[asterisk-users] Session Progress Transmission to Phone

Douglas Garstang dgarstang at oneeighty.com
Thu Dec 7 13:23:23 MST 2006


Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone.
Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone.

<-- SIP read from xxx.yyy.142.234:5060: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xxx.yyy.142.232:5060;branch=z9hG4bK675a9b71;rport=5060
From: "OneEighty Communications" <sip:4062944000 at xxx.oneeighty.com>;tag=as520d008c
To: <sip:112944087 at proxy3.xxx.oneeighty.com>;tag=1c282937849
Call-ID: 66006e8624237aef7bcc48c76bd6d0b0 at xxx.oneeighty.com
CSeq: 102 INVITE
Contact: <sip:5011 at xxxyyy.142.190>
Record-Route: <sip:xxx.yyy.142.234;lr=on;ftag=as520d008c>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.4.80A.027.002
Content-Type: application/sdp
Content-Length: 237

v=0
o=AudiocodesGW 283013199 283012901 IN IP4 216.187.142.190
s=Phone-Call
c=IN IP4 216.187.142.190
t=0 0
m=audio 6350 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

Doug.


> -----Original Message-----
> From: Csibra Gergo [mailto:gergo at csibra.hu]
> Sent: Thursday, December 07, 2006 12:17 PM
> To: gsalas at manta.telconet.net; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [asterisk-users] FXO USB that works with Asterisk?
> 
> 
> Thursday, December 7, 2006, 7:40:09 PM, Guillermo Salas M. wrote:
> 
> > Please, make a new message for a new question, do not reply a thread
> > with a different topic, and finally, use english.
> 
> Quoting a whole mail with headers and footers is bad too...
> 
> -- 
> Best regards,
>  Csibra Gergo                            mailto:gergo at csibra.hu
> 
> _______________________________________________
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