sr-users June 2013

sr-users@lists.kamailio.org

94 participants
170 discussions

Handling forking in kamailio
by Ram Anji 04 Jun '13

04 Jun '13

Hi, I want to handle the responses for forked Requests in kamailio. Please suggest me that where i can handle this in configuration. Here is my requirement. If Clinet A is registered with kamailio server on pc 1 and pc 2. If Client B calls A, Server is sending INVITE to A for both PC 1 and PC 2. As per current implementation Server closing the INVITE session if any one(pc1/pc2) responds with 486. But i want to handle the 486 response, and wait till other one(pc1/pc2) also responds instead of closing INVITE session. Regards, Raman

2 1

Re: [SR-Users] is_uri_host_local() maybe not considering IP:PORT
by DanB 03 Jun '13

03 Jun '13

Klaus, Thanks for the tip. I have been using (uri==myself) up to know but I was trying to move towards a more multidomain configuration, was kinda hoping to get rid of hardcoded stuff in particular servers. Daniel, Can the fact that port in ruri makes host part not longer matching as own domain be considered a bug or it is the accepted working mechanism? The "register_myself" parameter made me think that it should match host part at least (port is not important for me), but in my case port in ruri makes the whole host not matching. Hope I have clearly expressed my thoughts ;). Ta, DanB

2 3

is_uri_host_local() maybe not considering IP:PORT
by DanB 03 Jun '13

03 Jun '13

Guys, I was wondering if I am doing something wrong or domain module not considering properly local ips within is_uri_host_local(). Although I have domain loaded with register_myself on 1, the request going to my IP is simply not matching local domain (listening on 127.0.0.1 port 5070). Bellow the trace of such request: """ # U 2013/05/30 12:35:56.436485 10.10.10.21:5060 -> 127.0.0.1:5070 OPTIONS sip:127.0.0.1:5070 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK4866.b4f45983000000000000000000000000.0. To: <sip:127.0.0.1:5070>. From: <sip:ep@iec.itsyscom.com>;tag=ae9b2706b606c3acb0ebe4f1c8f81cee-f20d. CSeq: 10 OPTIONS. Call-ID: 467de807489c4482-3002(a)10.10.10.21. Max-Forwards: 70. Content-Length: 0. User-Agent: iClass4-EP 4.0.0. . # U 2013/05/30 12:35:56.440410 127.0.0.1:5070 -> 10.10.10.21:5060 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK4866.b4f45983000000000000000000000000.0. To: <sip:127.0.0.1:5070>;tag=46a6e639fa023622ac1ba4fea686e961.d61e. From: <sip:ep@iec.itsyscom.com>;tag=ae9b2706b606c3acb0ebe4f1c8f81cee-f20d. CSeq: 10 OPTIONS. Call-ID: 467de807489c4482-3002(a)10.10.10.21. Server: iClass4-AP 4.0.0. Content-Length: 0. """ My script looks something like bellow, so Address Incomplete should never be reached: """ #!define LISTEN_IP 127.0.0.1 #!define LISTEN_PORT 5070 ... listen=LISTEN_IP port=LISTEN_PORT auto_aliases=yes ... # ----- domain params ----- modparam("domain", "db_url", DBURL) modparam("domain", "register_myself", 1) ... if (is_method("OPTIONS") && is_uri_host_local()) { options_reply(); exit; } ... if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } """ I am running on git master with the test patch Daniel did few days back for me. Thanks in advance for any tip! DanB

