On Mar 16, 2006, at 3:24 AM, Aisling wrote:
<x-tad-smaller>Hi everyone,</x-tad-smaller>
<x-tad-smaller> </x-tad-smaller>
<x-tad-smaller>I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don’t work. – Strange.</x-tad-smaller>
<x-tad-smaller> </x-tad-smaller>
<x-tad-smaller>Anyhow I was getting an error:</x-tad-smaller>
<x-tad-smaller> </x-tad-smaller>
<x-tad-smaller>Process_sdp: No compatible codecs!</x-tad-smaller>
<x-tad-smaller>And from the SIP debug I could see that the incoming SIP INVITE was getting a sip response of 488 Unacceptable here from my asterisk server.</x-tad-smaller>
<x-tad-smaller> </x-tad-smaller>
<x-tad-smaller>After doing a bit of searching I determined that this might be the fault of the codec’s particularly the G729 codec. So in the peer block that I have for my PSTN provider in my sip conf I specified allow=g729.</x-tad-smaller>
<x-tad-smaller>I called my PSTN geographic number again and was delighted when the incoming calls worked. However when I next went to make an outgoing call (after having added in the “allow=g729” line), I got an infinite loop of warnings:</x-tad-smaller>
<x-tad-smaller> </x-tad-smaller>
<x-tad-smaller>WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)</x-tad-smaller>
<x-tad-smaller>WARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isn’t a multiple of 33 or 65 bytes long from RTP</x-tad-smaller>
<x-tad-smaller> </x-tad-smaller>
<x-tad-smaller>After those warnings I thought there might be a problem with the gsm codec so I commented the lines containing “allow=gsm” and still kept the line “allow=g729” because as I’ve said already incoming calls won’t work otherwise (but outgoing will).</x-tad-smaller>
<x-tad-smaller>This however just gave another warning:</x-tad-smaller>
<x-tad-smaller> </x-tad-smaller>
<x-tad-smaller>WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 4 while native formats is 256 (read/write=64/64).</x-tad-smaller>
<x-tad-smaller>When I comment this line out again I am back to my original situation where outgoing calls work and incoming don’t.</x-tad-smaller>
<x-tad-smaller> </x-tad-smaller>
<x-tad-smaller>Has anyone any idea how I can work around this?</x-tad-smaller>
<x-tad-smaller>I think telling us which type of gateway is between asterisk and the PSTN might be helpful in this case...
</x-tad-smaller>
Marty
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
