Re: [asterisk-users] re-invite (bypass asterisk) post call establishment

2008-04-22 Thread Steve Davies
2008/4/22 Benjamin Jacob [EMAIL PROTECTED]: [snip] So, my question : once the SDPs are exchanged, what will happen to the DTMFs sent by Asterisk using sendDTMF or the D option in dial. [snip] As far as I can tell, the D() option will be executed before the re-invite takes place, so Asterisk

Re: [asterisk-users] re-invite (bypass asterisk) post call establishment

2008-04-22 Thread Benjamin Jacob
Hi again, I tried this again, but the reInvite happens immediately after the 200 OK/ACK. And then the D() specified DTMF is sent. Attached is the SIP trace for the calls. I call (from Asterisk) - 0119198807x After connect, I dial - 31927x. This number 31927x is the conference

Re: [asterisk-users] re-invite (bypass asterisk) post call establishment

2008-04-21 Thread Steve Davies
On 21/04/2008, Benjamin Jacob [EMAIL PROTECTED] wrote: Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after

Re: [asterisk-users] re-invite (bypass asterisk) post call establishment

2008-04-21 Thread Benjamin Jacob
Apologies for not explaining the set up . Using AstMan API, I Originate a call to user A. User A is a conference bridge which needs pin authentication. So post 200 OK, I need to send DTMFs for that pin. After sending the pin, I Dial (using the Originate context) user B. Now user B is behind