sr-users April 2006

sr-users@lists.kamailio.org

188 participants
431 discussions

by Bogdan-Andrei Iancu

Hi all, looks like the anonymous cvs outage on SF lasts for some time and there are no good perspective :(. As we try to offer public access to the latest openser code, we switched the snapshots from the anonymous cvs to the developers cvs. The snapshots for the development branch will be generated 6 times per days. Please use them instead of pserver. http://www.openser.org/downloads/snapshots/openser-devel/ in the mean while we investigate the possibility of setting a non-SF anonymous CVS as backup - still searching on synchronizing the CVS (if someone has any experience on that, help will be really appreciated). sorry for the inconvenience! regards, bogdan

18 years, 7 months

[Users] Using two ser

by jvr_78＠yahoo.com.ar

Hi, I want use 2 ser, a.com and b.com When an user from a.com wants to call to an user from b.com, the messages are sent directly to b.com, and a.com doesn't realise about that. what can I do to make all messages coming from an user from a.com go ever through a.com? regards Javier Ramirez

18 years, 7 months

[Users] Coupling with SEMS?

by Rick van Rein

Hello, I am trying to get SEMS coupled to OpenSER. Is the coupling between OpenSER and SEMS lost? Can anyone tell me how to properly direct SEMS' reply_sockect_name to an OpenSER server? Thanks, -Rick

18 years, 7 months

[Users] avp question?

by Frogger

How can I lookup the correct username when routing a call where the domain of the TO needs to match the domain of the FROM. For example: 100(a)foo.bar needs to contact 200(a)foo.bar but there is also 200(a)newfoo.bar. I was thinking of avp lookup where I return the correct username of "200" where domain = "foo.bar". Am I on the right track here? If so, any starter hints would be greatly appreciated. F __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 7 months

Re: [Serusers] setflag question

by Dave

Thanks. But what is it associated with. When I do a register and determine that it is NAT'd client, I do a setflag. When I do this, what is this setflag associated with? With the location entry of the registered client? And when I us setflag when I receive an INVITE, is it associated with that INVITE pdu? --- Steve Blair <blairs(a)isc.upenn.edu> wrote: > > It is a memory resident flag. > > Dave wrote: > > >Hello, I do no understand where the FLAG is set > when > >using setflag. Is the SIP packet itself flagged (by > >inserting a header?) or is it something that is > done > >in the database? Thank you for any pointers > >Dave > > > >__________________________________________________ > >Do You Yahoo!? > >Tired of spam? Yahoo! Mail has the best spam > protection around > >http://mail.yahoo.com > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > -- > > ISC Network Engineering > The University of Pennsylvania > 3401 Walnut Street, Suite 221A > Philadelphia, PA 19104 > > > voice: 215-573-8396 > > 215-746-8001 > > fax: 215-898-9348 > > sip:blairs@net.isc.upenn.edu > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 7 months

[Serusers] Caller id Change

by Allan Chao \( 策略性新技術開發處申請 \)

Dear all: I have a question, if called user wants to display caller ID number, it's depend on FROM header (ex: From:1234 <sip5555(a)192.168.1.13> (1234 is caller id number , 5555 is gw number that called user will display 1234 on the sip phone) ),anyone know how to change the "1234" number which we wanted. Allan DISCLAIMER: Sample Disclaimer added in a VBScript.

18 years, 7 months

[Users] Problem Integrating OPENSER with ASTERISK

by fidiasm＠ctbctelecom.net.br

Hi there, I´m facing a problem with my voip callings. I use OPENSER as a proxy and an ASTERISK as the server which deals with all the SIP MESSAGES (including REGISTER) and is conected to PSTN. This solution works fine when an user tries to call to PSTN or any service on asterisk (i.e. VoiceMail), but if an user tries to call another user it´s not possible. It gives an error 500 - Server Internal Failure. Can anyone help me solve this problem ?? PS1 - Both servers are in the internet PS2 - I tried using both users with valid IP´s (not using nat) and the problem is the same. PS3 - I´m attaching 2 files, one with my openser configuration and another with debug of a call from 1234 to 4321 (See attached file: openser-debug.txt)(See attached file: openser.cfg.txt) Fídias Augusto

18 years, 7 months

[Serusers] setflag question

by Dave

Hello, I do no understand where the FLAG is set when using setflag. Is the SIP packet itself flagged (by inserting a header?) or is it something that is done in the database? Thank you for any pointers Dave __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 7 months

[Serusers] /var/run/ser.pid does not exist ser start failed.

by ravi reddy

hello dear sir/s recently i had installed astersik and it is running well and one week back i installed sip express router i made it installed well i hope so but when i start in /etc/init.d = serctl start iam getting this fault i dont know how to solve this problem can any one please help me thank you' regards, ravi.

18 years, 7 months

[Serusers] Fw: 400 Bad Request after an ACK

by Rosario Pingaro

Can someone help me to debug my problem? I have ser between asterisk and my clients. When I try to call a sip client, it is going to ring. But asap the callee pickup the phone the call goes down. Doing a logging on port 5060 i see that after the ack i get 400 bad request from the sip client. This is the trace: Session Initiation Protocol Request-Line: ACK sip:0681140017@83.211.248.158:62746 SIP/2.0 Method: ACK Resent Packet: False Message Header Record-Route: <sip:19x.6x.19x.4x;ftag=as6d07dd0a;lr=on> Via: SIP/2.0/UDP 19x.6x.19x.4x;branch=0 Via: SIP/2.0/UDP 19x.24x.16x.9x:5060;branch=z9hG4bK368ed369;rport=5060 From: "anonymous" <sip:asterisk@voip.convergenze.it>;tag=as6d07dd0a SIP Display info: "anonymous" SIP from address: sip:asterisk@voip.convergenze.it SIP tag: as6d07dd0a To: <sip:08281895109@voip.convergenze.it>;tag=b1385811e50f0aai1 SIP to address: sip:08281895109@voip.convergenze.it SIP tag: b1385811e50f0aai1 Contact: <sip:asterisk@19x.24x.16x.9x> Contact Binding: <sip:asterisk@19x.24x.16x.9x> URI: <sip:asterisk@19x.24x.16x.9x> SIP contact address: sip:asterisk@19x.24x.16x.9x Call-ID: 3ff0ae307575f1df61408b205af01196(a)voip.convergenze.it CSeq: 102 ACK User-Agent: Convergenze VoGW1 Max-Forwards: 16 Content-Length: 0 and then Session Initiation Protocol Status-Line: SIP/2.0 400 Bad Request Status-Code: 400 Resent Packet: False Message Header To: <sip:08281895109@voip.convergenze.it>;tag=b1385811e50f0aai1 SIP to address: sip:08281895109@voip.convergenze.it SIP tag: b1385811e50f0aai1 From: "anonymous" <sip:asterisk@voip.convergenze.it>;tag=as6d07dd0a SIP Display info: "anonymous" SIP from address: sip:asterisk@voip.convergenze.it SIP tag: as6d07dd0a Call-ID: 3ff0ae307575f1df61408b205af01196(a)voip.convergenze.it CSeq: 103 INVITE Via: SIP/2.0/UDP 19x.6x.19x.4x;branch=z9hG4bK9f79.7854d9e5.0 Via: SIP/2.0/UDP 19x.24x.16x.9x:5060;branch=z9hG4bK493c6be2;rport=5060 Record-Route: <sip:19x.6x.19x.4x;ftag=as6d07dd0a;lr=on> Server: Sipura/SPA2100-3.2.5(d) Content-Length: 0 Any help is appreciated. Regards Rosairio

18 years, 7 months

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sr-users April 2006