Sip to sip calls need to be put on hold to work!

Very strange problem. I’m installing 1.6 with dahdi 2.0. I finally got it installed and I can make incoming and outgoing calls. The problem is when I go to call from one sip phone to another the phone rings, the other person answers, and there is no voice. If I put the phone on hold (either way) then take them off hold the call goes through just fine.
Has anyone had a problem like this? If so what did you do to fix it.

I am just going to throw out a guess here… but it sounds to me like asterisk is taking itself out of the media path initially and having the two SIP UAs connect their media streams directly to each other.

When you request to put the call on hold, asterisk hops back in the middle to play the music on hold to the on agent. It probably doesn’t take itself out of the middle when you come off hold.

A good test of this would be to add “reinvite=no” and “canreinvite=no” to your sip.conf. If that fixes the problem, then you know that your two SIP devices are having trouble ‘seeing’ each other directly.

thanks for the reply. I tried it but no luck. I’ve turned sip debug on and all the handshaking seems to be there. I think your idea is on the right track, it seems to be ib\n the setup. I just can’t figure out where.