[asterisk-users] VoIP-GSM gateway problem

[Andreas v. Heydwolff]

I bought a MV-372 for 2 SIM cards as the one channel model seems to work 
well (see
http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk).
The setup is such:

    --------- Inet --> VoIP provider ---> POTS
   |
   |
(iax2, NAT)
   |
asterisk
(on abox with iptables fw)
   |
  (SIP, LAN)
   |----------> SNOM190 phones
   |
    ----------> SIP-GSM-module ---> SIM cards --> mobile phone networks

Sound, however, is too bad for the SIP to GSM module to be usable. Call 
initiation from the LAN to the GSM network works but the audio stream 
stops and continues for about 1 to 3 seconds each, irregularly 
alternating between various durations of both states. Latency is around 
300ms for the module which registers as a SIP extension. The machine is 
a PII with a 400MHz Celeron. Transmission is via the alaw codec, as 
recommended. The "RTP Packet Length" setting for the GSM module is 20ms.

Do any of you have suspicions why the module does not work as expected? 
(The vendor has not yet answered yet but it's weekend in Taiwan as 
well). Perhaps the GSM-module firmware is not up to par, and/or the SNOM 
doesn't cooperate well in the bridged connection.

--AvH

---------------------------------------
from sip.conf:

[general]
port=5060
externip=23.45.67.89
bindaddr=123.456.789.220
localnet=123.456.789.0
defaultexpirey=120
maxexpirey=3600
context=internal
disallow=all
allow=alaw
language=de
canreinvite=no


[GSM]
type=friend
host=dynamic
defaultip=123.456.789.222
secret=xxxxxxx
qualify=yes
username=xx
fromuser=xx
context=gateway
call-limit=2
dtmfmode=inband
allow=alaw
insecure=very

[SNOM190]
[3]
type=friend
host=dynamic
defaultip=123.456.789.221
secret=xxxxxx
qualify=yes

I've tried nat=yes and no
canreinvite=yes as well
qualify on and off in both clients