I have problem of using call file to make auto outbound dial through FXO channel. I put "Channel: DAHDI/1/xxxxxxxxxx" (xxxxxxxxxx is the destination PSTN number to dial). For some reason asterisk did not dial the number but the control came to the context that I defined in the call file as if the peer had answed the call. It works if I change the channel from DAHDI to a SIP channel like SIP/4567 or I dial DAHDI/1/xxxxxxxxxx from a SIP channel. I am using asterisk1.4.23.1. Is it a bug in this release? Thanks Ray -- 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/20090218/60b3fd4f/attachment.htm
Eric Wieling, Asteria Solutions Group
2009-Feb-18 23:34 UTC
[asterisk-users] call file FXO channel problem
Ray Chen wrote:> I have problem of using call file to make auto outbound dial through FXO channel. I put "Channel: DAHDI/1/xxxxxxxxxx" (xxxxxxxxxx is the destination PSTN number to dial). For some reason asterisk did not dial the number but the control came to the context that I defined in the call file as if the peer had answed the call. It works if I change the channel from DAHDI to a SIP channel like SIP/4567 or I dial DAHDI/1/xxxxxxxxxx from a SIP channel. I am using asterisk1.4.23.1. Is it a bug in this release? >Analog FXO ports are considered "answered" as soon as dialing is finished. The telco does not provide a signal to the calling device to indicate the far end answered the phone. This does not apply to PRI or FXS. Virtually all SIP service providers use PRIs. If the service provider used analog you would also experience this when dialing using SIP. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sales at asteriasgi.com