Hi all,
I'm trying to test last Git version of Janus SIP gateway with my asterisk-11 server and i'can't make a call from the demo application.
Other demo appplication like MCU and echo are running perfectly but not the sip gateway .....
When I try a call to an astersik extention, the REGISTER messages are OK but it's like astersik doesn't see the invite from janus.
I did some tcpdump on my Asterisk and i got this on the invite execution :
Via: SIP/2.0/UDP 192.168.1.143:57435;rport;branch=z9hG4bKeX0K4H00UjeH
v=0
o=- 3647685735571460965 6928923365833771788 IN IP4 192.168.1.143
s=-
t=0 0
m=audio 22636 RTP/AVP 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.143
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
Max-Forwards: 70
From: <
sip:80...@0.0.0.0:0>;tag=DmeFtSBm2y38H
To: <
sip:12...@192.168.1.147>
Call-ID: 7381d5b3-ae15-1232-23bd-080027a92b78
CSeq: 64540339 INVITE
Contact: 8000 <sip:80...@192.168.1.143:57435;transport=udp>
User-Agent: sofia-sip/1.12.11
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 0
The INIVITE request part seems to be missing :
INVITE
sip:12...@192.168.1.147 SIP/2.0
Does someone see that already ?
Regard,
Damien Fetis.