sr-users July 2007

sr-users@lists.kamailio.org

163 participants
262 discussions

Re: [Serusers] How SER communicate to other SIP server
by roayu 03 Jul '07

03 Jul '07

Hi, Olaf! I've used 'rewritehostport("sipB.net:5060")' under the INVITE message handler. The following are my configuration, pls correct me if I'm wrong. if(uri =~ "^sip:[0-9{8}@.*") { if(!proxy_authorise("","subscriber")){ proxy_challenge("","0"); break; }else if(!check_from()) { sl_send_reply("403","Use From ID"); break; }; consume_credentials(); rewritehostport("sipB.net:5060"); setflag(1); route(1); break; } When I dialed 8-digit of the number, it will pass to another SIP server(sipB.net). But on the TO header, I still found that the TO address still under the sipSER.net. You may refer to the following. From : <sip:12345@sipSER.net:5060> To : <sip:88888888@sipSER.net:5060> Does that means that it's still looping inside sipSER.net instead of routing to sipB.net? If yes, how do I change the TO header or make it route to sipB.net? Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Monday, July 02, 2007 4:55 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > I'm not sure what functions that I can use to include inside ser.cfg. Could > you pls let me know where I can see all those functions like > sl_send_reply(), so that I can use them. Have a look at <http://www.iptel.org/doc>. You will find several tutorials as well as reference documents, especially for the available modules. In addition, there is an excellent interactive search tool at <http://www.iptel.org/ser/doc/search> to lookup function definitions etc. Regards, Olaf

2 1

[Serusers] PA module SER 2_0_0 - no xcap query with auth parameter set to xcap
by tzieleniewski 03 Jul '07

03 Jul '07

Hi, I am trying to use the xcap authorization for PA during the SUBSCRITPION message servicing. I have the following code which work fine when auth parameter is set to 'none': if(lookup_user("$tu.uid", "@ruri")) { if (!t_newtran()) sl_reply_error(); drop; } (handle_subscription("registrar")) ... } but when I switch to auth 'xcap' I cant see any query send to XCAP server simulated on apache according to the presence handbook. Please point me what do I miss. Thanks Cheers Tomasz

2 7

[Serusers] xl_mod_init warning
by Todd Lewis 02 Jul '07

02 Jul '07

I using the example configuration file that comes with ser-2.0-rc1 No matter what I do, I cannot figure out what's causing this warning /usr/local/ser/sbin/ser[43254]: Maxfwd module- initializing /usr/local/ser/sbin/ser[43254]: WARNING: xl_mod_init: more IP 10.0.0.6 not used I have two ip's assigned to the server, using one Nic card Static IP 10.0.0.6 Virtual IP 10.0.0.4 NicCard em0 I have tried -------------------- listen=10.0.0.4 fork=yes/no --------------------- -------------------- listen-10.0.0.6 fork=yes/no -------------------- -------------------- listen=10.0.0.4 listen=10.0.0.6 fork=yes/no -------------------- -------------------- listen=em0 listen=10.0.0.4 fork=yes/no -------------------- -------------------- listen=em0 fork=yes/no -------------------- I always get the same error message. Any ideas?

1 0

Re: [OpenSER-Users] FreeRADIUS-CDRTool Rel. 1.0
by Dan-Cristian Bogos 02 Jul '07

02 Jul '07

Hi, my code has no connection to any BBUA. It is just providing the maximum call duration information back to OpenSER which can later do what u want it to do with this info. I personally use yate (yate.null.ro) to provide call cut when the calls are going out on PSTN, by adding an extra SIP Header in OpenSER and then later process it in Yate to timeout the calls. I guess there are unlimited possibilities later, if u have the information available in OpenSER AVPs, including creating your own timer module which sends a session cancelation order to RTPProxy (heard that it is possible to kill the calls with RTPProxy, but never done it). Let me know if I was clear enough. Cheers, DanB On 7/2/07, A.M. <agentm(a)themactionfaction.com> wrote: > > On Jul 2, 2007, at 11:10 , Dan-Cristian Bogos wrote: > > > All, > > > > as a personal gratitude for all the efforts involved in opensource > > projects, especially in OpenSER community, I would like to contribute > > myself with a bit of code, investing in something I considered useful > > for this community. > > > > Hereby, I inform u that the first release of FreeRADIUS-CDRTool > > connector module is available for download and use at sourceforge.net. > > Project link: http://sourceforge.net/projects/frad-cdrtool/ > > > > Basically, FreeRADIUS-CDRTool is a FreeRADIUS module written in python > > (therefore one might need to compile rlm_python in FreeRADIUS to make > > it work), able to connect on CDRTool Engine via the Telnet socket (the > > only way I found recommended by CDRTool folks) and perform different > > actions via Prepaid API, like return maximum value for the call, > > instruct CDRTool to lock accounts already in a call, instruct > > unlocking at end of the call and balance debiting, etc. This module > > should be used together with OpenSER radius authentication and > > accounting. > > If proper configured, FreeRADIUS-CDRTool can authorize calls for > > CDRTool Prepaid users. At the starting of each call the module will > > return the maximum call duration allowed, information which can be > > processed later in a bbua scenario to timeout the calls reaching 0 > > balance, and total credit available at starting of the call, all > > through SIP-AVP Radius Attributes. > > Hi- thanks for the information. How does your code interact with the > b2bua? Which b2buas are supported? > > Cheers, > M >

