[Asterisk-Users] Codec Issue

[Aisling]

Hi,
 
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 9but 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.
 
I have included my sip.conf code and extensions.conf code below:
 
;sip.conf
 
[general]
bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
canreinvite=no;
autocreatepeer=yes
nat=yes
;dtmfmode=info
;dtmfmode=rfc2833
insecure=very
registerattempts=0
 
;context=default
 
register => username at providerIP/1234
 
;To make outgoing calls specify this block
 
[providerIP]
type=peer
user=phone
host=providerIP
port=6060
fromdomain=providerIP
fromuser=username
secret=password
username=username
insecure=very
context=incomingpstn
authname=username
allow=gsm
allow=ulaw
allow=alaw
;allow=g729                         ;NBNB This is where the issue is
 
[314]
type=friend
username=314
canreinvite=no
context=from-provider
insecure=very
host=dynamic
nat=yes
dtmfmode=rfc2833
mailbox=314
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
 
[2092]
type=friend
username=2092
canreinvite=no
context=from-provider
insecure=very
host=dynamic
nat=yes
dtmfmode=rfc2833
mailbox=2092
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
 
 
;extensions.conf
 
[general]
 
static=yes
writeprotect = yes
allow=alaw
 
;specify context for receiving incoming calls
 
[from-provider]
 
include => createmenu
include => createconf
include => joinconf
include => playvoicemail
;include => internalExt
;include => incomingpstn
include => default
 
[createmenu]
;Create an IVR Menu
exten => 20005,1,Wait(2)
exten => 20005,2,Record(/tmp/asterisk-recording:gsm)
exten => 20005,3,Wait(2)
exten => 20005,4,Playback(/tmp/asterisk-recording)
exten => 20005,5,wait(2)
exten => 20005,6,Hangup
 
[createconf]
;Create a conference call
exten => 20006,1,Wait(1)
exten => 20006,2,MeetMe(|MD)
exten => 20006,3,Hangup
 
[joinconf]
;Join a conference call
exten => 20007,1,Answer
exten => 20007,2,Wait(1)
exten => 20007,3,MeetMe(|P)
 
[playvoicemail]
;listen to voicemails
exten => 171,1,VoicemailMain(${CALLERIDNUM})
 
;Send PSTN calls to Provider
exten => _X.,1,Dial(SIP/${EXTEN}@ipaddressofprovider)
exten => _X.,2,Hangup
 
[default]
 
;voicemail
exten => 314, 1,Dial(SIP/314,20)
exten => 314, 2,Voicemail(u314)
exten => 314, 102,Voicemail(b314)
exten => 314, 103,Hangup
 
exten => 2092, 1,Dial(SIP/2092,20)
exten => 2092, 2,Voicemail(u2092)
exten => 2092, 102,Voicemail(b2092)
exten => 2092, 103,Hangup
 
[incomingpstn]
 
;The below two lines dial a particular extension 
exten => 4590124,1,Wait(1)
exten => 4590124,n,Dial(SIP/314 at ipaddressofser,20,r)
 
 
 
 
 
 


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