Freeswitch We followed the instructions in http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc We are using Kamailio on kamailio 4.0.3 and Freeswitch FreeSWITCH Version 1.5.6b Everything is working fine with the default setting. But when we configured TLS what we noticed some errors. UAC configured with no TLS calling a UAC configured with TLS works ok. UAC configured with TLS calling a UAC configured with TLS gets disconnected 15-30 seconds after answering the call. UAC configured with TLS calling a UAC configured with no TLS gets disconnected 15-30 seconds after answering the call. I cant seem to find any error in the logs or in the ngrep. Can anyone assist me regarding this issue. Good day. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com

2 3

Problem on dialog and reInvite
by pars3c 05 Apr '14

05 Apr '14

Hi I've a problem of this type: On the first INVITE is all ok, insert record route with cookie of the dialog: Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.1661> The I recieive a 422 for the session timer too small . I resend the new invite with correct expire but with a new cookie dialog value Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.2661> Then the call is connected (Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.2661>). The there is a reInvite for the session timer , the script enter in the routine WITHINDLG . At this point the dialog seems to be lost. The accounting is made with dialog variable and in the 2 records INVITE and BYE , the fields corresponding to these values are missing. In all the other calls without the reInvite for session timers, the dialog is found and the variables are present. I don't understood what is problem ? Thanks for the help

1 1

Kamailio 4.1 and Cisco 79xx Integration Work
by Jonathan Hunter 04 Apr '14

04 Apr '14

Hi Guys, I'm doing some work, with Legacy Cisco phones, its a long story but I need to manipulate signaling due to firmware limitations. I am looking to implement a workaround, whereby I need to modify an ACK, so that it sends the RURI is sent to port 5060, and the destination Port at the IP layer is unchanged, I can do this with static values as below; sethostport("1.2.3.4:5060"); t_relay("1.2.3.4","38596"); In terms of doing this for multiple sites I need to use pseudo variables, so I wondered how this could be achieved, as t_relay or forward dont support dynamic values to this point, or do they? Any comments would be great. Many thanks Jon

2 2

Record-Route headers all over the place
by Phillman25 Kyriacou 04 Apr '14

04 Apr '14

Hi im receiving the following INVITE on my system, you can see that the Record-Route headers are all over the place, is there a way to group the Record-Route headers together followed by the via headers grouped together? INVITE sip:22000017@81.21.42.184;uniq=DBFF66459C85D6171B30016D2422B SIP/2.0 Record-Route: <sip:81.21.32.84;lr=on;ftag=as4c938f35;did=6da.a472> Record-Route: <sip:81.21.38.34;lr=on;ftag=as4c938f35> Via: SIP/2.0/UDP 81.21.32.84;branch=z9hG4bK6a96.dc3413ac122b4d4b8fd61bf19d95b1cf.0 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bK6a96.c81d2c2ad30db0820f05268ea67896f7.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-477861-22030313-22000017-74007 Record-Route: <sip:81.21.38.5;lr=on;ftag=as4c938f35> Record-Route: <sip:81.21.38.34;lr=on;ftag=as4c938f35;did=6da.0ff1> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bK6a96.5d12e421943fbaec75df9a6a8091d336.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7e570efe;rport=5060 Max-Forwards: 16 From: <sip:22030313@192.168.10.189>;tag=as4c938f35 To: <sip:22000017@81.21.38.34> Contact: <sip:22030313@192.168.10.189:5060> Call-ID: 74d7379b710c82f956d7cb31615c2581@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(11.7.0) Date: Fri, 04 Apr 2014 11:33:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 P-hint: outbound P-hint: outbound v=0 o=root 1396887739 1396887739 IN IP4 81.21.38.36 s=Asterisk PBX 11.7.0 c=IN IP4 81.21.38.36 t=0 0 m=audio 42746 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv thanks in advance! Phillip

