How to debug MixMonitor misbehaviour

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Thanks for the input. Yes, I did call out many times, but the recording doesn't happen even after
the call is bridged and there is two way audio. I also took out the "b"
option and so it should recording the ringing right (even before call is
bridged) but it doesn't do that or any recording at all. Any other suggestions as to what I can do to see why this is not recording? Regards, On Tue, May 3, 2011 at 2:13 AM, virendra bhati wrote: > Hi,
>
> As per your…

Asterisk Users 4.3 years ago 2 Answers

receive faxes

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does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? i've tried using "app_fax.so" with T38 but i keep
getting "Transmission failed"

Asterisk Users 4.3 years ago 10 Answers

dial from voicemail

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Hi
Is it possible to dial from within voicemail to reach another extension.
I would like my customers to have a choice of dialing 1 to get my cell
phone while in voicemail or to just leave a message at the tone. Thanks Kelly

Asterisk Users 4.3 years ago 1 Answer

Asterisk 1.6 Questions

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I have a couple of questions about asterisk 1.6:
Can you change codecs mid-call upon re-invite? Can you handle the SDP offer-answer in the 200-ACK instead of the usual
INVITE-200?
Thanks in advance for any insight.
Gary

Asterisk Users 4.3 years ago 0 Answers

Join and listen to conference call through web-interface

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Thanks, looks really helpful for managing connected users (half my problem). On the web-interface question, how do I create a website with a
[call-in] button? I'm using Drupal, so will make it a members only page. Basically they
click the [call-in] button, and straight away they're in the
conversation. It needs to grab input from mic, so I'm thinking Java or
Flash. Do you know of a solution which implements this? Thanks for all suggestions, Alec Taylor On Tue, May 3, 2011 at 4:13 PM, Andraž wrote:
> This will help you start:

Asterisk Users 4.3 years ago 0 Answers

Multiple cards using same IRQ - getting IRQ errors and hissing

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I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an
HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800
8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting
IRQ errors when running "dahdi show status". I see that sharing IRQs for Digium cards isn't recommended, so I'm
trying to set it so each card gets its own. From the few web sites
I've read so far, including Digium's FAQ site, I've…

Asterisk Users 4.3 years ago 1 Answer

Failed to get 160 samples from write factory

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Hello, I see a lot of these messages in the debug log : /[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples
from write factory 0xae17e18
[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples
from write factory 0xae17e18
[May 3 15:47:09] DEBUG[19081] audiohook.c: Read factory 0xae173e0 and
write factory 0xae17e18 both fail to provide 160 samples
[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples
from write factory 0xae17e18
[May 3 15:47:09] DEBUG[19081] audiohook.c: Read factory 0xae173e0 and
write factory 0xae17e18 both fail to provide 160 samples

Asterisk Users 4.3 years ago 0 Answers

asterisk 1.8 rpms and additional modules from source

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Hi All i have a asterisk install from rpm (digium yum) and i wish to install res
spandsp i comipled from source using the same asterisk version and copyied
the file to the module directory but when trying to load i get this WARNING[18463] loader.c: Module 'res_fax_spandsp.so' was not compiled with
the same compile-time options as this version of Asterisk another thing if i install the rpms and have the necessery like iksemel for
jabber installed will the res jabber be installed Thanks

Asterisk Users 4.3 years ago 0 Answers