sr-users July 2013

sr-users@lists.kamailio.org

92 participants
149 discussions

Kamailio4.0 + Siremis4.0
by Lucky Santiago 31 Jul '13

31 Jul '13

Sorry for being a newbie, I have newly installed Kamailio 4.0 + Siremis 4.0 and i have setup until properly registering extension/subscriber, what i want to attain for now is a simple exten -> kamailio/siremis -> carrier/telco that includes dialplan and trunking, it will really help if there is some sort of documentation managing it via the web interface or siremis. Thanks.

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Re: [SR-Users] Replication problems Registration Using t_replicate ().
by Raphael Borges 30 Jul '13

30 Jul '13

Daniel, I had another similar problem. Would you like a procedure similar to the socket parameter. It is possible to add a new header and assign the parameter socket in server standby somehow? Thanks in advance. 2013/7/23 Raphael Borges <raphaelsilvaborges(a)gmail.com> > Daniel, I had another similar problem. Would you like a procedure similar > to the socket parameter. It is possible to add a new header and assign the > parameter socket in server standby somehow? > > Thanks in advance. > Raphael > > > 2013/7/22 Raphael Borges <raphaelsilvaborges(a)gmail.com> > >> Thanks Daniel, your help was very useful. >> >> With this I created a header by append_hf () before calling the function >> t_replicate and standby server retrieved the value of this header and >> assigns this value to the parameter avp >> Example >> $avp(s:rcv) = $hdr(IP-source); >> >> Best regards >> Raphael >> >> >> >> Hello, >>> >>> the received is taken from an avp specified by parameter received_avp of >>> registrar module: >>> >>> http://kamailio.org/docs/modules/devel/modules/registrar.html#idp4433296 >>> >>> You can carry the original details via a header on the replicated >>> REGISTER (using append_hf()), then take the header on the second server and >>> store its value in the avp. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 7/18/13 9:05 PM, Raphael Borges wrote: >>> >>> *Hello,* >>> >>> >>> I have two servers in kamailio duality, I am using the function >>> t_replicate () for replication log as below: >>> Server 1 active >>> t_replicate ("xxx.xxx.xxx.2", "5060"); >>> Server 2 reserves >>> t_replicate ("xxx.xxx.xxx.1", "5060"); >>> >>> But when the record is replicated to the standby server the parameter >>> received is being filled with the active server IP, where the IP would >>> Correro UAC, registry problems generating the backup server. >>> >>> Example >>> Server 1 active xxx.xxx.xxx.1 >>> >>> $ Kamctl ul show >>> Domain :: location table = 512 records = 1 = 1 max_slot >>> AOR :: 4301 >>> Contact :: sip: 4301(a)yyy.yyy.yyy.100; transport = udp >>> Q = >>> Expires :: 4566 >>> Callid :: ccd0431c-6a4c-1231-0a94-7356c4f466e6 >>> CSeq :: 46740150 >>> User-agent :: stepo_LITE >>> Received :: sip: yyy.yyy.yyy.100: 5060 >>> State :: CS_SYNC >>> Flags :: 0 >>> CFLAG :: 0 >>> Socket :: udp: xxx.xxx.xxx.1: 5060 >>> Methods :: 8159 >>> >>> Server 2 Reserve xxx.xxx.xxx.2 >>> >>> $ Kamctl ul show >>> Domain :: location table = 512 records = 1 = 1 max_slot >>> AOR :: 4301 >>> Contact :: sip: 4301(a)yyy.yyy.yyy.100; transport = udp >>> Q = >>> Expires :: 6665 >>> Callid :: ec7cfd42-6a7c-1231-0f94-7356c4f466e6 >>> CSeq :: 46750472 >>> User-agent :: stepo_LITE >>> Received :: sip: xxx.xxx.xxx.1: 5060 >>> State :: CS_SYNC >>> Flags :: 0 >>> CFLAG :: 0 >>> Socket :: udp: xxx.xxx.xxx.2: 5060 >>> Methods :: 8159 >>> >>> >>> >>> Tkanks >>> >>> Raphael >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >> >

3 5

kamailio and web services
by Mino Haluz 30 Jul '13

30 Jul '13

Hi, we would like to remove any SQL in our configuration and to fetch all data from web services (REST/RPC/whatever). What do you think, what would be the best way to integrate webservices in kamailio? app_python? app_perl? When calling any method from app_python configured script, it executes everytime the interpreter right? I am afraid this will be a bit slow. M

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Re: [SR-Users] kamailio call problem
by Jason Sia 30 Jul '13

