[Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)

[Martin Joseph]

On Mar 16, 2006, at 3:24 AM, Aisling wrote:

> Hi everyone,
>  
> 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.
>  
> Anyhow I was getting an error:
>  
> Process_sdp: No compatible codecs!
> 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.
>  
> 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.
> 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:
>  
> WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 256, 
> while native formats is 8 (read/write = 8/8)
> WARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isn’t 
> a multiple of 33 or 65 bytes long from RTP
>  
> 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).
> This however just gave another warning:
>  
> WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 4 
> while native formats is 256 (read/write=64/64).
> When I comment this line out again I am back to my original situation 
> where outgoing calls work and incoming don’t.
>  
> Has anyone any idea how I can work around this?
>  
I think telling us which type of gateway is between asterisk and the 
PSTN might be helpful in this case...

Marty

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