[Asterisk-Dev] SIP rx issues

[Ben Kramer]

Hiya,

both phones and the asterisk server are on the same network, and there
is no NAT'ing involved.

Cheers,

Ben.

On Thu, 2004-04-01 at 00:21, Rob Gagnon wrote:
> 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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> >
> 
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-- 
Ben Kramer <ben at voicetronix.com.au>