Upgrade to 1.4.0 causes one way audio and strange sip INVITE

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

v=0
o=root 1589 1589 IN IP4 10.10.98.1
s=session
c=IN IP4 10.10.98.1
t=0 0
m=audio 18976 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

But when I use 1.4.0 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=z9hG4bK9F1AE6;received=10.10.96.245
From: sip:365@10.10.96.245;tag=DAA848D0-135
To: sip:9999@10.10.98.1;tag=as3a34a433
Call-ID: FDADC758-BD1C11DB-BD09FD57-FCCD7D58@10.10.96.245
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:9999@10.10.98.1
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 6266 6266 IN IP4 10.10.98.229
s=session
c=IN IP4 10.10.98.229
t=0 0
m=audio 2260 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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?

Thanks,
schu

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.

I just built the latest SVN “Asterisk SVN-trunk-r55716” and the problem went away. It sounds like a bug was resolved and that 1.4.1 should work fine.

schu

I got the same NOTICE today in bith Asterisk 1.4.19.1/2 branch.