asterisk users - Mar 2006 - Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)

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?
 
Many thanks in advance,
Aisling.
 


-------------------Legal  Disclaimer---------------------------------------

The above electronic mail transmission is confidential and intended only for the
person to whom it is addressed. Its contents may be protected by legal and/or
professional privilege. Should it be received by you in error please contact the
sender at the above quoted email address. Any unauthorised form of reproduction
of this message is strictly prohibited. The Institute does not guarantee the
security of any information electronically transmitted and is not liable if the
information contained in this communication is not a proper and complete record
of the message as transmitted by the sender nor for any delay in its receipt.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/a07dcb48/attachment.htm

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

-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: text/enriched
Size: 5014 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/8f81c6ee/attachment.bin