Hello,<br><br>I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly with a CISCO mediaGW in order to send faxes to the PSTN using T.38.<br><br>When Asterisk sends the initial INVITE containing the T.38 media offer in the SDP, the CISCO answers with a 488 Not Acceptable Media.<br>
Apparently, it looks like a configuration problem in the CISCO, but I have tested the CISCO with the Zoiper client and it successfully sends faxes. The only difference I have noticed between the Asterisk and Zoiper is that whereas the Asterisk sends the T.38 SDP information in the initial INVITE, Zoiper establishes a voice call first and then re-negotiates(with a re-INVITE) the session in order to send the T.38 media. <br>
Is it possible to make Asterisk work like this? or is this a problem in the configuration of the CISCO? Any ideas?<br><br>Thanks in advance.<br><br>Regards,<br><br>Santi<br><br><br>**The call-file I'm using is:<br><br>
Channel: SIP/080999999999@outbound-<div id=":3n" class="ii gt">calls<br>MaxRetries: 3<br>WaitTime: 30<br>Set: LOCALSTATIONID=22222<br>
Set: LOCALHEADERINFO=T38 fax<br>Set: T38CALL=1<br>Set: T38TXDETECT=yes<br>CallerID: 22222<br>Context: fax-out<br>Extension: 22222<br>priority:1</div><br><br>My sip.conf file is:<br><br>sip.conf<br>[general]<br>bindport=5060 <div id=":3n" class="ii gt">
; UDP Port to bind to (SIP standard port is 5060)<br>
bindaddr=192.168.222.160 ; IP address to bind to (0.0.0.0 binds to all)<br>domain=192.168.222.160 ; Add IP address as local domain<br><br>t38pt_udptl=yes<br><br>[outbound-calls]<br>type=friend<br>context=openser<br>
allow=all<br>;dtmfmode=info<br>host=10.100.222.201<br>insecure=very<br>canreinvite=no<br>pedantic=no<br>call-limit=10<br><br>The extensions.conf file<br><br>[fax-out]<br>exten =>s,1,Set(FAXFILE=/root/santi/fax/prueba.tif)<br>
<div id=":3n" class="ii gt">exten =>s,n,SipDTMFMode(inband)<br>
exten =>s,n,SendFax(${FAXFILE})<br>exten =>s,n,Hangup</div><br></div><br>The SIP trace is:<br><br><br>INVITE <a href="mailto:sip%3A0809999999@10.100.222.201" target="_blank">sip:0809999999@10.100.222.201</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.222.160:5060;branch=<div id=":2z" class="ii gt">z9hG4bK306b777c;rport<br>
Max-Forwards: 70<br>From: "22222" <<a href="mailto:sip%3A22222@192.168.222.160" target="_blank">sip:22222@192.168.222.160</a>>;tag=as43e12927<br>To: <<a href="mailto:sip%3A0809999999@10.100.222.201" target="_blank">sip:0809999999@10.100.222.201</a>><br>
Contact: <<a href="mailto:sip%3A22222@192.168.222.160" target="_blank">sip:22222@192.168.222.160</a>><br>Call-ID: <a href="mailto:4f9fb8387458a3c6205e2c4467e48ad2@192.168.222.160" target="_blank">4f9fb8387458a3c6205e2c4467e48ad2@192.168.222.160</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.6.0.6<br>Date: Tue, 10 Mar 2009 11:29:44 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 290<br><br>v=0<br>o=root 525135648 525135648 IN IP4 192.168.222.160<br>s=Asterisk PBX 1.6.0.6<br>c=IN IP4 192.168.222.160<br>t=0 0<br>m=image 4222 udptl t38<br>a=T38FaxVersion:0<br>a=T38MaxBitRate:9600<br>
a=T38FaxRateManagement:transferredTCF<br>a=T38FaxMaxBuffer:400<br>a=T38FaxMaxDatagram:400<br>a=T38FaxUdpEC:t38UDPFEC<br><br>#<br>U +0.015757 <a href="http://10.100.222.201:5060/" target="_blank">10.100.222.201:5060</a> -> <a href="http://192.168.222.160:5060/" target="_blank">192.168.222.160:5060</a><br>
SIP/2.0 488 Not Acceptable Media<br>Reason: Q.850;cause=65<br>Date: Tue, 10 Mar 2009 11:29:18 GMT<br>From: "22222" <<a href="mailto:sip%3A22222@192.168.222.160" target="_blank">sip:22222@192.168.222.160</a>>;tag=as43e12927<br>
Allow-Events: telephone-event<br>Content-Length: 0<br>To: <<a href="mailto:sip%3A0809999999@10.100.222.201" target="_blank">sip:0809999999@10.100.222.201</a>>;tag=417D2718-582<br>Call-ID: <a href="mailto:4f9fb8387458a3c6205e2c4467e48ad2@192.168.222.160" target="_blank">4f9fb8387458a3c6205e2c4467e48ad2@192.168.222.160</a><br>
Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK306b777c;rport<br>Server: Cisco-SIPGateway/IOS-12.x<br>CSeq: 102 INVITE<br><br><br>#<br>U +0.000164 <a href="http://192.168.222.160:5060/" target="_blank">192.168.222.160:5060</a> -> <a href="http://10.100.222.201:5060/" target="_blank">10.100.222.201:5060</a><br>
ACK <a href="mailto:sip%3A0809999999@10.100.222.201" target="_blank">sip:0809999999@10.100.222.201</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.222.160:5060;branch=z9hG4bK306b777c;rport<br>Max-Forwards: 70<br>From: "22222" <<a href="mailto:sip%3A22222@192.168.222.160" target="_blank">sip:22222@192.168.222.160</a>>;tag=as43e12927<br>
To: <<a href="mailto:sip%3A0809999999@10.100.222.201" target="_blank">sip:0809999999@10.100.222.201</a>>;tag=417D2718-582<br>Contact: <<a href="mailto:sip%3A22222@192.168.222.160" target="_blank">sip:22222@192.168.222.160</a>><br>
Call-ID: <a href="mailto:4f9fb8387458a3c6205e2c4467e48ad2@192.168.222.160" target="_blank">4f9fb8387458a3c6205e2c4467e48ad2@192.168.222.160</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.6.0.6<br>Content-Length: 0<br><br></div>