sr-users February 2007

sr-users@lists.kamailio.org

158 participants
286 discussions

[Serusers] Rewriting To: field
by Stéphane Alnet 23 Feb '07

23 Feb '07

Hello, I am doing interop testing with Level3, and need to be able to rewrite the To: field in requests to match their requirements. I can fake the rewrites inside route[], but although the packets sent out look like they use the new "To:", the header is never parsed again, so the internal structures of SER still hold the former (incorrect) value. When a request hit failure_route[] (for example when a CANCEL is sent when an INVITE times out) the original, incorrect To: header is then sent (and there's no way to modify the content of the CANCEL from within failure_route[]). Is there a standard way to alter the internally-stored value for To: (and From:, since I'm asking)? If not, could someone point me to the right place in the code to start adding this kind of feature? Thanks, Stéphane Alnet -- http://carrierclass.net/ http://www.linkedin.com/in/stephalnet

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RE: [Users] OpenSER Configuration Generator
by Watkins, Bradley 22 Feb '07

22 Feb '07

Hrmmm... I consistently get a Rails error when I get to the config_create stage. This is on both a Linux and Windows machine (a dedicated T1 and a proxied DS3, respectively for connection) using Firefox. Any ideas? - Brad > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Andreas Granig > Sent: Thursday, February 22, 2007 9:46 AM > To: users(a)openser.org > Subject: [Users] OpenSER Configuration Generator > > All, > > I've finished the first version of sip:wizard, an online > SER/OpenSER configuration generator: http://www.sipwise.com/wizard > > Currently, support for OpenSER 1.1 is available. In the > future, also new OpenSER releases as well as SER will be supported. > > At the moment, we have config features for basic routing, > far-end NAT traversal, ENUM and PSTN in place. This will also > be extended, depending on suggestions from you, the users. > > I hope you'll find it useful. Feedback is of course very much > appreciated. > > Cheers, > Andreas > > > This e-mail is confidential and may well also be legally > privileged. If you have received it in error, you are on > notice of its status. Please notify us immediately by reply > e-mail and then delete this message from your system. Please > do not copy it or use it for any purposes, or disclose its > contents to any other person: to do so could be a breach of > confidence. Thank you for your cooperation. > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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[Users] OpenSER Configuration Generator
by Andreas Granig 22 Feb '07

22 Feb '07

All, I've finished the first version of sip:wizard, an online SER/OpenSER configuration generator: http://www.sipwise.com/wizard Currently, support for OpenSER 1.1 is available. In the future, also new OpenSER releases as well as SER will be supported. At the moment, we have config features for basic routing, far-end NAT traversal, ENUM and PSTN in place. This will also be extended, depending on suggestions from you, the users. I hope you'll find it useful. Feedback is of course very much appreciated. Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation.

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[Users] status codes
by Ibrahim Hamouda 22 Feb '07

22 Feb '07

Running openser 1.2 failure_route[1] xlog("L_INFO", "status codes are [$rs] [$rr]\n"); gives "status codes are [<null>] [<null>] any ideas? Ibrahim Hamouda Sphinx Information Technologies Inc. IT Canada International Tel: 403-668-6880 Fax: 403-229-0407 N.A.: 1-877-500-7664 Mobile: 403-714-3336

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[Serusers] Bug in 0.9.6? International characters in Contact header
by Roger Lewau 22 Feb '07

22 Feb '07

Today I discovered that SER does not respond well to international chars in contact header of at least 200 OK. The 200 OK message had the below contact header. The two dots are the swedish chars ä and ö. Contact: Hampus H.ggl.f <sip:330039@217.168.167.xyz:5061>. The error logged from SER was: Feb 22 09:22:27 sip /usr/local/sbin/ser[40800]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: mk_proxy: could not resolve hostname: "Hampus" Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: uri2proxy: bad host name in URI <sip:Hampus> Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: t_forward_nonack: failure to add branches The UA is a PAP2-EU device running latest firmware. The text Hampus H.ggl.f was entered in the display name field of PAP2. Is this an error with SER or is it illegal to use international characters in Contact header? Kind regards Roger

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[Users] Active sessions - dialog
by Michel Bensoussan 22 Feb '07

22 Feb '07

Hello I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization according to codec, and allow or disallow new sessions, depending on resources. On a first stage I just want to limit the session to a fix number. For that, for each new session request (INVITE), I need to know how many active session there is already. I think using the module dialog but it is experimental. What is exactly its status? Is it usable for now? Is there another way to get information about the number of active sessions? Thanks. Regards, Michel

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[Users] moved temporarily - different port, same host
by Benko 22 Feb '07

