sr-users

sr-users@lists.kamailio.org

20 participants
37471 discussions

[Kamailio-Users] LCR
by Kris Amy 10 Feb '10

10 Feb '10

Hi All, I've been ripping my hair out for a little with load_gws(). How do I know what the lcr_id is and the caller_uri? I would have thought that $fu for caller_uri would have been sufficient but it seems not. Cheers, Kris

3 2

[Kamailio-Users] kamctl and lcr table
by Kris Amy 10 Feb '10

10 Feb '10

Hi, Could we get the lcr-create.sql updated to have the ping field? Cheers, Kris

2 3

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 09 Feb '10

09 Feb '10

Hello, I was thinking of having an IRC meeting to discuss next steps within sip router. Now that 3.0 branches are out for a while, would be good to plan a bit 3.1. For me Friday (or even Thursday if not too early for others to make arrangements) this week would be good, say 14:00 or 15:00UTC to allow fair time for US. If this date does not fit, please make proposals. The channel is #sip-router on freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 * http://www.asipto.com/

1 0

[Kamailio-Users] problems with mi_xmlrpc command
by Anton Roman 09 Feb '10

09 Feb '10

Hi all, I'm trying to send xmlrpc commands to Kamailio with Siremis but I get the error below: Fault Code: 5 Fault Reason: Didn't receive 200 OK from remote server. (HTTP/1.1 400 Bad Request) I get the same error when I manually send the xmlrpc command with curl. On the other hand, mi_xmlrpc module seems to be correctly loaded, but nothing is written in the log file (set with modparam("mi_xmlrpc", "log_file", "/var/log/abyss.log")). Does anybody know where the problem might be? Best regards, thanks in advance. Anton Roman These are the traces of the xmlrpc message flow: T +11.569284 127.0.0.1:37821 -> 127.0.0.1:8080 [AP] POST HTTP/1.0 User-Agent: PEAR XML_RPC Host: 127.0.0.1 Content-Type: text/xml Content-Length: 135 <?xml version="1.0" encoding="UTF-8"?> <methodCall> <methodName>system.listMethods</methodName> <params> </params> </methodCall> ## T +0.000363 127.0.0.1:8080 -> 127.0.0.1:37821 [AP] HTTP/1.1 400 Bad Request # T +0.000146 127.0.0.1:8080 -> 127.0.0.1:37821 [AP] Content-type: text/html # T +0.001982 127.0.0.1:8080 -> 127.0.0.1:37821 [AFP] Connection: close Date: Mon, 08 Feb 2010 14:04:08 GMT Server: XMLRPC_ABYSS/1.06 <HTML><HEAD><TITLE>Error 400</TITLE></HEAD><BODY><H1>Error 400</H1><P>Bad Request</P><p><HR><b><i><a href="http://xmlrpc-c.sourceforge.net">ABYSS Web Server for XML-RPC For C/C++</a></i></b> version 1.06<br></p></BODY></HTML> ####

2 2

[Kamailio-Users] Stateless forward
by Antonio Reale 09 Feb '10

09 Feb '10

Hi all, I'm trying to setup a very basic scenario with kamailio 3.0. It is placed in front of a softswitch and it has to modify some headers before forward (stateless) messages to softswitch. If I use route { rewritehost("192.168.10.1"); t_relay(); } all work, but if I want to do that stateless and I use forward() or send() instead of t_relay(), I get a syntax error: "bad forward argument" or "bad send argument". If I use ... forward("udp:192.168.10.1:5060"); ... I see in the log /usr/local/sbin/kamailio[3675]: : <core> [proxy.c:271]: ERROR: mk_proxy: could not resolve hostname: "udp:192.168.10.1:5060" /usr/local/sbin/kamailio[3675]: ERROR: <core> [route.c:1026]: fixing failed (code=-478) at cfg:/usr/local/etc/kamailio/kamailio.cfg:77 Same error using the long format of send() function. I can't see any particular hint on the documentation, what I'm missing? Thank you. Best regards. -- Antonio

3 6

[Kamailio-Users] Install Documentation
by Kris Amy 09 Feb '10

09 Feb '10

Hi All, Just thought that you might want to put down that libpcre is required for lcr in the install document. (Since the other main modules have their requirements listed). Cheers, Kris