30 Jul '13

Hi, Thanks for the advice. Here is the ngrep logs: U 2013/07/29 18:23:52.994331 49.144.184.97:41998 -> 198.23.160.81:5060 INVITE sip:639195015475@198.23.160.81 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.107:41998;branch= z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:639178864952@49.144.184.97:41998>. To: <sip:639195015475@198.23.160.81>. From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.5.3 stamp 70576. Content-Length: 357. . v=0. o=- 1375107832761441 1 IN IP4 192.168.0.107. s=X-Lite 4 release 4.5.3 stamp 70576. c=IN IP4 192.168.0.107. t=0 0. m=audio 53084 RTP/AVP 123 9 0 8 97 100 98 101. a=rtpmap:123 opus/48000/2. a=fmtp:123 useinbandfec=1. a=rtpmap:97 speex/8000. a=rtpmap:100 speex/16000. a=rtpmap:98 ILBC/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2013/07/29 18:23:52.994422 198.23.160.81:5060 -> 49.144.184.97:41998 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.107:41998 ;received=49.144.184.97;branch=z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport=41998. To: <sip:639195015475@198.23.160.81 >;tag=80403214130c49515b8f7d7842a4c119.ef65. From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="198.23.160.81", nonce="51f67b16000000fae8c70592af760c78fa39cd43f84c1a6b". Server: OpenSIPS (1.8.3-notls (x86_64/linux)). Content-Length: 0. . U 2013/07/29 18:23:53.299025 49.144.184.97:41998 -> 198.23.160.81:5060 ACK sip:639195015475@198.23.160.81 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.107:41998 ;branch=z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport. Max-Forwards: 70. To: <sip:639195015475@198.23.160.81 >;tag=80403214130c49515b8f7d7842a4c119.ef65. From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. CSeq: 1 ACK. Content-Length: 0. . U 2013/07/29 18:23:53.317842 49.144.184.97:41998 -> 198.23.160.81:5060 INVITE sip:639195015475@198.23.160.81 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.107:41998 ;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:639178864952@49.144.184.97:41998>. To: <sip:639195015475@198.23.160.81>. From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="639178864952",realm="198.23.160.81",nonce="51f67b16000000fae8c70592af760c78fa39cd43f84c1a6b",uri=" sip:639195015475@198.23.160.81 ",response="1605e4e6b20dbbd6be5917c94dd3961e",algorithm=MD5. Supported: replaces. User-Agent: X-Lite release 4.5.3 stamp 70576. Content-Length: 357. . v=0. o=- 1375107832761441 1 IN IP4 192.168.0.107. s=X-Lite 4 release 4.5.3 stamp 70576. c=IN IP4 192.168.0.107. t=0 0. m=audio 53084 RTP/AVP 123 9 0 8 97 100 98 101. a=rtpmap:123 opus/48000/2. a=fmtp:123 useinbandfec=1. a=rtpmap:97 speex/8000. a=rtpmap:100 speex/16000. a=rtpmap:98 ILBC/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2013/07/29 18:23:53.318191 198.23.160.81:5060 -> 49.144.184.97:41998 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.0.107:41998 ;received=49.144.184.97;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport=41998. To: <sip:639195015475@198.23.160.81>. From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. CSeq: 2 INVITE. Server: OpenSIPS (1.8.3-notls (x86_64/linux)). Content-Length: 0. . U 2013/07/29 18:24:03.328456 198.23.160.81:5060 -> 49.144.184.97:41998 SIP/2.0 477 Send failed (477/TM). Via: SIP/2.0/UDP 192.168.0.107:41998 ;received=49.144.184.97;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport=41998. To: <sip:639195015475@198.23.160.81 >;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9. From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. CSeq: 2 INVITE. Server: OpenSIPS (1.8.3-notls (x86_64/linux)). Content-Length: 0. . U 2013/07/29 18:24:03.582094 49.144.184.97:41998 -> 198.23.160.81:5060 ACK sip:639195015475@198.23.160.81 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.107:41998 ;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport. Max-Forwards: 70. To: <sip:639195015475@198.23.160.81 >;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9. From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. CSeq: 2 ACK. Content-Length: 0. Thanks, Jason On Mon, Jul 29, 2013 at 3:33 PM, Gertjan Wolzak <g.wolzak(a)foize.com> wrote: > Hello Jason,**** > > ** ** > > What you should do is on the Kamailio server do an ngrep, that > information will make it possible for the people on the list to take a look > and help you.**** > > ** ** > > By only telling your experiences it is hard to help you.**** > > ** ** > > Just give the command: ngrep –qt –W byline port 5060 and then do the > test calls, so from android to xlite and vice versa.**** > > ** ** > > Save the info and paste it in the case.**** > > ** ** > > Good luck.**** > > ** ** > > Gertjan**** > > ** ** > > *From:* sr-users-bounces(a)lists.sip-router.org [mailto: > sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Jason Sia > *Sent:* zondag 28 juli 2013 10:52 > *To:* sr-users(a)lists.sip-router.org > *Subject:* [SR-Users] kamailio call problem**** > > ** ** > > Hi,**** > > I installed kamailio v 3.1.6 rpm. I used the default configuration. > Kamailio started. I have two clients one is an android phone using native > sip client, and the other one x-lite. I can call x-lite to phone but not > the other way around. When I restarted kamailio, I can call phone to > x-lite but not the other way around. The first one I did when I restart > kamailio will be the only leg that will be working, then I have to restart > again if I want to do the reverse. Has anyone encountered this problem? > How do I fix it? I'm on a LAN, kamailio server and the two clients are on > the same LAN.**** > > ** ** > > Thanks, > Jason**** > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