22 Feb '07

Hello! I'm having a issue with a 302 moved temporarily from a UA i want to attach to my proxy. The UA replies to a INVITE from the proxy with a 302 MOVED TEMPORARILY and i'm using "get_redirects" from the uac_redirect-module to extract the contact. However, i don't know how to continue from there as the domain part in the redirect-contact is the hostname of the proxy but the port is 5065 and transport=TCP (see debug for details) and if i simply relay it, it would be relayed back to the proxy on the wrong port 5065. I guess i could use the request branch pseudo variable($br) and extract the port from it(or change the domain-part to the actual domain/ip of the UA and relay it), still i don't really know how to do it. This is the sip-trace of the relevant dialog part: 2xx.xxx.xxx.x06 is the openser-host, 6x.xxx.xxx.x93 is the useragent ###################################### #### T 2xx.xxx.xxx.x06:56669 -> 6x.xxx.xxx.x93:5060 [AP] INVITE sip:4989111111@proxydomain.org SIP/2.0. Record-Route: <sip:2xx.xxx.xxx.x06;transport=tcp;r2=on;lr;ftag=as784d1ec2>. Record-Route: <sip:2xx.xxx.xxx.x06;r2=on;lr;ftag=as784d1ec2>. Via: SIP/2.0/TCP 2xx.xxx.xxx.x06;branch=z9hG4bK96db.2c3c0006.0. Via: SIP/2.0/UDP 2xx.xxx.xxx.97:5060;branch=z9hG4bK52fef574;rport=5060. From: "SIPPHONE1" <sip:089222222@otherproxydomain.org>;tag=as784d1ec2. To: <sip:4989111111@proxydomain.org>. Contact:<sip:089222222@21x.xxx.xxx.97>. Call-ID:7fc122590665124f347fb2d956ba4c07@sip.at.telgo.cc. CSeq: 102 INVITE. User-Agent: MyMediaGW1. Max-Forwards: 69. Date: Wed, 21 Feb 2007 13:02:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 492. . v=0. o=root 14334 14334 IN IP4 2xx.xxx.xxx.97. s=session. c=IN IP4 2xx.xxx.xxx.97. t=0 0. m=audio 10380 RTP/AVP 0 8 18 4 3 97 111 5 10 7 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:111 G726-32/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:10 L16/8000. a=rtpmap:7 LPC/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - - . # T 6x.xxx.xxx.x93:5060 -> 2xx.xxx.xxx.x06:56669 [AP] SIP/2.0 100 Trying. FROM: "SIPPHONE1"<sip:089222222@otherproxydomain.org>;tag=as784d1ec2. TO: <sip:4989111111@proxydomain.org>. CSEQ: 102 INVITE. CALL-ID: 7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org. MAX-FORWARDS: 70. VIA: SIP/2.0/TCP 2xx.xxx.xxx.x6;branch=z9hG4bK96db.2c3c0006.0,SIP/2.0/UDP 2xx.xxx.xxx.97:5060;branch=z9hG4bK52fef574;rport=5060. CONTENT-LENGTH: 0. . ## T 6x.xxx.xxx.x93:5060 -> 2xx.xxx.xxx.x06:56669 [AP] SIP/2.0 302 Moved Temporarily. FROM: "SIPPHONE1"<sip:089222222@otherproxydomain.org>;tag=as784d1ec2. TO: <sip:4989111111@proxydomain.org>;tag=b3addacf4f. CSEQ: 102 INVITE. CALL-ID: 7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org. MAX-FORWARDS: 70. VIA: SIP/2.0/TCP 2xx.xxx.xxx.x06;branch=z9hG4bK96db.2c3c0006.0,SIP/2.0/UDP 213.208.4.97:5060;branch=z9hG4bK52fef574;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. CONTACT: <sip:4989111111@proxydomain.org:5065;transport=TCP>. . ## T 2xx.xxx.xxx.x06:56669 -> 6x.xxx.xxx.x93:5060 [AP] ACK sip:4989111111@proxydomain.org SIP/2.0. Via: SIP/2.0/TCP 213.208.4.106;branch=z9hG4bK96db.2c3c0006.0. From: "SIPPHONE1" <sip:089222222@otherproxydomain.org>;tag=as784d1ec2. Call-ID: 7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org. TO:<sip:4989111111@proxydomain.org>;tag=b3addacf4f. CSeq: 102 ACK. User-Agent: OpenSer (1.1.0-tls (i386/linux)). Content-Length: 0. . ###################################### Either i'm lacking understanding of the openser functionalities and sip redirect in general and simply don't see how this could be accomplished easier or i guess i need to grab "sip:4989111111@proxydomain.org:5065;transport=TCP" and extract the right port and protocol from it to relay it correctly(don't know how to do that though), i'm not sure. Currently i'm using a simple t_relay("tcp:6x.xxx.xxx.x93:5065"); to get it working, but i'd like to implement it in a more generic way. thx for all answers! best regards christian

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[Serusers] URI Pattern Matching
by Jon Farmer 22 Feb '07

22 Feb '07

Hi I am trying to find a way to detect if the Invite URI starts with a *. How do I match this as ser normally sees * as a wildcard. Regards Jon -- Jon Farmer Telford, Shropshire, UK

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[Users] Install on OpenBSD: cannot start openser
by Ben Ferguson - backcountry.com 22 Feb '07

22 Feb '07

Hello all, I've just installed openSER on OpenBSD and I'm now trying to get openSER to start and am having trouble. I try openserctl start and here's what i get: # openserctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded Starting OpenSER : \E[37;31mERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed # After a bunch of reading it seems as if openserctl is now obsolete (???) and that you should use the init script. I can't find reference for anyone running it like this on OpenBSD. I have found openser.init file in /<src directory>/packaging/gentoo and /<src directory>/packaging/rpm, but i am not sure how to use them on OpenBSD (I'm rather new to OpenBSD... I'm used to the rpm style of things.) I've searched and searched but can't find any info... So. anybody out there using openSER on OpenBSD? Or anybody know howto (or whereto find the info to) set up the start script for openSER on OpenBSD? Any help is greatly appreciated! Thanks, Ben

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RE: [Users] call never reaches the failure_route - Please help
by vallimamod abdullah 22 Feb '07

22 Feb '07

On 2/22/07, "Ibrahim Hamouda" <ihamouda(a)itcanint.net> wrote: > > There is already a xlog entry in the beginning of the failure route, and no > it doesn't reach it. > > > Any other ideas? Have you tried to put the 't_on_failure("1")' statement to the main route (instead of the invite route) ? -- Regards, -vma .

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sr-users February 2007