I have a problem of using call file to make an auto dial out call through FXO channel. I defined the channel in the call file as "Channel: DAHDI/1/8775203463" When I put the call file under the /var/spool/asterisk/outgoing dir it did not call out but came to the context I defined in extensions.conf as if the callee had answered the call. If I make a call from an extension to DAHDI/1/8775203463 it'll success. . If I change the channel to SIP/8000 and put the call file under /var/spool/asterisk/outgoing it is also success - it calls the extension 8000 and the controle goes to the context after the extension 8000 answers the call. I am using asterisk 1.4.23.1. Is it a bug introduced in this release? Thanks. -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090217/ba6a0dc7/attachment-0001.htm
You should post the call file. Also, I'd use DAHDI/G1 instead of DAHDI/1 as that ties the call to a specific port/line (perhaps what you want to do?) _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ray Chen Sent: Tuesday, February 17, 2009 2:05 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] call file bug? I have a problem of using call file to make an auto dial out call through FXO channel. I defined the channel in the call file as "Channel: DAHDI/1/8775203463" When I put the call file under the /var/spool/asterisk/outgoing dir it did not call out but came to the context I defined in extensions.conf as if the callee had answered the call. If I make a call from an extension to DAHDI/1/8775203463 it'll success. . If I change the channel to SIP/8000 and put the call file under /var/spool/asterisk/outgoing it is also success - it calls the extension 8000 and the controle goes to the context after the extension 8000 answers the call. I am using asterisk 1.4.23.1. Is it a bug introduced in this release? Thanks. -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com <http://www.mail.com/Product.aspx> ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090217/603564a9/attachment.htm
Asterisk Asterisk
2009-Feb-17 20:46 UTC
[asterisk-users] Updated modules to be released (FaxDetect, GenderDetect, MachineDetect, others)
I will be releasing updated versions to many of the detection modules next week.
They include better support of Asterisk 1.2, 1.4, and 1.6, better detection,
better parameters, an easier build system, and usability is enhanced.
The updated modules include:
* FaxDetect, LineDetect, and MachineDetect - which many are presently using
* PlayDetect and BackgroundDetect - playback with specification of detection
modules to use
* GenderDetect, NoiseDetect, and AnswerDetect - new modules
Contact me off the list if you need updated modules or have questions, comments,
or feedback.
Justin Newman
nt_jnewman at yahoo.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090217/9ceed80b/attachment.htm
On Tue, Feb 17, 2009 at 8:04 PM, Ray Chen <ray1017 at techie.com> wrote:> I have a problem of using call file to make an auto dial out call through > FXO channel. I defined the channel in the call file as "Channel: > DAHDI/1/8775203463" When I put the call file under the > /var/spool/asterisk/outgoing dir it did not call out but came to the context > I defined in extensions.conf as if the callee had answered the call. If I > make a call from an extension to DAHDI/1/8775203463 it'll success. . If I > change the channel to SIP/8000 and put the call file under > /var/spool/asterisk/outgoing it is also success - it calls the extension > 8000 and the controle goes to the context after the extension 8000 answers > the call. I am using asterisk 1.4.23.1. Is it a bug introduced in this > release? >It's probably the result of FXO lines having very little signalling. IOW, asterisk picks up the FXO line and dials the number... By then it has no way of knowing wheather the other party answered or not. That's probably why your getting your "other leg" too early in the process. Maybe you could try to Wait() for a few seconds on your dialplan. -- exvito