1 0

[OpenSER-Users] FreeRADIUS-CDRTool Rel. 1.0
by Dan-Cristian Bogos 02 Jul '07

02 Jul '07

All, as a personal gratitude for all the efforts involved in opensource projects, especially in OpenSER community, I would like to contribute myself with a bit of code, investing in something I considered useful for this community. Hereby, I inform u that the first release of FreeRADIUS-CDRTool connector module is available for download and use at sourceforge.net. Project link: http://sourceforge.net/projects/frad-cdrtool/ Basically, FreeRADIUS-CDRTool is a FreeRADIUS module written in python (therefore one might need to compile rlm_python in FreeRADIUS to make it work), able to connect on CDRTool Engine via the Telnet socket (the only way I found recommended by CDRTool folks) and perform different actions via Prepaid API, like return maximum value for the call, instruct CDRTool to lock accounts already in a call, instruct unlocking at end of the call and balance debiting, etc. This module should be used together with OpenSER radius authentication and accounting. If proper configured, FreeRADIUS-CDRTool can authorize calls for CDRTool Prepaid users. At the starting of each call the module will return the maximum call duration allowed, information which can be processed later in a bbua scenario to timeout the calls reaching 0 balance, and total credit available at starting of the call, all through SIP-AVP Radius Attributes. I have tested the module in a Debian Linux environment, python2.4 and FreeRADIUS 1.6, CDRTool version 5.1.4, on low call load. I would not recommend yet using it in a production environment till we get some stability feedback. Have fun! Kind regards, DanB

1 0

[OpenSER-Users] mediaproxy StopRecordSerializer when closing channel
by Dzejms 02 Jul '07

02 Jul '07

Hi, I have this common problem Jul 2 16:03:25 localhost mediaproxy[5566]: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected xxx.xxx.xxx.xxx:46948 at -0x5f92e794> (exceptions.NameError:global name 'StopRecordSerializer' is not defined [asyncore.py|readwrite|94] [asyncore.py|handle_read_event|391] [/usr/local/mediaproxy/modules/rtphandler.py|handle_read|441] [/usr/local/mediaproxy/modules/rtphandler.py|process|569]) Mediaproxy version is 1.8.2 it's running on debian 4.0 etch, witch python 2.4 (also tried 2.5), i have python-mysqldb module installed, also python-dev-all . Problem still exist. When i'm using 1.7.2 version od mediaproxy, it seems to be ok. I'm also using 1.8.2 mediaproxy version on debian sarge with python 2.3 where everything seems to be ok. Any ideas? Broken debian packages? Dzejms.

1 0

[OpenSER-Users] manipulate INVITES
by Patrick Miccio 02 Jul '07

02 Jul '07

Hello everybody, since this is my first post and I am kinda new to OpenSER, bare with me ;) I have been trying to manipulate invites for the last 2 days and cant find the proper way to reach what is expected. my goal: send invites to specific PBX (Asterisk for example) with only the desired extension and not the full username, this is based on a avp_check if that UAC needs only extensions, some do, some don't ;) Here is what I do: First off all I extract the real username (in subscribers) and extension from the R-URI and place it in "$avp(callee_user)" and "$avp(callee_exten)" ############# my solution No1 is "working" avp_pushto("$ru/username", "$avp(callee_user)"); avp_copy("$avp(callee_exten)","$avp(callee_tmp)/gd"); if( subst('/^To:(.*)sip:[^@]*(@[a-zA-Z0-9.]+.*)$/To:\1sip:$avp(callee_tmp)\2/ig') ){}; -> callee_user is 012345678 -> callee_exten is 200 -> $ru is sip:012345678@xxx.xxx.xxx.xxx;user=phone -> subst [ To: <sip:012345678200@xxx.xxx.xxx.xxx;user=phone> ] with [To: <sip:200@xxx.xxx.xxx.xxx;user=phone> ] Those 3 lines work "fine", the testing Asterisk PBX gets the call correctly. The problem is that I should not change the "To:" part, it will make problems with call forwards etc... ############# my solution No.2 NOT working. if (subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(callee_tmp)@\2;/i')){$}; doesn't work because OpenSER of course cant find 200 in the userloc.... ############# Generally I would think that changing the INVITE uri itself like in solution No.2 is the best way, couldn't find any proper way to make that work though. Does anyone have an idea how to realize this properly, or even if this makes sense ;) ? thx in advance, Patrick.

