I am working on a new call flow which involves doing a blindxfer of a call into a conference. When I do this, I get one way audio for that party. I have seen patches and posts about something similiar but nothing definitive and no real resolution.
Has anyone else run across this and was able to solve it.
I am using 1.6b9, SIP, G711.
I resolved the issue and wanted to post back about it. It might be a bug, but so obscure that no one else would care.
Here is the scenario:
I was placing call to 5551111 via pstn using a cisco 2600 from a sip phone (1000) using the g option to split to caller and callee in the dialplan after answer. After some prompting, they would be joined in a meetme conf. After a period of time, I would kick the callee out of the conf and then dial out to another number (5551234) on the pstn through the same 2600. Now 5551111 and 5551234 are talking. The 5551234 user would press the blindxfer key *11 and the 5551111 user would be joined back into the original conf with 1000.
Everything worked perfectly except when 55551234 pressed *11. When 5551111 was back in the conf, 1000 could not hear them. I traced and found that the rtp was being sent from 5551111 to the port of 5551234 on the cisco still even though 5551234 was hung up. This shouldn’t be happening since * should have been in the media path the entire time.
I was able to stop this behavior by setting canreinvite=no in sip.conf. This should have not been necessary since the T and t options tell asterisk to stay in the media path.