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:[EMAIL PROTECTED]>;tag=as520d008c
To: <sip:[EMAIL PROTECTED]>;tag=1c282937849
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]
> Sent: Thursday, December 07, 2006 12:17 PM
> To: [EMAIL PROTECTED]; 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:[EMAIL PROTECTED]
> 
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