2 4

[OpenSER-Users] help with using avp...
by liran tal 02 Jul '07

02 Jul '07

Hey everyone, I'm using OpenSER 1.1.1 SVN version. I'd like to save the Authorization Digest Username as it appears in header into a variable (avp) so that later I can use it to set an accounting attribute in the radius. >From the SIP Packet the information is given like this: Authorization: Digest username="dsdsds", realm="1.2.3.4", algorithm=MD5, uri="sip:1.2.3.4", nonce="xxx" Where I would like to grab the username part "dsdsds". >From the docs I could only conclude that doing it would be something like this: avp_write("$hdr[Authorization]","s:12"); avp_subst("$avp(s:12)", "/.*username=(.*),/\1/"); I'm not even sure about that piece of code, if someone can help... Thanks, Lir.

2 1

[Serusers] RLS module in ser 2_0_0 - subscriptions handling
by tzieleniewski 02 Jul '07

02 Jul '07

Hi, I am using the RLS module for RLS service and resource list subscriptions. Ive done everything according to the presence handbook. When SER sends NOTIFY messages to the subscribing UAs every instances of the subscribed resources is in the pending state. How are those subscriptions handled by ser?? Do I have to include presence_b2b molude so that those subscriptions in the pending state would be handled?? Thanks in advance Cheers Tomasz

2 3

[Serusers] routing rtpstream through nat within vpn
by Martin Drasar 02 Jul '07

02 Jul '07

Hello everyone, I have following scenario: +----------+ | VPN | |+--------+| || MRCP || |+--------+| | || | |+--------+| +---------+ || Proxy ====| Solaris | |+--------+| +---------+ +----------+ I want both sip messages and rtp stream to go through proxy between mrcp servers and solaris box. Currently I am transferring sip messages using call forwarding (avpops module) and trying to use rtpproxy for rtp stream forwarding. For rtpproxy to work I have set up NAT on proxy, but still have issues. This is the invite packet sent from proxy to MRCP server: INVITE sip:UAS@10.1.1.100:5060 SIP/2.0 Record-Route: <sip:10.0.0.1;ftag=3d423348;lr=on> <=## Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKe359.5e010b95.0 <=## Via: SIP/2.0/UDP 10.0.0.6:5070;branch=z9hG4bK-d8754z-5212be016a2d3c2a-1---d8754z-;rport=5070 Max-Forwards: 16 Contact: <sip:sender@10.0.0.6:5070> To: <sip:synthesizer@10.0.0.1:5060;transport=UDP> <=## From: <sip:sender@10.0.0.6:5070>;tag=3d423348 Call-ID: MTZkYWE2MGYxNTgyNmY0YTRmNzkwNjAyNmU5OGQ0ZGU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY Content-Type: application/sdp Content-Length: 282 v=0 o=- 1183369050 1183369050 IN IP4 10.0.0.6 s=- t=0 0 m=application 35050 TCP/MRCPv2 <=## c=IN IP4 10.0.0.1 <=## a=connection:new a=setup:active a=cmid:1 a=resource:speechsynth m=audio 53384 RTP/AVP 0 c=IN IP4 10.0.0.1 <=## a=mid:1 a=recvonly a=direction:active a=nortpproxy:yes Lines with "<=##" are the lines with which I think I have problem. Because mrcp servers receive 'To' and 'Via' with address that is outside vpn, they cannot reply. As a result I send x times invite and end with 408 timeout. Also there is problem with line "m=application 35050 TCP/MRCPv2". rtpproxy changes that port, but according to mrcp specification I need it to be exactly 9. Even when changing PORT_MIN and PORT_MAX both to 9, I sometimes get port 10. Could you please advice me what to do? Am I trying in right direction? Will I have to alter the sources to make it work like I need to? Thanks for your answers Martin

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 July 2007