[Asterisk-Users] IAX calls dropout on button press

[Shaun Ewing]

Hello all,

Over the weekend, I setup and linked an Asterisk box at another site to the
Asterisk box here.

The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100
phones. The phones at the other end are Grandstream BT-100 SIP phones. The
Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream
phones run 1.0.4.68.

Both Asterisk boxes are running stable CVS "CVS-06/07/04-16:18:54".

The phones all use G711u to their respective Asterisk boxes. GSM is used
between the Asterisk boxes. The boxes are 60ms apart.

When on a call, the quality of the call is perfect. I've spent around 3
hours over 2 days on calls between the two systems.

Today when I was on a call I noticed by accident that if you press a button
(1 through 0 and */#) the call drops out after a few seconds. With the help
of a colleague at the remote site we discovered the following:

Grandstream <-> Asterisk <-> IAX <-> Asterisk <-> Grandstream. Any phone
presses button, the call drops out.

Cisco <-> Asterisk <-> IAX <-> Asterisk <-> Grandstream. Cisco presses
button, call drops out. Grandstream presses button, call doesn't drop out.

It doesn't seem to matter which side originated the call.

What can be seen with SIP debug is 5 of these:

Retransmitting #1 (no NAT):
INFO sip:7300 at 192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0549e133
From: "Shaun Ewing" <sip:7011 at 192.168.0.1>;tag=as3dbbd61d
To: <sip:7300 at 192.168.0.254>;tag=ba97574145fedb8a
Contact: <sip:7011 at 192.168.0.1>
Call-ID: 5f3fb48b72348c2243355b793185e569 at 192.168.0.1
CSeq: 104 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=7
Duration=250
I$P
 to 192.168.0.254:5060

We then get something like:

Jun  7 14:34:06 WARNING[1125329600]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call 5f3fb48b72348c2243355b793185e569 at 192.168.0.1 for
seqno 103 (Request)
set_destination: Parsing <sip:7300 at 192.168.0.254;user=phone> for
address/port to send to
set_destination: set destination to 192.168.0.254, port 5060
Reliably Transmitting:
BYE sip:7300 at 192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0549e133
From: "Shaun Ewing" <sip:7011 at 192.168.0.1>;tag=as3dbbd61d
To: <sip:7300 at 192.168.0.254>;tag=ba97574145fedb8a
Contact: <sip:7011 at 192.168.0.1>
Call-ID: 5f3fb48b72348c2243355b793185e569 at 192.168.0.1
CSeq: 105 BYE
User-Agent: Asterisk PBX
Content-Length: 0

All the other debug bits indicate that the call terminated normally.

I was wondering if anybody had experienced anything like this before?

Regards,

Shaun