Hello,
I try to append a URI to the SIP dial syntax, however the URI were not shown
in the sip debug message. I have read one of
the post in the list which actualy show the URI string in the debug message
(at the To: field). Is there any setting I need to set or turn on during
compilation of asterisk? I have the head version of asterisk and my
extension.conf setting is proveded below:
exten => 777,1,Answer
exten =>
777,2,SetVar(VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml)
exten => 777,3,Dial(SIP/1234@192.168.1.72,10,t)
exten => 777,4,Hangup
SIP Debug message:
*CLI> dial 777
-- Executing Answer("OSS/dsp", "") in new stack
<< Console call has been answered >>
-- Executing SetVar("OSS/dsp",
"VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml") in new
stack
-- Executing Dial("OSS/dsp",
"SIP/1234@192.168.1.72|10|t") in new stack
We're at 192.168.1.74 port 18952
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:1234@192.168.1.72 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.74:5060;branch=z9hG4bK280927bb
From: "asterisk" <sip:asterisk@192.168.1.74>;tag=as2e2564e0
To: <sip:1234@192.168.1.72>
Contact: <sip:asterisk@192.168.1.74>
Call-ID: 294f37ba6b5b17be38fbf31022fabfb2@192.168.1.74
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 07 Mar 2005 16:21:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 263
Thanks
CFC
