[Asterisk-Users] one way sound with x-lite (sip) -3rd attempt !

[Thorsten Trapp]

Hi List, 

Additional with the latest tries from the below
I get a nice random seg fault when I hangup on PSTN.
(With obviously no sound on x-lite, still!)

asterisk -vvvvgc

results after hanging up the pstn line in:
-- Executing Hangup("SIP/1087997-d79f", "") in new stack
== Spawn extension (sip-phone-out, h, 2) exited non-zero on 
'SIP/phonenumber-d79f'
Segmentation fault
>

Since there is no normal release cycle can somebody
give us advise which asterisk/X-Lite/chan_capi versions
work well together ?
(date and time of CVS version)

Thanks in adavnce,
Thorsten


---------------------------------------------------------------
Hi all,

Still having the one way sound problem.
Any suggestions how to hunt the problem down ?

Regards,
Thorsten


---------------------------------------------------------------
Hi all,

We have a very basic * installation for testing purposes.
The * is connected to PSTN with BRI and setup with X-Lite
over plain lan. (local IP's)

OS: Linux/Debian unstable.
Asterisk CVS-10/29/03-23:46:26
chan_capi

On the IP side:
X-lite (build: 1084)

Calling and get calls on PSTN from X-Lite is no problem.
We only get sound from PSTN to X-lite.
Never from X.-lite to PSTN. 

The soundmeter on X-lite shows activity ... (not muted, correct device...)
When pressing numbers while having these silent calls in x-lite is playing
DTMFs at the PSTN phone side.

sip.conf:

[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
allow=all

[1*phonenumber*]
type=friend
username=NAME
secret=testpass
auth=md5
nat=no
host=dynamic
reinvite=no
dtmfmode=inband
callerid="Test" <*phonenumber*>
context=sip-phone-out


Any suggestions ?

Thanks,
Thorsten