sr-users December 2009

sr-users@lists.kamailio.org

67 participants
96 discussions

[Kamailio-Users] log not showing when using uac_replace_from( ) function

by Robert R

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

14 years, 4 months

[Kamailio-Users] log not showing when using uac_replace_from( ) function

by Robert R

14 years, 4 months

[Kamailio-Users] Kamailio Perl OpenSER::AVP

by Örn Arnarson

Hello everyone, I've been successfully using the kamailio Perl module, but now I'm trying to set AVPs. Unsuccessfully. Whenever I do "use OpenSER::AVP;" and restart kamailio, the following error messages appear and kamailio doesn't start: Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:XS_OpenSER__Message_log: perl error: Can't locate OpenSER/AVP.pm in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.10.0 /usr/local/share/perl/5.10.0 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.10 /usr/share/perl/5.10 /usr/local/lib/site_perl .) at /etc/kamailio/checkip.pl line 4. Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:XS_OpenSER__Message_log: perl error: Can't locate OpenSER/AVP.pm in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.10.0 /usr/local/share/perl/5.10.0 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.10 /usr/share/perl/5.10 /usr/local/lib/site_perl .) at /etc/kamailio/checkip.pl line 4. BEGIN failed--compilation aborted at /etc/kamailio/checkip.pl line 4. Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:perl:parser_init: failed to load perl file "/etc/kamailio/checkip.pl". Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:init_mod: failed to initialize module perl Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:main: error while initializing modules Now, I cannot find this AVP module within the perl package anywhere. I downloaded the source of 1.5.0 and it's nowhere to be found. It is, however, in the .1.5 documentation. Does anyone have any info on this for me? Best regards, Örn

14 years, 4 months

[Kamailio-Users] Using Sip Message in a third application.

by Olivier Détour

Hello, I'm using Kamailio 1.5.3. I configured it in a proxy mode. I would like to know what is the best way to forward the SIP message to a C application via a socket, block kamailio.cfg script, and reforward it to kamailio to continue his treatment. I found a ugly bay using PV as lock with a while loop, but this solution is not acceptable for me. I saw SEAS module, but it is only for Java Application server. Do you know a simple way to do it with an existing module ? Do I have to write my own module ? Thanks for your response, Regards, -- Olivier Détour EPITA 2009

14 years, 4 months

topoh: remove Via headers

by Klaus Darilion

Hi! IIRC correctly. topoh modules rewrites existing Via headers. Would it be possible to remove existing headers completely so that the forwarded request has only one Via header? regards klaus

14 years, 4 months

[Kamailio-Users] Integrate Skype into Kamailio (Must support video)

by General Lee

Hi, I would like to integrate Skype into my Kamailio video system? Has anyone implemented this successfully before? Or suggest how this could be done? Key features I would like are: 1. Must support Video 2. Ideally support H264 video codec 3. Share presence information 4. Ability to make calls in both directions I would appreciate any information you could provide. Thanks, -- View this message in context: http://old.nabble.com/Integrate-Skype-into-Kamailio-%28Must-support-video%2… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

14 years, 4 months

[Kamailio-Users] ERROR:tm:add_uac: maximum number of branches exceeded

by alex pappas

Hello there, I'm using dispatcher to serial fork a call through many gateways. For a call to number 2XXXXXXXXX , first a try to connect through gateways A,B,C after that D,F , after that E,R , after that M,N and finally after those K,L. When the call is trying gateway N then I'm getting the ERROR:tm:add_uac: maximum number of branches exceeded. I found on internet that I'm exceeding the MAX_BRANCHES: branch=t->nr_of_outgoings; if (branch==MAX_BRANCHES) { LOG(L_ERR, "ERROR:tm:add_uac: maximum number of branches exceeded\n"); ret=E_CFG; goto error; } >From a trace I got , I see that kamailio is sending to each destination IP 8 INVITES and if there is no answer to that is continuing with next destination gateway sending also there 8 INVITES. When the number of INVITES is 52 then I get this error. Does anyone can suggest a way to change the MAX_BRANCHES MAX number ? Is is important to keep open the INVITEs from the first gateways because is an LCR scenario and if I get an answer from these gateways is important to complete through them. Any help is apreciated Thanks Alexandros

14 years, 4 months

[Kamailio-Users] Random looped back audio issues with RTPPROXY

by tony＠everestbn.com

Hello, I'm having random audio issues (<10% of the time) with blind call transfers on a Polycom phone. According to RTPPROXY's logs, it seems it is substituting the same IP and port for the caller's and callee's address causing one end to hear their own voice while the other end hears nothing. INFO:rxmit_packets: caller's address filled in: a.b.c.199:17856 (RTP) INFO:rxmit_packets: guessing RTCP port for caller to be 17857 INFO:rxmit_packets: callee's address filled in: a.b.c.199:17856 (RTP) INFO:rxmit_packets: guessing RTCP port for callee to be 17857 The SDP in the SIP messages has all the correct IPs and ports and it appears from a SIP overview everything should work fine. Here is some background information: I am using Kamailio 1.5.3 with RTPPROXY 1.2.1. SIP PATH is as follows: PSTN Gateway---Kamailio--SIP Proxy Server--Polycom Phone RTP PATH: PSTN Gateway---Kamailio----Polycom Phone *There are nor firewalls or filters. Call Flow: I place a call from the gateway to the ip phone. Call is answered. When I do a blind transfer from the Polycom, the SIP Proxy sends an invite with no SDP, the gateway responds with a 200 with SDP and the SIP Proxy then sends an ACK with SDP. As I mentioned, this works most of the time and the SIP messages look identical for when the transfer works and when it doesn't. When it doesn't work the gateway end hears their own voice while the recipient Polycom transfer (usually another phone on the same proxy) hears nothing. Here is a portion of my configuration file: ..... if (is_method("INVITE")) { if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))){ rtpproxy_offer("fcr"); setflag(12); } } if (is_method("ACK")) { if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))) { rtpproxy_answer("fcr"); } } t_on_reply("1"); .... } onreply_route[1] { if (status=~"(180)|(183)|(2[0-9][0-9])"){ if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))) { if (isflagset(12)) { rtpproxy_answer("fcr"); } else { rtpproxy_offer("fcrl"); } } } } Any help would be greatly appreciated! Thanks, Tony

14 years, 4 months

Re: [Kamailio-Users] [sr-dev] sr and kamailio autogenerated tarballs

by Daniel-Constantin Mierla

On 12/2/09 1:50 PM, Andrei Pelinescu-Onciul wrote: > Daily snapshots of sr_3.0, kamailio_3.0 and sr master branch can be > downloaded from: http://sip-router.org/tarballs/sr/ . > Note that new tarballs are generated only if there are changes for the > corresponding branch (so if you don't see a tarball with today's date it > means there was no change from the previous one). > > The tarballs are generated via make tar, which makes sure the correct > repository version will be included in the compiled binary (make tar > generates first autover.h before creating the archive). > Thanks Andrei! I cc-ed Kamailio users mailing list, since there might be people interested as well. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

14 years, 4 months

[Serusers] Memory leak issue

by Piyush.Bansal＠relianceada.com

Hi All, I'm using SER version 2.0.0-rc1 as a SIP server and in that facing lot of memory leaks issue. I'm not able to enable the memory logs in SER. Can anybody help me identifying memory leaks or tell me how to enable memlogs so that I myself will try removing all the leaks... Plz help me ASAP, its urgent... Thanx n regards, --Piyush The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

14 years, 4 months

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sr-users December 2009