Hello everyone,,
Strange issue:

Provider:- Simple Signal

Issue:

Whenever there is fax coming in, the call drops within a few seconds. If you call the same number and talk or leave a message or even put an analog phone and answer on the extension, the call has not problem.

The minute you send fax tones, the call drops - please help.

See logs below:

***************************** SIP message buffer start *****************************
INVITE sip:2147052058@24.155.238.18:5060 SIP/2.0
Via: SIP/2.0/UDP 208.77.200.13:5060;branch=z9hG4bK1opde1009gr01bk7g 641.1
From: <sip:208.77.200.13>;tag=SD8kghd01-209075441-1271026844322-
To: "demo trunk"<sip:4692875690@elinkdashboard.com;ssig=ACC-banjcfdj1lb94>
Call-ID: SD8kghd01-ad4951b2c6e0e5d15c3ee6ec414d8ddd-vrvvfv3
CSeq: 932004178 INVITE
Contact: <sip:208.77.200.13:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOT IFY
Accept: multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards: 9
Content-Type: application/sdp
Content-Length: 172

v=0
o=BroadWorks 119788678 1 IN IP4 208.77.200.13
s=-
c=IN IP4 208.77.200.13
t=0 0
m=audio 56092 RTP/AVP 0 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
***************************** SIP message buffer end ******************************

18:00:43 Try to send SIP message # (11/04/2010 23:00:43:554 GMT) # UDP # 343 bytes # buff size 1 # from: 24.155.238.18:5060 # to: 208.77.200.13:5060

***************************** SIP message buffer start *****************************
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.77.200.13:5060;branch=z9hG4bK1opde1009gr01bk7g 641.1
To: "demo trunk" <sip:4692875690@elinkdashboard.com;ssig=ACC-banjcfdj1lb94>
From: <sip:208.77.200.13>;tag=SD8kghd01-209075441-1271026844322-
CSeq: 932004178 INVITE
Call-ID: SD8kghd01-ad4951b2c6e0e5d15c3ee6ec414d8ddd-vrvvfv3
Content-Length: 0