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 => [EMAIL PROTECTED]/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/[EMAIL PROTECTED]) 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/[EMAIL PROTECTED],20,r)
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