sr-users August 2010

sr-users@lists.kamailio.org

47 participants
72 discussions

help to fix bug in nathelper module
by truong ngoc THANH 27 Aug '10

27 Aug '10

hi all, I have configure nathelper in kamailio, but it does not work as I expect. I using function : force_rpt_proxy("","115.X.X.190") to change ip_add of SDP packet. But I see that value c= <ip add > not change to c=<115.X.X.190>. thanks for help. ThanhTruong HTK-INC Co.,Ltd http://www.htk-inc.com/ http://pbxvn.com/forum/

1 0

LCR prefix based
by inge 26 Aug '10

26 Aug '10

Hi, I'm still disapointed after reading the documentation of LCR module. I use SER 0.9.9 and I'm wondering if it's possible to use LCR based on destination prefix ? I mean that if I call 056XXXXX I want to use GW1. For 06XXXX I want to use GW2, etc. Thank you for your opinion. Regards, Adrien

4 4

Kamailio 1.5 - Dispatcher priority usage
by Geoffrey Mina 26 Aug '10

26 Aug '10

Hello, I have a question about the "priority" in the dispatcher module. I looked in the docu-wiki and there was nothing which actually described how this field is used (exactly), so I can only guess or ask the list. Say I have 3 destinations in a set: set: 1 dest: sip:1.2.3.4:5060 priority: 1 set: 1 dest: sip:1.2.3.5:5060 priority: 1 set: 1 dest: sip:1.2.3.6:5060 priority: 0 I am using ds_select_domain("1","4") to retrieve my list of destinations. In this scenario, would the calls only route to 1.2.3.6 if .4 and .5 were both out of service? Thanks.

3 4

help to configure RTP stream with NAT.
by truong ngoc THANH 26 Aug '10

26 Aug '10

hi all, i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so how to configure in kamailio or which module make RTP stream direct from sip client to another one ? please suggest if anyone know. thanks. TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh

4 8

SIP routing problem when kamailio uses 2 interfaces
by Gnaneshwar Gatla 26 Aug '10

26 Aug '10

Hello All, I'm using Kamailio as a Border controller for my VoIP Research project at my school. The problem I'm facing is Kamailio routes the traffic to the private network where my asterisk machine is listening. The asterisk machine responds to the Kamailio using the public network but not the private. Is this an issue with Asterisk? If so how do I resolve it? client kam_public Kam_pvt Ast_pvt Ast_public | | | | | | | | | | | INVITE | | | | |---------------->| | INVITE | | | | |---------->| | | | | | | | | | | | | | | | | | | | 404 | | | |<----------|-----------|-----------| | | | | | | | | ACK | | | | |---------->| | | | | | | | 404 | | | | |<----------------| | | | | | | | | | ACK | | | | |---------------->| | | | | | | | | | | | | | Regards Gnaneshwar

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Re: [SR-Users] removing stuff
by Daniel-Constantin Mierla 25 Aug '10

25 Aug '10

please cc all the time the mailing list: sr-users(a)lists.sip-router.org On 8/25/10 1:16 PM, Anders wrote: > Thanks Daniel. I found out that the sip_trace was pretty straight > forward to turn off, by running "kamctl fifo sip_trace off". So far, > so good - no more massive data collection on that. Now I need to stop > the radius accounting from collecting data to the 'radacct' table. I > want to continue whatever goes to 'acc' - but I don't know if that > data comes from the radius accounting too(?). So what do I do to stop > collecting to 'radacct' but continue to 'acc'? Accounting to radius is done by setting radius flag. Just remove all parameters to acc module that starts with radius_* Cheers, Daniel > Thanks!! > > --Anders > > On Mon, Aug 23, 2010 at 11:09 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >> Hello, >> >> On 8/23/10 9:32 AM, Anders wrote: >>> Hi, >>> >>> This concerns third party tools...I suspect. >>> >>> First - I run Kamailio 1.5 and use Siremis to calculate the calls >>> (actually just the stored procedure and cron job defined, not the >>> interface). >>> >>> I have earlier been messing around with CDRTools and radius, but I >>> don't use these anymore. However, they seem to still be generating >>> lots of data on this server and overloading MySql - so I really want >>> to get rid of them. Specifically are my radacct and sip_trace tables >>> in MySql getting huge, and keep being populated...and I want to stop >>> this 'madness'. What do I do to stop/remove this? without putting the >>> current installation at risk) >> what do you want to clean? the radius accounting and siptrace from your >> config? >> >> If yes, then just edit kamailio config and remove parameters for radius >> accounting and the siptrace module as well as sip_trace() function and >> siptrace parameter from there. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com/ >> >> -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

