How many licenses you have (show g729 should give you this info)
Scott Moseman wrote: > 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. <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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 >>>> > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users