[Asterisk-Users] Codec negotation with re-invites..

[Billy Huddleston]

Okay, I add allow=g729 into the [general] section of sip.conf...

I can now place calls via ulaw or g729 without any problems..  simply by
setting the allow= in the phone's sip entry..

However, INBOUND is a whole nother problem...
I get a really strange "buzz" sound on inbound calls.. and...  here is a
snippit of show sip channels while the call is in progress..

1.1.1.24   8659342199  505b634c5cc  00103/00000  00000ms  0000ms  ULAW
1.1.1.29   8656914260  B392830B-17  00102/00102  00000ms  0000ms  G729A

.24 is the sip phone, .29 is the gateway

I'm totally lost on this.


----- Original Message ----- 
From: "Alex Volkov" <avolkov at bpvn.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, March 12, 2004 3:23 PM
Subject: Re: [Asterisk-Users] Codec negotation with re-invites..


> Sounds to me that your asterisk first negotiates g729 with your phone,
then
> negotiates ulaw with the gateway (since it *is* the preferred codec in
your
> config), and on a re-invite the logic breaks up either in the phone or in
> the gateway (or perhaps in the asterisk itself, I am not absolutely clear
on
> the details of re-invites). Try changing the order of codec preference for
> the gateway and see if that fixes your g729 phone and breaks the ulaw
phone
> at the same time.
>
> Alex.
>
> ----- Original Message -----
> From: "Billy Huddleston" <billy at nxs.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, March 12, 2004 2:02 PM
> Subject: [Asterisk-Users] Codec negotation with re-invites..
>
>
> > I'm about over this.. okay,, here is what I got..
> >
> > [general]
> > port = 5060                     ; Port to bind to
> > bindaddr = 0.0.0.0              ; Address to bind to
> > context = inbound                       ; Default for incoming calls
> > tos=lowdelay
> > tos=184
> > disallow=all                    ; Disallow all codecs
> > allow=ulaw
> >
> > [gateway]
> > type=friend
> > host=1.1.6.9
> > canreinvite=yes
> > qualify=yes
> > dtmfmode=rfc2833
> > context=default
> > disallow=all
> > allow=ulaw
> > allow=g729
> >
> > [sipphoneg729]
> > type=friend
> > secret=password
> > nat=yes
> > host=dynamic
> > canreinvite=yes
> > qualify=200
> > context=longdistance-g729
> > dtmfmode=rfc2833
> > mailbox=2199
> > disallow=all
> > allow=g729
> >
> > [sipphoneulaw]
> > type=friend
> > secret=password
> > nat=yes
> > host=dynamic
> > canreinvite=yes
> > qualify=200
> > context=longdistance
> > dtmfmode=rfc2833
> > mailbox=2199
> > disallow=all
> > allow=ulaw
> >
> >
> > okay, when I place a call from sipphoneulaw to the outside world via
> > gateway, everything works fine..
> > If I place a call from sipphoneg729, it doesn't work..  One leg to the
> > gateway will be ulaw, the leg to the phone will be g729, and, I have 1
way
> > audio.. The sip phone can hear anything from the gateway, but, the
gateway
> > can't hear the phone.
> >
> > I've even went as far as to setup a seperate context for the g729 phone
> and
> > do this..
> > ,SetVar,SIP_CODEC=g729  which, says it sets it to g729, but it's still a
> > ulaw call..  Guys, this is a real problem... We're going be doing mixed
> > configs.. and if a gateway says it can do both, and phone says it can
only
> > do one... then we should be using the compatable codec...  PLEASE help..
> > This is going to cause problems in our rollout.
> >
> > Thanks, Billy
> >
> >
> >              +--------------------------------------------------+
> >              | Billy Huddleston   Senior Systems Administrator  |
> >              | Net-Express                  http://www.nxs.net  |
> >              | 114 Sherway Rd.             Voice: 865-691-2011  |
> >              | Knoxville, TN  37922          Fax: 865-691-9894  |
> >              | billy at nxs.net                                    |
> >              +--------------------------------------------------+
> >
> >
>
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