sr-users April 2013

sr-users@lists.kamailio.org

94 participants
133 discussions

Re: [SR-Users] sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 06 Apr '13

06 Apr '13

hi, i have set the following flag it is still not capturing outbound SIP messages. { setflag(22) sip_trace(); } Daniel-Constantin Mierla<http://www.mail-archive.com/search?l=sr-users@lists.sip-router.org&q=from:%…> Fri, 05 Apr 2013 00:45:16 -0700<http://www.mail-archive.com/search?l=sr-users@lists.sip-router.org&q=date:2…> Hello, On 3/31/13 4:27 PM, Ramaseshi reddy kolli wrote: Hi, I am using sip_trace function after next_branches(), it is not capturing outbound SIP INVITE request can anyone please help me. iirc, you have to set a flag to trace outgoing messages. the sip_trace()function just takes the snapshot of processed messages. Cheers, Daniel

2 1

sip_capture columns
by Grant Bagdasarian 06 Apr '13

06 Apr '13

Hello, I couldn't find any documentation on what the table columns of the sip_capture module are used for. Most of them are straightforward, but I still don't know what the following columns mean and what values to expect: - PID_user - Proto - Family - Rtp_stat - Type - Node Type has for example values 1 and 2. Im guessing 1 is for requests and 2 is for responses. I hope someone could clear this out for me. Thanks, Grant

2 2

No RTP between rtpproxy and smartphone application
by aaaa aaaa 05 Apr '13

05 Apr '13

Hello, I have the following topology. Kamailio as SIP Proxy and Asterisk as B2BUA. In Kamailio I use rtpproxy for NAT users. The problem that I have is the following: -When the CSipSimple is registered the first call is always unsuccesfull. The following calls are always successful. If the application remains idle for a while the next call is unsuccessful again. From traces I saw that everything is OK in signaling part. -When I try with Zoiper and 3CX I have 1/20 successful calls. In any case from Asterisk side these calls can be seen as active but no media. Thanx, Theo

3 2

Changing Domain in To URI
by qmail＠polarismail.com 05 Apr '13

05 Apr '13

Hello, As it happens we run Kamailio 3.2.0 on Debian Lenny which for some reason does not support uac_replace_to()! Anyhoo, we are trying to change the domain part in the To URI. What is the best way to do it ? Thank you!

2 1

sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 05 Apr '13

05 Apr '13

Hi, I am using sip_trace function after next_branches(), it is not capturing outbound SIP INVITE request can anyone please help me. Thanks Ram

2 1

Kamailio 4.0 configuration
by Martin Krška 04 Apr '13

04 Apr '13

Hi everyone, I'm trying to configure some basic functions in Kamailio 4.0. Now let's say I have two users, for example Alice and Bob. When Alice adds Bob to her contacts, I need from her to ask Bob for permission to watch his presence. It's probably a trivial question but could anyone tell me what do I need for this to work, like what modules are responsible for this etc. Best Regards, Martin

2 1

db_mode 3 in usrloc module
by Morten Isaksen 04 Apr '13

04 Apr '13

Hi, I have an older installation of Kamailio (3.1.1). It is configured with modparam("usrloc", "db_mode", 2) I tried to change this to modparam("usrloc", "db_mode", 3) to not have to restart kamailio when I change the alias table directly from SQL. But then kamailio shutsdown after about one second with this error: Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [db.c:408]: invalid version 0 for table trusted found, expected 5 (check table structure and table "version") Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: permissions [trusted.c:250]: error during table version check. Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module permissions (/usr/local/lib/kamailio/modules_k/permissions.so) Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 28, pid 11510, "tcp receiver child=6" Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11512]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 44 Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11482]: ALERT: <core> [main.c:738]: child process 11510 exited normally, status=255 Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11482]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD I have not made any changes to the permissions/trusted module, so I suspect it is some shared database conection problem. Is it a known bug in this version? I would prefer not to upgrade the installation and hopes one of you can point me to a patch I can backport. -- Morten Isaksen

