I just upgraded to 1.4.0 but only get one way audio when a cisco router with a voip card calls in. The funny thing is that it works fine when I call out though the same gw. I did some debugging and found that when I’m running asterisk 1.2.11 I get this:
Reliably Transmitting (no NAT) to 10.10.96.245:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.96.245:5060;branch=z9hG4bKA51D99;received=10.10.96.245
From: sip:365@10.10.96.245;tag=DBA5679C-730
To: sip:9999@10.10.98.1;tag=as3ddbd7b2
Call-ID: 9D62B69D-BD4311DB-BE2AFD57-FCCD7D58@10.10.96.245
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:9999@10.10.98.1
Content-Type: application/sdp
Content-Length: 237
10.10.98.1 is asterisk
10.10.96.245 is the cisco gw
10.10.98.229 is the phone.
It looks like asterisk is telling the cisco to send the audio stream directly to the phone in the case of asterisk 1.4.0. Anyone know why this is and how to fix it?
I am having a similar problem since installing 1.4 :-
Here is the Asterisk Log for a regular call :-
Executing [00414499222@internal:1] Dial(“SIP/99-0973bb20”, “SIP/engin/0414499222”) in new stack
– Called engin/0414499222
[Feb 17 15:43:01] NOTICE[4978]: chan_sip.c:14329 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 21
– Call on SIP/engin-0974a550 left from hold
– SIP/engin-0974a550 is making progress passing it to SIP/99-0973bb20
– SIP/engin-0974a550 is ringing
– Call on SIP/engin-0974a550 left from hold
– SIP/engin-0974a550 is making progress passing it to SIP/99-0973bb20
[Feb 17 15:43:10] NOTICE[5000]: rtp.c:781 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 202.61.13.40
– Call on SIP/engin-0974a550 left from hold
– SIP/engin-0974a550 answered SIP/99-0973bb20
== Spawn extension (internal, 00414499222, 1) exited non-zero on ‘SIP/99-0973bb20’
This call audio works fine but when i get one way audio i get the following :-
Executing [00414499222@internal:1] Dial(“SIP/99-0973bb20”, “SIP/engin/0414499222”) in new stack
– Called engin/0414499222
– Call on SIP/engin-0974a550 left from hold
– SIP/engin-0974a550 is making progress passing it to SIP/99-0973bb20
– SIP/engin-0974a550 is ringing
– Call on SIP/engin-0974a550 left from hold
– SIP/engin-0974a550 is making progress passing it to SIP/99-0973bb20
– Call on SIP/engin-0974a550 left from hold
– SIP/engin-0974a550 answered SIP/99-0973bb20 – Packet2Packet bridging SIP/99-0973bb20 and SIP/engin-0974a550
== Spawn extension (internal, 00414499222, 1) exited non-zero on ‘SIP/99-0973bb20’
– parse_srv: SRV mapped to host mel-sbc02.byo.engin.com.au, port 5060
So whatever the following is, causes the problems :-
– Packet2Packet bridging SIP/99-0973bb20 and SIP/engin-0974a550
Seems to be that canreinvite=no is being ignored and it is trying to pass the audio directly to the phone and obvioulsy the NAT is not allowing it.