Russell Bryant wrote: > On Aug 11, 2008, at 12:04 PM, SIP wrote: > > >> SIP wrote: >> >>> When calling from our SIP proxy through Asterisk to the PSTN >>> provider, >>> we support reINVITES which tend to work seamlessly. >>> >>> However, when creating a call file which essentially connects a call >>> from the SIP proxy to the SIP proxy, Asterisk wants to stay in the >>> RTP >>> media path. I understand that this is sort of the idea behind a >>> bridged >>> channel, but is there any way to avoid it? Is there any way to say >>> "Connect this number and this number and then get out of the way," >>> or >>> is this a design limitation? >>> >>> >> No ideas on this one? I've tried everything I can think of and then >> some >> and still can't get Asterisk out of the media path. I can do it if I >> don't originate the call with Asterisk, but only use Asterisk to >> connect >> one leg of the call, but if I use Asterisk to connect both legs, no >> luck. >> >> Going about this the wrong way? >> > > > Asterisk will re-INVITE the media away from itself as long as it > doesn't have a reason to need access to the media. For example, if > you've enabled call recording, then clearly Asterisk needs access to > the media. Other reasons include enabling features controlled via > DTMF when the DTMF follows the media path. > > Nobody can help any further without seeing the details of your > configuration. > > -- > Russell Bryant > Senior Software Engineer > Open Source Team Lead > Digium, Inc. > > It's a rather simple config, really.
Peer: [vitel-termination] type=peer host=outbound1.vitelity.net username=myuser fromuser=myuser trustrpid=yes sendrpid=yes secret=mysecret allow=all canreinvite=yes Call file: Channel: SIP/[EMAIL PROTECTED] MaxRetries: 1 RetryTime: 10 WaitTime: 30 Application: AGI Data: /usr/local/click-to-call/dialtest.pl|1YYYYYYYYYY Callerid: XXXXXXXXXX The dialtest.pl just issues a dial to the 1YYYYYYYYYY number after Asterisk rings the 1XXXXXXXXXX number. Very basic, simple click-to-call stuff. You think maybe it's an issue with transcoding? I haven't tried forcing a single codec to see, but I'm pretty sure that Vitelity does g711 and g729. As we don't have any g729 licenses installed, would it not simply ignore those (not sure how Asterisk negotiates codecs in a situation like that, to be honest. It's always seemed like a bit too much jiggerypokery for my tastes) ? N. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