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siremis 4.0 login issue
by Premchandiran 30 Jul '13

30 Jul '13

Hi All, I am working on siremis 4.0 , after successful installation through web interface I tried to login using admin as username and password I am getting a popup debug window which is blank. Log is not written, NOTICE.log is generated which contains 07/29/2013','09:12:37','NOTICE','ErrorHandler','preg_replace(): The /e modifier is deprecated, use preg_replace_callback instead',''. In apache2 error log I can see following errors Undefined class constant 'MYSQL_ATTR_USE_BUFFERED_QUERY' in /opt/siremis-4.0.0/openbiz/bin/BizSystem.php on line 405, referer: http://192.168.134.145/siremis/index.php/user/login I kindly request to overcome the issue. Thanks and Regards, Prem Chandiran M

1 0

cnxcc module insufficient credit
by Jayesh Nambiar 30 Jul '13

30 Jul '13

Hi All, I was testing out the cnxcc module and it works well for prepaid credit handling. One problem that I saw with the module is that if the credit is insufficient it still allows the call to pass and then disconnect within a second. Is it possible that if the credit is insufficient for the call, the dialog can be terminated before the call is placed with a 4xx kind of response !! Just a thought. Thanks, --- Jayesh

2 6

Problem with forward on busy.
by LAA 29 Jul '13

29 Jul '13

Hello Daniel, I think that debugger module was released with Kamailio version 3.1.0. wasn't it? As I'm running Kamailio 3.0.0, I have set up debug=9. Here you have my config file, the raw capture of the call and the lines in kamailio log file. I see some messages regarding cpl-c module, and I was experiencing some problems with it too. I tested without cpl-c module and I get the same trace but no messages in log file. Thanks in advance. Regards. Luis. https://www.dropbox.com/s/t79ic02b6h85ppj/onbusy_DCM_txt https://www.dropbox.com/s/20c5pqilocey052/kamailio.cfg LOG ******************* ul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618] : : tm [t_hooks.c:211]: BUG:tm:register_tmcb: no transaction found Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: cpl-c [cpl_proxy.h:482]: failed to register TMCB_RESPONSE_OUT callback Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: cpl-c [cpl_run.c:1040]: runtime error Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: tm [tm.c:1168]: ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established

2 2

rtpproxy-ng doesn't work with mediaproxy-ng
by Khue Nguyen Minh 29 Jul '13

29 Jul '13

Hi all, I am using rtpproxy-ng to control mediaproxy-ng. I was install and config follow this guide: https://github.com/sipwise/mediaproxy-ng when I run kamailio with rtpproxy-ng module and mediaproxy-ng I got error: mediaproxy-ng[25216]: Failed to properly parse UDP command line '30514_2 d7:command4:pinge' from 127.0.0.1:54621, using fallback RE ERROR: rtpproxy-ng [rtpproxy.c:1381]: rtpp_test(): proxy responded with invalid response How I can fix it? Thanks, Khue.

4 10

MSRP communication through Kamailio
by Rajkumar Kanniappan 29 Jul '13

29 Jul '13

Hi, Is it possible to make MSRP to pass through kamailio instead of peer to peer? For me, my sip client registers with kamailio without any problem and able to chat with other sip client as well. But the chat communication using MSRP is always peer to peer, because of the SDP negotiation. But I need make the msrp messages to pass through the kamailio server. Please help me in configuring the kamailio. Thanks ________________________________ SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 3

siremis 4.0 login issue
by Premchandiran 29 Jul '13

29 Jul '13

1 0

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sr-users July 2013