Summary:ASTERISK-06253: app_dial connects but no voice
Reporter:Eldad Ran (eldadran)Labels:
Date Opened:2006-02-04 09:57:04.000-0600Date Closed:2006-05-11 09:56:17
Versions:Frequency of
Environment:Attachments:( 0) XXXXXXX31922-OK.txt
( 1) XXXXXXX53374-Ghost.txt
Description:I try to set a callback service, using SIP over the latest asterisk version 1.2.4,  The 1st call is initiated by an Originate manager event, and then a dialplan Dial command tries to dial and bridge the two chanels. 80% of the calls that get connected have a silence from the callee side. I know that as I've monitored the calls, and the callee side has a silence record, from the Dial attempt on wards.
This happen no matter what load the system is at.


I'm using SIP, with G711 alaw codec, OS madriva,
Comments:By: BJ Weschke (bweschke) 2006-02-04 18:51:10.000-0600

what version of Asterisk? can you do a "show version" pls? Can you also pls post a trace with "set debug" and "set verbose" ?


By: Eldad Ran (eldadran) 2006-02-11 10:04:46.000-0600

XXXXX*CLI> show version
Asterisk 1.2.4 built by root @ XXXX on a i686 running Linux on 2006-01-31 08:06:03 UTC
I've set a debug file, I'll have some more traffic for the service on moday, then I'll post it here.

By: Eldad Ran (eldadran) 2006-02-13 14:18:17.000-0600

The files loaded are debug logs:
XXXXXXX31922-OK.txt - is a good bridged call.
XXXXXXX53374-Ghost.txt - is a silent call, while the called party doesn't hear the caller, in the monitor file you can hear the called party speaking (hello hello).

By: BJ Weschke (bweschke) 2006-02-13 19:55:59.000-0600

Using "rtp debug ip xx.xx.xx.xx" in conjunction with your debugging, do you see rtp packets coming and going for these calls that have no audio or no? This smells like a network problem with RTP packets not getting through external to Asterisk itself.

By: Eldad Ran (eldadran) 2006-02-14 02:51:14.000-0600

The caller listen to an annoncment and then put in the target phone number by DTMF, up to this point everything is working, so I guess there is RTP traffic to this point, the problem is beyond that point, while bridging to the dialed number. The external gateway doesn't know about the bridge at all, it is all internal to Asterisk.
I'll set up an RTP debug today and try to filter out the ghost calls packets.

By: Eldad Ran (eldadran) 2006-02-28 05:50:05.000-0600

too much data, I couldn't capture the data. still have this problem.

By: Eldad Ran (eldadran) 2006-03-27 10:15:49.000-0600

It looks like it happens when the 1st let disconnects after the Dial command initiated, and the Dial command continues and connect to the other side. both channels are on and it can be viewed by using 'sip show channels' or 'show channels', I'm getting the BYE signal from the gateway if I'm not in the Dial command (have 'sip debug' on, and I saw it hitting the machine). if the called party (2nd) hangsup both channels are cleared.
for dialing I use options 'gh'.
any ideas?

By: Serge Vecher (serge-v) 2006-05-03 10:58:36

eldadran: please try the latest stable branch. Thanks.

By: Serge Vecher (serge-v) 2006-05-11 09:56:17

eldadran: if you are able to reproduce this problem with latest 1.2 branch code (rev > 26000), please reopen the bug with debugging information requested by bweschke.