[asterisk-users] My G729 problem re-visited

Power, Paul C. ppower at oneeighty.com
Mon Oct 15 13:30:01 CDT 2007


Have you figured out if asterisk is crashing or not? 

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Scott Moseman
> Sent: Friday, October 12, 2007 2:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] My G729 problem re-visited
> 
> Gateway sends Asterisk an INVITE (using g729) Asterisk sends 
> Phone an INVITE (using g711 or g729) Phone sends Asterisk an 
> OK (using g711) Asterisk sends Gateway an OK (with no RTP 
> choice) Gateways ends the conversation
> 
> I can setup the Phone to use g729 and it will reply with an 
> OK for g729, but the OK to the Gateway will still stay empty. 
>  Only when I enable g711 on the Gateway will this work.  I 
> have experienced this on
> 2 different models of gateways so far.
> 
> I included my config for both the Gateway and the Phone in my 
> original message, hoping that maybe I was configuring the 
> Gateway wrong in Asterisk?  But no one has said anything so 
> I'm assuming its okay.
> 
> Phone (g729) to Phone (g729) works
> Phone (anything) to Gateway (g711) works Phone (anything) to 
> Gateway (g729) does NOT work
> 
> I'm licensed for g729 (although I'm told I should not need it 
> for pass through).  And it will transcode when the phone is 
> g729 and the gateway is g711.  But for whatever reason I 
> can't use g729 on the gateway side of the calling process?
> 
> Thanks,
> Scott
> 
> 
> 
> On 10/12/07, Power, Paul C. <ppower at oneeighty.com> wrote:
> >
> > Is the call being dropped or is Asterisk takng a core dump?
> >
> > I have core dump issues with g729 and asterisk 1.4.11, but 
> my set up 
> > is a little different than yours...
> >
> >
> > > -----Original Message-----
> > > From: Scott Moseman
> > > Sent: Friday, October 12, 2007 10:22 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] My G729 problem re-visited
> > >
> > > No ideas on this one from anyone?  I suppose I'm going to need to 
> > > pay for some Digium support because this is a really unusual 
> > > problem.
> > > Does anyone else have a gateway that speaks g729 to Asterisk and 
> > > works?  For whatever reason, Asterisk refuses to reply 
> back to any 
> > > of my gateways using g729.  Phone (g729) to phone
> > > (g729) works.  Phone
> > > (g729) to Asterisk to gateway (g711) works.  But attempt g729 
> > > between Asterisk and a gateway and it fails -- every time.
> > > Asterisk responds to the gateway but never includes any codecs in 
> > > the packet, unless it's g711.  My configurations are shown below.
> > >
> > > Thanks,
> > > Scott
> > >
> > >
> > > On 9/26/07, Scott Moseman <scmoseman at gmail.com> wrote:
> > > >
> > > > Ok, I built a test system to duplicate my problem and
> > > provide myself a
> > > > platform that I can mess around with to try and break 
> any features.
> > > > My problem is G729 pass-through from a gateway to a phone.
> > > I think I
> > > > even have transcoding working, which makes me more confused
> > > on what's
> > > > wrong with my pass-through. It must be a configuration issue.
> > > >
> > > > The basics...
> > > >
> > > > *CLI> core show version
> > > > Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 
> running Linux
> > > >
> > > > *CLI> show modules like 723
> > > > Module Description Use Count
> > > > codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 
> > > > Simple Timestamp File Format 0
> > > >
> > > > *CLI> show modules like 729
> > > > Module Description Use Count
> > > > codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw 
> G729 data 0
> > > >
> > > > *CLI> show translation
> > > > [truncated]
> > > > g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex
> > > ilbc g726 g722
> > > > ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 -
> > > > g729 5 2 2 2 2 2 1 3 - - 11 2 -
> > > >
> > > > The configuration...
> > > >
> > > > [gateway]
> > > > type=friend
> > > > host=gateway
> > > > context=default-inbound
> > > > disallow=all
> > > > allow=g729
> > > >
> > > > [phone]
> > > > type=friend
> > > > context=sip
> > > > host=dynamic
> > > > username=phone
> > > > secret=scott
> > > > dtmfmode=RFC2833
> > > > disallow=all
> > > > allow=g729
> > > > callerid=Scott
> > > > qualify=yes
> > > > canreinvite=no
> > > >
> > > > exten => 1266,1,Dial(SIP/[number],30,t) exten => 
> 1266,2,Congestion
> > > >
> > > > exten => 1266,1,Dial(SIP/[number],30) exten => 1266,2,Congestion
> > > >
> > > > (The same results using both of the above dialplans...)
> > > >
> > > > The environment...
> > > >
> > > > PSTN -> Gateway -> Asterisk -> Phone
> > > >
> > > > What I'm seeing works...
> > > >
> > > > With the gateway setup to send both G711 and G729, it sends
> > > an INVITE
> > > > which includes both G711 and G729 codecs. Asterisk sends an
> > > INVITE to
> > > > my phone with only G729. The call is made and there's a
> > > conversation
> > > > in G711 with the gateway and G729 with the phone. I assume
> > > this means
> > > > Asterisk is transcoding.
> > > >
> > > > What I"m seeing fails...
> > > >
> > > > With the gateway setup to send only G729, it sends an INVITE to 
> > > > Asterisk which includes only G729. Asterisk send an 
> INVITE to the 
> > > > phone using G729, too. The 200 OK from the phone to the 
> Asterisk 
> > > > includes G729. The 200 OK going from Asterisk to the
> > > gateway doesn't
> > > > include ANY codec. The call is dropped the moment I pickup
> > > the phone
> > > > to answer the call.
> > > >
> > > > My question...
> > > >
> > > > Why does Asterisk not want to respond to my gateway in G729?
> > > > Even if the gateway requests it, Asterisk seems to just 
> ignore it.
> > > > From the transcoding call, and phone to phone G729 calls, I
> > > have proof
> > > > that Asterisk knows how to handle G729 calls.
> > > >
> > > > Where do I go from here???
> > > >
> > > > Thanks,
> > > > Scott
> > > >
> 
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