asterisk users - Jun 2004 - IAX calls dropout on button press

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@192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0549e133
From: "Shaun Ewing" <sip:7011@192.168.0.1>;tag=as3dbbd61d
To: <sip:7300@192.168.0.254>;tag=ba97574145fedb8a
Contact: <sip:7011@192.168.0.1>
Call-ID: 5f3fb48b72348c2243355b793185e569@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@192.168.0.1 for
seqno 103 (Request)
set_destination: Parsing <sip:7300@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@192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0549e133
From: "Shaun Ewing" <sip:7011@192.168.0.1>;tag=as3dbbd61d
To: <sip:7300@192.168.0.254>;tag=ba97574145fedb8a
Contact: <sip:7011@192.168.0.1>
Call-ID: 5f3fb48b72348c2243355b793185e569@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

Grandstream Phones do have a bug where they sometimes send an UDP packet to
port 0 instead of port 5060. The effect is that you see 5 times the error
message

 WARNING4101: chan_sip.c:601 retrans_pkt: Maximum retries exceeded on call
 1c4b33fae3ec4116@192.168.233.67 for seqno 22209 (Critical Response)

on the asterisk box, because Asterisk sends some Command to the BudgetTone,
and the Budgettone answers, but to port 0. So Asterisk retries ... up to 5
times.

See http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone under
"Early Dial" (althought the bug appears with other SIP traffic as
well).

For a crude fix, you can use the IP port redirection of Linux:

iptables -t nat -A PREROUTING -p udp -i eth0 --dport 0 -j REDIRECT --to-port
5060


-- 
http://www.holgerschurig.de

ok richard at GS saya  to please mail you * debug with the error to support 
@grandstream.com so they can see it...