help to fix error in carrierroute module.
by truong ngoc THANH 25 Aug '10

25 Aug '10

hi all, please help to correct configure in carrierroute module. i have configure carrierroute in kamailio as bellow. ========= kamailio.cfg === route[1] { # route calls based on hash over callid # choose route domain 0 of the default carrier if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In case of failure, re-route the request t_on_failure("1"); # Relay the request to the gateway t_relay(); } } failure_route[1] { # In case of failure, send it to an alternative route: if (t_check_status("408|5[0-9][0-9]")) { #choose route domain 1 of the default carrier if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ } else { xlog("failure 2"); t_on_failure("2"); t_relay(); } } } =========\ in database: mysql> select * from carrierroute; +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ | 1 | 1 | 1 | 1000 | 0 | 0 | 1 | 0 | 192.168.1.42 | | | FRANCE | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ 1 row in set (0.00 sec) mysql> select * from carrier_name; +----+---------+ | id | carrier | +----+---------+ | 1 | default | +----+---------+ mysql> select * from domain_name; +----+---------+ | id | domain | +----+---------+ | 1 | default | +----+---------+ ================ in server 192.168.1.40 : subscriber table: mysql> select * from subscriber; +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | 1 | 192.168.1.40 | 1 | | b8ee4a7080092d7fd27357de1799b111 | d1343e4f0259199ed6b06d6338d39903 | NULL | | 2 | 2 | 192.168.1.40 | 2 | | 25f61f90a19afe508629127838c11509 | 7f4ec87f39f48d0d8f8a537cfc34c5c9 | NULL | | 3 | 3 | kamailio.com | 3 | | 63a131bf02b4123073970731a4525470 | 64e49a4f89317c5ee757ba0366ec9307 | NULL | | 4 | 1000 | 192.168.1.40 | 1000 | | 8e9f00da8e49cc8b844b6cee1752b949 | 2bff4783b0ce3412bf33e7400fbbedce | NULL | | 5 | 4 | 192.168.1.40 | 4 | | 037cf4277ce31c104e4b1bc7bbcf8ad4 | cb2321d9ac8bfe749cedc1d66cad5d73 | NULL | | 6 | 1 | kamailio.com | 1 | | 31fb6891a2fcb9b13a952ce3f12359c9 | 84add6cffd100f8ca3c8660c3c9617b5 | NULL | | 7 | 1 | 192.168.1.42 | 1 | | ad57b2abdf288984bc4a897a9e7f2ba9 | 60b9ab690724e5a535df794131e2345f | NULL | | 8 | 1000 | 192.168.1.42 | 1000 | | 0a9fb3a1de135a32b9090798a0dcb240 | bf5a95934a59b480dbf85abe386144b6 | NULL | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ I try to make call from 1(a)192.168.1.40 to 1000(a)192.168.1.42 but the call failed. sip client 1(a)192.168.1.40 show that : server failure and sip client 1000(a)192.168.1.42 ringing. ============ sip trace for 192.168.1.40: U 192.168.1.40:5060 -> 192.168.1.42:5060 INVITE sip:1000@192.168.1.42 SIP/2.0. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKf92a.649f39e3.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport=50495. Max-Forwards: 69. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 279. . v=0. o=3cxVCE 96031755 384028260 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40002 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.40:5060 -> 192.168.1.10:50495 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL). Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport=50495. To: <sip:1000@192.168.1.40:5060>;tag=ae07808f44ebda0182a54cf170e36dbe.ba60. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 INVITE. Server: kamailio (3.0.1 (i386/linux)). Content-Length: 0. . U 192.168.1.10:50495 -> 192.168.1.40:5060 ACK sip:1000@192.168.1.40:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport. Max-Forwards: 70. To: <sip:1000@192.168.1.40:5060>;tag=ae07808f44ebda0182a54cf170e36dbe.ba60. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 ACK. Content-Length: 0. and sip trace in 192.168.1.42: ==== U 192.168.1.40:5060 -> 192.168.1.42:5060 INVITE sip:1000@192.168.1.42 SIP/2.0. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Max-Forwards: 69. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 280. . v=0. o=3cxVCE 227296140 367695825 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40000 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.42:5060 -> 192.168.1.40:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Server: kamailio (3.0.1 (i386/linux)). Content-Length: 0. . U 192.168.1.42:5060 -> 192.168.1.10:6872 INVITE sip:1000@192.168.1.10:6872;rinstance=c308344d9524d1a4 SIP/2.0. Record-Route: <sip:192.168.1.42;lr=on>. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKc4e.f9fad397.0. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Max-Forwards: 68. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 280. . v=0. o=3cxVCE 227296140 367695825 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40000 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.10:6872 -> 192.168.1.42:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKc4e.f9fad397.0. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Record-Route: <sip:192.168.1.42;lr>. Record-Route: <sip:192.168.1.40;lr=on>. Contact: <sip:1000@192.168.1.10:6872;rinstance=c308344d9524d1a4>. To: <sip:1000@192.168.1.40:5060>;tag=660be63a. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 0. ======================== and in server 192.168.1.42:subscriber table: mysql> select * from subscriber; +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | 1000 | 192.168.1.42 | 1000 | | 0a9fb3a1de135a32b9090798a0dcb240 | bf5a95934a59b480dbf85abe386144b6 | NULL | | 2 | 1 | 192.168.1.42 | 1 | | ad57b2abdf288984bc4a897a9e7f2ba9 | 60b9ab690724e5a535df794131e2345f | NULL | | 3 | 1 | 192.168.1.40 | 1 | | b8ee4a7080092d7fd27357de1799b111 | d1343e4f0259199ed6b06d6338d39903 | NULL | please suggest to get correct configue. thanks for looking. TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646begin_of_the_skype_highlighting 0984 480 646 end_of_the_skype_highlighting Y!M: ngoc217thanh