2 7

Problem with expired SIP Etag
by David 04 Apr '13

04 Apr '13

Hello, I have two Kamailio servers. Running 3.3.2. One uses PUA to send PUBLISH to the second which has presence module. When the presence server replies to the PUBLISH, it includes a SIP-ETag and an Expires header. So, in the case I caught on my servers yesterday, the reply had an Expires header of 34. Trouble is, 60 seconds later, when PUA sent the PUBLISH, it had the SIP-ETag in the PUBLISH header even though the entity tag was expired. But RFC3903 says that when the publish expires, the entity tag expires as well. The publish was for a dialog-info and both PUBLISHes were generated using the pua_mi module command pua_publish. I don't set the etag in the pua_publish command, it's kamailio that remembers it. Here is grep pua kamailio.cfg on the first server : loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_dialoginfo.so" modparam("pua_dialoginfo", "include_localremote", 1) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "include_callid", 0) modparam("pua", "hash_size", 15) modparam("pua", "db_mode", 0) modparam("pua_dialoginfo", "send_publish_flag", 13) modparam("pua", "update_period", 6 ) # I added this to see if the problem would be fixed modparam("pua_dialoginfo", "caller_confirmed", 1) I didn't capture the mi udp packets, but I did get the SIP, see below I think the problem has to do with expires_offset. I set this to reduce SUBSCRIBE expiration by 30 seconds, but it is also reducing PUBLISH expiration by 30 seconds. Do you think this might be the source of my problem? What else might be causing this ? Is there anyway to set the expires_offset for SUBSCRIBE but not for the PUBLISH? Any other ideas what might be going on here? Thanks, David Here is the SIP I captured on the pua server : U 2013/03/29 14:01:41.005025 PUBLISH sip:user161@omnity.net SIP/2.0. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK267e.e988a546.0. To: sip:user161@omnity.net. From: sip:user161@omnity.net;tag=047a42f14f0e800c0a21487d4bc09012-c8fe. CSeq: 10 PUBLISH. Call-ID: 79a5f23d217e3597-9109@proxyip. Content-Length: 0. Max-Forwards: 70. Event: dialog. Expires: 66. SIP-If-Match: a.1363343972.1019.567098.10. U 2013/03/29 14:01:41.298041 SIP/2.0 200 OK. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK267e.e988a546.0. To: sip:user161@mydomain.net;tag=f0253efb8005057b3cc0ab8fa35ff1d9-0dd9. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-c8fe. CSeq: 10 PUBLISH. Call-ID: 79a5f23d217e3597-9109@proxyip. Expires: 36. SIP-ETag: a.1363343972.1009.565433.11. Content-Length: 0. U 2013/03/29 14:03:14.317797 PUBLISH sip:user161@mydomain.net SIP/2.0. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK543f.efdb2d8.0. To: sip:user161@mydomain.net. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-fab2. CSeq: 10 PUBLISH. Call-ID: 79a5f23d214a91de-9057@proxyip. Content-Length: 271. Max-Forwards: 70. Event: dialog. Expires: 91. SIP-If-Match: a.1363343972.1009.565433.11. Content-Type: application/dialog-info+xml. . <?xml version='1.0'?> <dialog-info xmlns='urn:ietf:params:xml:ns:dialog-info' version='0' state='full' entity='user161(a)omnity.net'> <dialog id='x_dialogid(a)192.168.2.5' direction='recipient'> <state>early</state> </dialog> </dialog-info> U 2013/03/29 14:03:14.319691 SIP/2.0 412 Conditional request failed. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK543f.efdb2d8.0. To: sip:user161@mydomain.net;tag=f0253efb8005057b3cc0ab8fa35ff1d9-9add. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-fab2. CSeq: 10 PUBLISH. Call-ID: 79a5f23d214a91de-9057@proxyip. Server: OmniVigil 5.2-rc1. Content-Length: 0.

1 0

Rewrite Prefix in Carrieroute
by Ali Jawad 04 Apr '13

04 Apr '13

Hi I do have a server with kamailio running on carrireroute, recently the need to rewrite prefixes arose, so I did enter the rewrite prefix into the relevant rows/columns in carrierroute tdb table and did reload but nothing changed. The call is matched to it's needed destination but the prefix is the same. In my config I do have the following line if(!cr_route("default", "ata.nymgo.com", "$rU", "$rU", "call_id")){ Regards

1 0

New developer: Torrey Searle
by Daniel-Constantin Mierla 04 Apr '13

04 Apr '13

Hello, I want to announce that a new person got developer GIT write access to repository: Torrey Searle - he is going to add and maintain the sipt module, facilitating operations over SIP-T/SIP-I messages, the patch being on tracker: - http://sip-router.org/tracker/index.php?do=details&task_id=281&project=1 His git commit id is: tsearle My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users April 2013