[asterisk-users] My G729 problem re-visited

Power, Paul C. ppower at oneeighty.com
Fri Oct 12 14:35:09 CDT 2007


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: 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 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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