[Asterisk-Dev] SIP rx issues

[Rob Gagnon]

Are both phones behind the same NAT router/firewall and both using SIP port
5060?

If so, it won't work unless each device is on a different port since the
router will map the port/ip combination to only 1 device at a time.

Rob

----- Original Message ----- 
From: "Ben Kramer" <ben at voicetronix.com.au>
To: <asterisk-dev at lists.digium.com>
Sent: Tuesday, March 30, 2004 10:08 PM
Subject: RE: [Asterisk-Dev] SIP rx issues


>
> 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
> >
> > _______________________________________________
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> -- 
> Ben Kramer <ben at voicetronix.com.au>
>
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