2 1

[Kamailio-Users] change domain at to header
by Adnan Çelikel 09 Feb '10

09 Feb '10

Dear All, I want to change a field (Sip to address host part) at sip message header. I think avpops will solve this.If anybody has a sample confg and send to me I will be very glad.Regards,Adnan. #---------------------------------------------------- #7777 CALL ROUTING #-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-* if(is_method("INVITE") && (uri=~"sip:7777")){ rewritehostport("192.168.1.10:5060"); route(1); exit; } #---------------------------------------------------- at this rule, the invite is going to 192.168.1.10 with the host part as abc.com, we want to send with another domain. Instead of abc.com domain, I want to use 192.168.1.10 domain ; To:<sip:7777@abc.com> To:<sip:7777@192.168.1.10> _________________________________________________________________ Windows Live Hotmail: Arkadaşlarınız Facebook'taki güncellemelerinizi doğrudan Hotmail®'den alır. http://www.microsoft.com/windows/windowslive/see-it-in-action/social-networ…

5 11

Routing Problems
by Andreas Rehbein 08 Feb '10

08 Feb '10

Hello, we use Kamailio 3.0 on a Red Hat EL5.4 system with openssl 0.9.8e (the current Red Hat OpenSSL version). We want to use Kamailio 3.0 in front of our Call Server (OpenSER 1.3.2) as a security gateway. So the Call Server should only deal tcp, while the Security Gateway terminates the TLS Data which he receives from the User Agents but forwards the SIP-Messages via TCP to the Call Server. UA ---SIP_over_TLS---> Security Gateway (Kam3.0) ---SIP_over_TCP---> Call Server (OpenSER1.3.2) UA<---SIP_over_TLS--- Security Gateway (Kam3.0) <---SIP_over_TCP--- Call Server (OpenSER1.3.2) Nearly everything works fine in this scenario: User Agents are able to register and when they are sending INVITES the callee receives it. The problem we have right now is: if Kamailio 3.0 receives BYE we get 477 Unfortunatly error on sending to next hop occured. It seems that the messages are ok... I attached a text file with the BYE message and the errors. Explanation for the text file: * UA1: 192.168.0.126 * UA2: 192.168.0.176 *Security Gateway (Kam3.0): 192.168.0.89 *Call Server (OpenSER1.3.2): 192.168.0.106 Do you have any suggestions? Thank you very much in advance! Regards Andreas Dipl.-Ing. (FH) Andreas Rehbein FH Frankfurt am Main - University of Applied Sciences FB 2 - Informatik und Ingenieurwissenschaften Forschungsgruppe für Telekommunikationsnetze Kleiststraße 3 60318 Frankfurt am Main Tel.: +49 (0) 69 1533 3613 Fax: +49 (0) 69 1539 5224 961 E-Mail: <mailto:rehbein@e-technik.org> rehbein(a)e-technik.org Internet: <http://www.e-technik.org/frame_forschung.htm> http://www.e-technik.org/frame_forschung.htm

2 2

[Kamailio-Users] Maximum retries exceeded on transmission
by Daniel-Constantin Mierla 06 Feb '10

06 Feb '10

On Sat, Feb 6, 2010 at 11:10 AM, Jon Farmer <viperdudeuk(a)gmail.com> wrote: > Sorry it was a problem I had with gmail. > It happens to me from time to time, but I couldn't find the real cause, whether is gmail or thunderbird. Thunderbird hangs a while, reports a delivery error, then message appears twice... Daniel > On 5 Feb 2010 12:21, "Iñaki Baz Castillo" <ibc(a)aliax.net> wrote: > > 2010/2/5 Jon Farmer <viperdudeuk(a)gmail.com>: > > Hi > > Please don't send the same smail twice. Thanks. > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Maximum retries exceeded on transmission
by Daniel-Constantin Mierla 06 Feb '10

06 Feb '10

Hello, On Fri, Feb 5, 2010 at 10:56 AM, Jon Farmer <viperdudeuk(a)gmail.com> wrote: > Hi > > I am getting this dreaded error on a asterisk box when call from one > UA to another using Asterisk as a media relay and openser as a SBC and > proxy. > > chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission > 1310650592(a)10.100.0.101 > > If I make a call from a UA to the PSTN through Asterisk all is > fine. This only happens when calling from one UA to another on the > same LAN. > > I have posted an example sip trace here http://pastebin.com/m2262b0ef > ACK is not accepted as I could spot quickly, 200ok being retransmitted. Probably there is something wrong with dialog attributes, after Fosdem I can check it better. Daniel > My openser.cfg is here http://pastebin.com/d5e84aaf7 > > I realise this is a issue with Asterisk but I have not been able to > find a fix. Therefore I figure there might be something I can do on > openser. The problem seems to be with a ACK not being returned but my > SIP knowledge is not good enough to know how to fix it. Any help is > greatly appreciated. > > Regards > > Jon > -- > Jon Farmer > Tel: 07795 118140 > Email: viperdudeuk(a)gmail.com > Twitter: @viperdudeuk > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

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