I’ve asterisk 1.4.15 and sip trtunk to my provider. I Can send fax from my asterisk box anywhere with T.38 . However when I send fax from PSTN to my asterisk there is problem. Here is sip messages:
myprovider-------------------------------------------------------------- Asterisk
invite(in SDP m=audio)------------------------------------------->
100 trying<---------------------------------------------------------------
180 Riging<--------------------------------------------------------------
200 OK<-------------------------------------------------------------------
ACK------------------------------------------------------------------------>
invite(in SDP m=image 17900 udptl t38)----------------->
##########################################
INVITE
sip:473196@XXXX:5060 SIP/2.0
Via: SIP/2.0/UDP YYYY:5060;x-route-tag="tgrp:AZ"
From: sip:922125@YYYY;tag=C79C7B0C-FB6
To: sip:473196@XXXX;tag=as4e6a4f02
Date: Tue, 15 Apr 2008 14:22:01 GMT
Call-ID: 15AF6127-A2E11DD-BEA0FBB2-B1FB486C@YYY
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 363735103-170791389-3198024626-2986035308
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 102 INVITE
Max-Forwards: 6
Remote-Party-ID: sip:922125@YYYY;party=calling;screen=yes;privacy=off
Timestamp: 1208269321
Contact: sip:09132922125@YYYY:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 394
v=0
o=CiscoSystemsSIP-GW-UserAgent 8081 616 IN IP4 YYY
s=SIP Call
c=IN IP4 YYYY
t=0 0
m=image 17900 udptl t38
c=IN IP4 YYYY
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
###################################
100 TRYNG<--------------------------------------------------------
488 Not acceptable here<------------------------------------
###################################
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP YYY:5060;x-route-tag=“tgrp:AZ”;received=YYYY
From: sip:922125@YYYY;tag=C79C7B0C-FB6
To: sip:473196@XXXX;tag=as4e6a4f02
Call-ID: 15AF6127-A2E11DD-BEA0FBB2-B1FB486C@YYYY
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
#########################
I tried canreinvite for provider and for sip client buth without succes.
If this version of asterisk needs same patches please give me a pice of adavace.
Rgds kajana.