* You are viewing the archive for August 15th, 2012

How To Input NULL In CDR Report

Hi all, In the CDR report I would like to filter search to display all the calls that has a null value (blank) for the “Dst. Channel” field. I try to input space, , $null$, but it display error. Just left the field blank will display ALL.

So what is the correct value to be inputted as null or blank?

Thanks for the help :)

BR, Anam.

UDP Miss A Hangup On SIP

Is it possible to miss a UDP SIP packet to hangup a call?
Using 1.4.43 I had a call from on Asterisk box (server) to a low end client (chan_alsa) not hangup.

Could this be due to missed UDP SIP packet to hangup?

Is there anyway for a client asterisk (chan_alsa again) to monitor the connection and if the channel is not there to hangup?

Thanks,

Jerry

Send Fax From Asterisk

Thanks for sharing the link. Actually I’m looking for a different approach without installing/using third party i.e. a user sends an email to Asterisk
(which is also running mail service), as Asterisk receives the mail where the mail contains attachment and subject contains destination number, Asterisk will download the file and capture the number and later send fax to destination number just like ‘.call’ file.

Does anyone worked on this scenario? If yes/no, please let me know at earliest.




please check it. might be it will help

Incompatible Voice Frame Ulaw/alaw

Hi list!

When I receive an incoming call from a SIP peer where I’ve configured

disallow=all allow=alaw
(and no other codec)

I can see the following NOTICE on the console:

Dropping incompatible voice frame SIP/peer07-0000007c of format ulaw since our native format has changed to (alaw)

My question is: where can I change the native format from ulaw to alaw
(or something else)? Is ulaw, as the native format, adjustable in some config file or is it hard-coded into Asterisk?

It doesn’t seem to have any effect on the voice quality but the messages on the console are quite annoying.

PS: The peer doesn’t support ulaw.

PPS: Asterisk 10.7.0

Thanks a lot!
Markus

Extensions DTMF

Greetings

Recently I’ve noticed some of the extensions on our VoIP server are not beign recognized by the IVR of a few destinys I’ve tested. I press que IVR
number but it simply don’t transfer. This is not ocurring to all extensions. I’m using rfc2833 to all extensions and Elastix on CentOS 5.5.