I got this working on 1.2.9 and 1.4.11. We use Polycom phones with and OpenSER proxy and AudioCodes gateway to the PSTN world. Progressinband is set to never.
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Chris Ziomkowski > Sent: Wednesday, October 03, 2007 1:27 AM > To: [email protected] > Subject: Re: [asterisk-dev] Why does chan_sip disable early > bridging??? > > Not sure how that would work, given that the code below > specifically prohibits it. What version are you running? This > is the latest code from 1.4.11. I have never used the "r" > option in Dial as I want to hear remote ringback. I never > want to genererate it locally. > > For some reason, SIP channels are specifically prohibited for > early media except in the case where you have specified > Direct RTP according to the code below. (which is confusing > in an of itself because if you are using direct rtp then you > don't need Asterisk to do media routing, early or otherwise, do you?) > > In any case, the line in chan_sip.c appears to be incorrect > and I have corrected it in my version. I would hope the > maintainers can look into this and either explain what the > original purpose of the statement was, or else remove it if > it is in fact a bug. > > > Power, Paul C. wrote: > > >I got early media to work for me by omitting the 'r' option > in the dial > >statement. > > > > > > > >>-----Original Message----- > >>From: [EMAIL PROTECTED] > >>[mailto:[EMAIL PROTECTED] On Behalf Of Chris > >>Ziomkowski > >>Sent: Monday, October 01, 2007 4:58 AM > >>To: [email protected] > >>Subject: [asterisk-dev] Why does chan_sip disable early bridging??? > >> > >>OK everyone, > >> > >>I have been banging my head against a wall for 2 days > trying to figure > >>out why I can't get ringback on my SIP to SIP calls through > Asterisk. > >>Eventually, I tracked it down. The call to ast_early_bridge > actually > >>completes without error, but the problem appears in chan_sip, in > >>sip_set_rtp_peer() at or about line 17204 in the 1.4.11 version of > >>Asterisk: > >> > >>if (chan->_state != AST_STATE_UP && !global_directrtpsetup) > /* We > >>are in early state */ > >> return 0; > >> > >>Apparently, any attempt at early media over SIP is thwarted in the > >>chan_sip module. After commenting out this line, my calls bridge > >>fine and I hear the ringback and other progress messages. I > have not > >>yet noticed any ill effects from doing this. > >> > >>My question is: what was the purpose of this statement? Is there > >>another/better way to get early media on SIP to SIP calls? > >> > >>Thanks in advance, > >> > >>Chris > >> > >> > >> > >> > >>_______________________________________________ > >> > >>Sign up now for AstriCon 2007! September 25-28th. > >>http://www.astricon.net/ > >> > >>--Bandwidth and Colocation Provided by http://www.api-digital.com-- > >> > >>asterisk-dev mailing list > >>To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-dev > >> > >> > >> > > > > > > > > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
