[Asterisk-Dev] SIP rx issues

Ben Kramer ben at voicetronix.com.au
Tue Mar 30 21:08:28 MST 2004


Hiya,

here are the details:
* everything is on the same network
* adelsip is a bugetone101 SIP phone
* adelsip3 is a SipToneII SIP phone

[adelsip]
type=friend
host=dynamic
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=192.168.3.50
context=local

[adelsip3]
type=friend
host=dynamic
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=192.168.3.52
context=local

Every thing else in the sip.conf file is the default.

Cheers,

ben.

On Wed, 2004-03-31 at 13:32, woody+asterisk at solutionsfirst.com.au wrote:
> > -----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
> 
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
-- 
Ben Kramer <ben at voicetronix.com.au>




More information about the asterisk-dev mailing list