[Asterisk-Dev] SIP rx issues

[woody+asterisk at solutionsfirst.com.au]

> -----Original Message-----
> From: asterisk-dev-admin at lists.digium.com 
> [mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of Ben Kramer
> Sent: Wednesday, 31 March 2004 13:50
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] SIP rx issues
> 
> Hiya,
> 
> I have been working on the chan_vpb.c channel driver and was 
> testing the
> SIP functionality. What I discovered was that I was not able to get in
> audio from the SIP phone, but I could hear the other party. Just to
> confuse matters this didnt happen all the time.
> To confirm it was a SIP issue, I made a SIP->SIP call via asterisk. I
> had no audio path what so ever. I then tested the two phones, 
> directly,
> and it worked fine.
> I am basicly new to SIP to any ideas/pointers would be appreciated. I
> have looked through the last couple of months of the mailing list
> archives and cant find any mention of my issues. I am using the latest
> development branch of Asterisk CVS.

The SIP protocol uses a control connection and an audio (RTP) connection for
each direction.  It seems that the audio connection is getting from asterisk
to the SIP phones, but not in the other direction.

Mostly this is caused by firewalls or NAT boxes, can you post details of any
between the SIP phones and the asterisk box?

Also it would be handy to post the relevant (changed) parts of your
sip.conf.

Cheers,
Woody