[Asterisk-Users] Problem with call originating from Cisco 7940 SIP phone to a SIP peer

[Mark Raming]

Hi,

I'm having the following setup:

Asterisk server (duh) running 1.0.0 stable release
Cisco 7940 SIP phone.
SIP peer to bridge to the PSTN world (www.pilmo.com)

With the sip.config shown below, things work like a charm (changed phone
numbers and passwords to protect the innocent). The Cisco uses the ULAW
codec as can be expected, and the GSM codec is used with pilmo, so asterisk
does the transcoding.

However, when I remove the disallow=xxx and allow=gsm lines under the
[pilmo] section, things become weird: I can still make outgoing calles. They
end up using the a-law codec for both the Cisco and the pilmo channel. On
the cisco, I can hear the calling/called  party, but the calling/called
party cannot hear me. I have also configured DIAX as a user. Making the same
call from Diax (still using alaw with pilmo) works fine.

Any clues? Am I doing something wrong? Is this a known problem for which I
should get a more recent version of the code?

Rgds,

mark
[general]
context=external
port=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
allow=alaw
useragent=AddPac SIP Gateway
nat=no
register => 31165570909:hr0921dsf at nelson.ritstele.com:5060/31165570909
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0

; Incomming calls from Pilmo
[212.26.192.155]
type=user
insecure=yes
context=incomming

; Outgoing calls to Pilmo
[pilmo]
type=peer
insecure=yes
fromdomain=nelson.ritstele.com
host=nelson.ritstele.com
fromuser=31165570909
allow=gsm
disallow=ulaw;
disallow=alaw;
dtmfmode=rfc2833

[2201]
type=friend
username=2201
secret=1234abcd!
host=dynamic
nat=never
mailbox=2201
context=internal
canreinvite=yes
dtmfmode=rfc2833