3 3

Asterisk 11.4.0 + Kamailio 4.0.1 integration
by Ravindra Gowda 03 Jun '13

03 Jun '13

Dear All, I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in Ubuntu 12.04 LTS version. I tried it by using the following link : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. But after all, when i tried to start Kmailio server with /etc/init.d/kamailio start command , i am getting the following errors: asterisk-kamailio@asterisk-kamailio:/$ sudo /etc/init.d/kamailio start [sudo] password for asterisk-kamailio: /etc/init.d/kamailio: 1: /etc/init.d/kamailio: sed: not found /etc/init.d/kamailio: 118: /etc/init.d/kamailio: arithmetic expression: expecting primary: "" what could be the wrong, and how can i solve it? Please share your valuable suggestions on this. Any help will greatly appreciate. *Thank you and Best regards,* *R Ravindra **Gowda* *Engineer - Systems and Products*. *Thrikasa Technologies* *| Office: +91 40 23260434 | Mob: +918885284050 * *| Fax:+91 40 23221045* *| email: ravindra(a)thrikasa.in* *| Web: www.thrikasa.in* 'If you don't give up,you cannot fail'. The information contained in this communication is proprietary to Thrikasa Technologies., and/or third parties, may contain classified or privileged information, and is intended only for the use of the intended addressee thereof. If you are not the intended addressee, please be aware that any use, disclosure, distribution and/or copying of this communication is strictly prohibited.If you receive this communication in error, please notify the sender immediately and delete it from your computer.

2 1

Faild to add Subscription via WebRTC2SIP websocket GW
by Raj Roy Ghandhi 03 Jun '13

03 Jun '13

Hi Friends, I am using Kamailio 4.0 SIP Proxy GW + WebRTC2SIP 2 web socket GW to maintain the presence status of SIPMP5 clients. When I subscriber for presence statue of contact I get the following error and no record get inserted into the "active_wathcers" table. 5(7435) WARNING: <core> [msg_translator.c:2499]: TCP/TLS connection (id: 0) for WebSocket could not be found 5(7435) ERROR: tm [t_msgbuilder.c:1367]: assemble_via: via building failed 5(7435) ERROR: tm [t_msgbuilder.c:1540]: build_uac_req(): Error while assembling Via 5(7435) ERROR: tm [uac.c:338]: t_uac: Error while building message 5(7435) ERROR: presence [notify.c:1591]: in function tmb.t_request_within 5(7435) ERROR: presence [subscribe.c:1613]: send Notify not successful 5(7435) ERROR: presence [subscribe.c:2129]: in function handle_expired_record 5(7435) ERROR: presence [subscribe.c:464]: Failed to delete subscription from memory Here is the SIPML5 console log for that perticular area State machine: tsip_dialog_generic_Any_2_InProgress_X_oSubscribe tsk_utils.js?svn=20:110 SEND: SUBSCRIBE sip:1005@sip.mysip.com SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLmNGlDNlChMuoGqF9qFzMS7dOk5h0E5T;rport From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com> Contact: "undefined"<sip:1004@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27396 SUBSCRIBE Expires: 200 Content-Length: 0 Max-Forwards: 70 Event: presence Accept: application/pidf+xml tsk_utils.js?svn=20:110 session event = type = connecting - description = Connecting... test_api.html:51 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=20124;received=192.168.1.55;branch=z9hG4bKLmNGlDNlChMuoGqF9qFzMS7dOk5h0E5T From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com>;tag=b01221cac0ed2edd8409481b9ce2d72c.ac8c Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27396 SUBSCRIBE Content-Length: 0 Proxy-Authenticate: Digest realm="sip.mysip.com ",nonce="UaaBx1GmgJtqCoZzmu9Tskw+AmewKCiC",stale=FALSE Server: kamailio (4.0.1 (x86_64/linux)) tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=20:110 SEND: SUBSCRIBE sip:1005@sip.mysip.com SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKpPobVdaUhPHunxtfAuQFOlkfEeGs2ksE;rport From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com> Contact: "undefined"<sip:1004@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27397 SUBSCRIBE Expires: 200 Content-Length: 0 Max-Forwards: 70 Proxy-Authorization: Digest username="1004",realm="sip.mysip.com ",nonce="UaaBx1GmgJtqCoZzmu9Tskw+AmewKCiC",uri="sip:1005@sip.mysip.com ",response="223e983b8e2de2b04dbcd2ac9fc0b17d",algorithm=MD5 Event: presence Accept: application/pidf+xml tsk_utils.js?svn=20:110 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=SIP/2.0 202 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=20124;received=192.168.1.55;branch=z9hG4bKpPobVdaUhPHunxtfAuQFOlkfEeGs2ksE From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 Contact: <sip:192.168.1.15:8080;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27397 SUBSCRIBE Expires: 200 Content-Length: 0 Server: kamailio (4.0.1 (x86_64/linux)) tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_InProgress_2_Connected_X_2xx tsk_utils.js?svn=20:110 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=NOTIFY sip:1004@192.168.1.55:20124;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.15:8080 ;branch=z9hG4bK4b8a.ab27a051000000000000000000000000.0 From: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 To: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 Contact: <sip:192.168.1.15:8080;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 2 NOTIFY Content-Length: 0 User-Agent: kamailio (4.0.1 (x86_64/linux)) Max-Forwards: 70 Event: presence Subscription-State: active;expires=200 tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_Any_2_Any_X_iNotify tsk_utils.js?svn=20:110 session event = type = i_ao_request - description = OK test_api.html:51 session event = type = connected - description = Connected test_api.html:51 SEND: SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.1.15:8080 ;branch=z9hG4bK4b8a.ab27a051000000000000000000000000.0 From: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 To: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 Contact: <sip:1004@df7jal23ls0d.invalid;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 2 NOTIFY Content-Length: 0 tsk_utils.js?svn=20:110 session event = type = i_notify - description = Incoming NOTIFY test_api.html:51 NOTIFY content = test_api.html:72 NOTIFY content-type = null Please guide me on this. Best Regards, Roy.