2 1

What going on this SDP
by jaflong jaflong 04 Apr '14

04 Apr '14

Hi List, Can anyone help me understand why this is getting rejected Please note the specific message further dow the log. "Failed to parse SessionDescription. Failed to parse audio codecs correctly" This is on Chrome. On Firefox There is a further message in the console "Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning: Transport protocol type unsupported (UDP/TLS/RTP/SAVPF). | SDP Parsing Error: Invalid port format (17296) specified for transport protocol (Unsupported), parse failed." JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414 JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | TRANSPORT | sending WebSocket message: INVITE sip:9822@10.1.1.101 SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581 Max-Forwards: 69 To: <sip:9822@10.1.1.101> From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Contact: <sip:ce5egl03@flogvr403sb2.invalid;transport=ws;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 1744 v=0 o=- 3746191339358890844 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 m=audio 65223 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 10.10.10.63 a=rtcp:65223 IN IP4 10.10.10.63 a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0 a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0 a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0 a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0 a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0 a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0 a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0 a=ice-ufrag:Dgp8HIJdmr1lFPCQ a=ice-pwd:2yYxerrscdbTQhr0vbCTiju9 a=ice-options:google-ice a=fingerprint:sha-256 C8:E9:57:CB:85:63:F7:C5:FC:15:3D:8B:A8:10:94:F4:C9:BB:86:48:E3:EE:A0:5E:FA:42:14:55:6F:68:3F:B7 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:3445528109 cname:QbLF+sVLqHbEqUxY a=ssrc:3445528109 msid:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 768e2e47-bc86-473d-bc2c-6e2340ace772 a=ssrc:3445528109 mslabel:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 a=ssrc:3445528109 label:768e2e47-bc86-473d-bc2c-6e2340ace772 jssip-0.3.0.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581;rport=56527;received=10.10.10.63 To: <sip:9822@10.1.1.101> From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Server: DXI WebRTC Content-Length: 0 Warning: 392 10.10.10.48:6443 "Noisy feedback tells: pid=23455 req_src_ip=10.10.10.63 req_src_port=56527 in_uri=sip:9822@10.1.1.101 out_uri=sip:9822@10.10.10.111:5443 via_cnt==1" jssip-0.3.0.js:670 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WSS flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK9149581 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov To: <sip:9822@10.1.1.101>;tag=as06b3db08 Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Server: Easycall Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:9822@10.10.10.111:5443;transport=TLS> Content-Type: application/sdp Content-Length: 801 v=0 o=root 431209641 431209641 IN IP4 10.10.10.111 s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.10.10.111 t=0 0 m=audio 30490 UDP/TLS/RTP/SAVPF 0 126 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=ice-ufrag:5236e84b43b5d10c117e8ead0a340138 a=ice-pwd:51005fde4e6e9d3f1879fbbc15e0f092 a=candidate:Ha1f026f 1 UDP 2130706431 10.10.10.111 30490 typ host a=candidate:S5bec7504 1 UDP 1694498815 91.236.117.4 30490 typ srflx a=candidate:Ha1f026f 2 UDP 2130706430 10.10.10.111 30491 typ host a=candidate:S5bec7504 2 UDP 1694498814 91.236.117.4 30492 typ srflx a=connection:new a=setup:active a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 a=sendrecv jssip-0.3.0.js:670 Failed to parse SessionDescription. Failed to parse audio codecs correctly. jssip-0.3.0.js:4512 JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-0.3.0.js:2523 JsSIP | TRANSPORT | sending WebSocket message: ACK sip:9822@10.10.10.111:5443;transport=tls SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK431640 Max-Forwards: 69 To: <sip:9822@10.1.1.101>;tag=as06b3db08 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 ACK Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 jssip-0.3.0.js:519 JsSIP | TRANSPORT | sending WebSocket message: BYE sip:9822@10.10.10.111:5443;transport=tls SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK7689766 Max-Forwards: 69 To: <sip:9822@10.1.1.101>;tag=as06b3db08 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7436 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 jssip-0.3.0.js:519 JsSIP | RTC SESSION | closing INVITE session 43oclsi0sva6n347bk5c6tmeble9ov jssip-0.3.0.js:4193 JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392 JsSIP | DIALOG | dialog 43oclsi0sva6n347bk5c6tmeble9ovas06b3db08 deleted jssip-0.3.0.js:2543 JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0.js:187 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WSS flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK7689766 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov To: <sip:9822@10.1.1.101>;tag=as06b3db08 Call-ID: 43oclsi0sva6n347bk5c CSeq: 7436 BYE Server: Easycall Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0

4 6

Load new module in Kamailio
by Patrik Kristel 04 Apr '14

04 Apr '14

Hello all, I would like to ask, how can I load new module in Kamailio 4.1.2? Actually, I have an issue, when I tried to compile my kamaiio.cfg I've got error: root@kamailio:/usr/local/# kamailio -c kamailio.cfg loading modules under /usr/local/lib64/kamailio/modules/ 0(25392) ERROR: <core> [sr_module.c:587]: load_module(): ERROR: load_module: could not find module <websocket> in </usr/local/lib64/kamailio/modules/> 0(25392) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 323, column 12-25: failed to load module 0(25392) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command ws_handle_handshake 0(25392) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1083, column 27: unknown command, missing loadmodule? ERROR: bad config file (2 errors) when I look physically to the /usr/local/lib64/kamailio/modules/ there is some modules, but websocket.so is missing. So, how can I get and load module in Kamailio? Thank you for help! BR, Patrik Kristel

2 1

Crash on REGISTER
by Igor Potjevlesch 03 Apr '14

03 Apr '14

Hello, I experienced an issue yesterday. I move a customer to Kamailio 4.0.4 (previously he was on an old SER instance without issue). He uses the same SIP account for connecting two IPBX (same platform, same firmware). When the second REGISTER came on Kamailio and, may be 500ms/1s after (I have sufficient time to see 200 OK (2 bindings) on Wireshark), Kamailio crash with coredump generated. I know that I must analyse the coredump but I hope that could be a known issue. In addition, when I try to run "gdb" on the core files, I got the following: Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m 256 -M 64'. Program terminated with signal 11, Segmentation fault. #0 0x00000030f2230f30 in escape_string_for_mysql () from /usr/lib64/mysql/libmysqlclient.so.16 Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.9-33.el6.x86_64 libcom_err-1.41.12-12.el6.x86_64 libconfuse-2.6-2.el6.rf.x86_64 libselinux-2.0.94-5.3.el6.x86_64 mysql-libs-5.1.61-4.el6.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64 openssl-1.0.0-20.el6_2.5.x86_64 zlib-1.2.3-27.el6.x86_64 I guess that I'm missing packages to troubleshoot. Many thanks. Regards, Igor.

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sr-users April 2014