Thanks for the reply, Johansson. Sorry if my question was not very clear... What I need is that asterisk accepts a REFER command from the client, sending the call to a non local domain. The scenario is this: I receive a call from PSTN and dial a sip address that contains one of my applications (running in a separate machine). This application receives input from the user and then transfers the call to another application (in a third machine). The call from PSTN is going to be in asterisk (that got the call in first place) all the time, just the other end will change depending on user input. Bellow is a sip debug from this operation. Asterisk is running in 201.73.67.5:5060 and my first application is at "[EMAIL PROTECTED]:5080". This application then tries to transfer the call to a second application located at "[EMAIL PROTECTED]:5070", but asterisk ignores the part after the "@" from the uri and tries sending the call to the extension 5070 in the context "from-sip-external". I had a similar problem with redirects (302), but I solved it using the option promiscredir=yes inside "sip.conf". I've already tried setting the option "domain=" in sip.conf but that didn't help...
<-- SIP read from 201.73.67.7:5080: REFER sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 201.73.67.7:5080;rport;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx Max-Forwards: 70 From: <sip:[EMAIL PROTECTED]>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA To: "3130296800" <sip:[EMAIL PROTECTED]>;tag=as26b5df58 Contact: <sip:201.73.67.7:5080> Call-ID: [EMAIL PROTECTED] CSeq: 15651 REFER Event: refer Expires: 300 Accept: message/sipfrag;version=2.0 Allow-Events: presence, refer Refer-To: sip:[EMAIL PROTECTED]:5070 Referred-By: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- (15 headers 0 lines) --- Transfer to 5070 in from-sip-external Transfer from 0778 in from-sip-external Transmitting (no NAT) to 201.73.67.7:5080: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 201.73.67.7:5080;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx;received=201.73.67.7;rport=5080 From: <sip:[EMAIL PROTECTED]>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA To: "3130296800" <sip:[EMAIL PROTECTED]>;tag=as26b5df58 Call-ID: [EMAIL PROTECTED] CSeq: 15651 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing Thiago > > 13 apr 2008 kl. 17.46 skrev <[EMAIL PROTECTED]>: > > I made a similar question in a previous thread, but there was no > > answer, so I think I was not very clear making the question. What > I > > need is some configuration that works like "promiscredir=yes" in > > sip.conf that enables me to do the same thing with transfer > (REFER), > > letting me transfer a sip call to a non local sip address. > > > I'm still not really sure what you ask for, but I'll give it a try. > > The transfer() dialplan application supports generating a REFER > from > Asterisk to the client. If the call is not answered, it will send > 302, > if the call is in UP state (answered), Asterisk will send a REFER. > Try > it. > > Best regards, > /Olle > > > --- > * Olle E. Johansson - [EMAIL PROTECTED] > * Asterisk Training http://edvina.net/training/ > * Asterisk SIP Masterclass, Orlando, Florida Next week > * A few seats left - register today! > > Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