2 2

Web and SIP Telephony
by Doddle WebPhone 24 Aug '10

24 Aug '10

This may be useful for calling Kamailio SIP server right from web pages / browsers ( webphone/click2call) For Instance, a web phone link to call: <a href=" http://widget.doddlephone.com/embed/web3phone.jsp?sipserver=proxy.ideasip.c…<http://widget.doddlephone.com/embed/web3phone.jsp?sipserver=proxy.ideasip.c…> "> Tel: +1 234 567 890 </a> We can embed it as a widget/iframe or even deploying/hosting the web-driven phone on our Web servers and integrating with our applications ( for instance, filling such SIP parameters dynamically with PHP, .NET, Ruby,etc) Sergio

2 4

removing stuff
by Anders 23 Aug '10

23 Aug '10

Hi, This concerns third party tools...I suspect. First - I run Kamailio 1.5 and use Siremis to calculate the calls (actually just the stored procedure and cron job defined, not the interface). I have earlier been messing around with CDRTools and radius, but I don't use these anymore. However, they seem to still be generating lots of data on this server and overloading MySql - so I really want to get rid of them. Specifically are my radacct and sip_trace tables in MySql getting huge, and keep being populated...and I want to stop this 'madness'. What do I do to stop/remove this? without putting the current installation at risk) Thanks!! //Anders

2 1

Re: [SR-Users] Kamailio-3.0.3 Config file error‏
by Anthony Mark 23 Aug '10

23 Aug '10

Dear Sir, I just wanted to inform the good people on this forum that i had corrected the problem with the Config file, and the server is running. I noticed on going over the file again that i Fat fingered some keys ;) Sorry for any inconvenience my post may have caused anyone. Thanks for the good work you are doing. Best Regards, Anthony.

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sr-users August 2010