2 1

Does Kamailio check for CSeq, Call-ID when unregister ?
by Khoa Pham 03 Jun '13

03 Jun '13

Hi, When I want to unregister, I have 2 ways 1. Append ;expires=0 in Contact field 2. Add another Expires : 0 header with CSeq, Call-ID not the same with when i register Does Kamailio check for the saem Cseq, Call-ID to allow unregistration ? -- Khoa Pham HCMC University of Science www.fantageek.com

3 2

Asterisk 12.04 LTS + kamailio 4.0.1 Integration problem.
by Ravindra Gowda 03 Jun '13

03 Jun '13

Dear All, I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in Ubuntu 12.04 LTS version. I tried it by using the following link : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. But i am unable to accomplish it, i just ended with some mysql errors. I am very new to this aspect. So can anyone suggest me any other standard reference procedure or any tutorials to do this. PS: my host Architecture: i686 Any help will greatly appreciate. *Thank you and Best regards,* *R Ravindra **Gowda* *Thrikasa Technologies* *| Office: +91 40 23260434 | Mob: +918885284050 * *| Fax:+91 40 23221045* *| email: ravindra(a)thrikasa.in* *| Web: www.thrikasa.in* 'If you don't give up,you cannot fail'. The information contained in this communication is proprietary to Thrikasa Technologies., and/or third parties, may contain classified or privileged information, and is intended only for the use of the intended addressee thereof. If you are not the intended addressee, please be aware that any use, disclosure, distribution and/or copying of this communication is strictly prohibited.If you receive this communication in error, please notify the sender immediately and delete it from your computer.

2 1

{voipfreak.net}
by Laszlo 02 Jun '13

02 Jun '13

http://guardianwarriorfoundation.com/zffhwbdc/tadou.hdoho

1 0

[6/2/2013 12:59:07 PM]
by Laszlo 02 Jun '13

02 Jun '13

http://helptocats.ru/zwwnzzty/zldopj.wjfxdwp

1 0

Re: [SR-Users] SIP Trunks
by Keith Hubner 01 Jun '13

01 Jun '13

Hi, Authentication for users is working fine for me, I add a user to the db and the phone can register. I am looking to connect a SIP trunk but there doesn't seem to be any authentication? What I want is to allow users to connect from any IP but only allow SIP trunks to connect from certain IPs and authenticate. Thanks, Keith

2 1

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sr-users June 2013