Displaying 2 results from an estimated 2 matches for "as3ed791f3".
2008 Nov 07
1
Outgoing SIP calls dropped after 30 seconds.
...) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.82.224.202:5060:
INVITE sip:+18881231234 at 216.82.224.202 SIP/2.0
Via: SIP/2.0/UDP <public IP>:5060;branch=z9hG4bK6ea30a1a;rport
From: "8881231234" <sip:+18881231234 at public IP>;tag=as3ed791f3
To: <sip:+18005551212 at 216.82.224.202>
Contact: <sip:+18881231234 at public IP>
Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 07 Nov 2008 19:06:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCR...
2007 Jul 12
0
No subject
...s get it wrong.
In your case there is a Record-Route header in the response so the ACK
request should be being sent to that address. Perhaps your firewall is
not correctly mangling that to allow the request to find its way back
to your Asterisk server.
Record-Route: <sip:216.82.224.202;lr;ftag=as3ed791f3>
Regards,
Greyman.