Asterisk 13 FAX


Hello team! I’m planning to add fax functionality to my PBX. From research it seems that there is 2 options: spandsp and Digium. I lean towards Digium app, licensing is fine. However, they don’t have download for v13 Should I just download their version for v12 Asterisk? Any other suggestions on what to use, what works best? I have a pretty good plan on what I’m going to do but unsure which one to pick. Callcentric is my SIP provider and it supports T.38 Thanks Ivan --

Asterisk Users 3 months ago 2 Answers

No Audio While Call Forwarding, Yes Audio With Followme


Hi all,

the subject says it all. Technical details: - Asterisk - Behind NAT - Using external SIP Provider

The call forwarding is tested both with this functionality on the phone and with configuration in the dialplan. In the latter case a database variable is set to the external number, if set a Dial command calls this number. So really nothing fancy (actually I followed the example on ).

sip.conf has nat=yes, externip= ... and I tried every setting of directmedia in the providers configuration part.

Followme works flawlessly, so I'm really wondering if this is a NAT issue.

Can anyone point…

Asterisk Users 3.1 years ago 2 Answers

Asterisk 11 Queue Calls - Emulate Dial(b) Functionality


I currently run an Asterisk 10 system with hotdesking functionality set up. Several of the users have worked with a system in the past that supported BLF on their IP phones, and would like their current phones to behave in a similar fashion. Right now I have a really kludgy system that mostly works, but doesn't consistently trigger the cleanup macro to "clear" the device state on the end of a call. Rather than continue to beat my head against the wall playing "which context isn't firing an h extension to dump calls into the cleanup macro", I decided to…

Asterisk Users 3.2 years ago 2 Answers

Please dont tell me this is impossible


I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI->command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?

Asterisk Users 3.3 years ago 5 Answers

Digium IP Phones - Teleworker Capability?


We couldn't see anything about this on the Digium site, but maybe
someone here can comment? Do the new Digium phones provide good "teleworker" functionality? The benchmark we're comparing against is the capabilities of Mitel
3300 IP systems with Mitel 5330 IP phones (running their proprietary
MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select
TW mode and the IP of the gateway server). The phone reboots and it
is ready to be used (once the Mitel border gateway is set to recognize
the unit's ID, based on…

Asterisk Users 3.4 years ago 4 Answers

Adhearsion 2.0 Release For Asterisk 1.8+


Today marks another milestone in the Adhearsion project: the release of Adhearsion 2.0.  There has been a fury of activity in the last few days as we have worked hard to update documentation and release a brand new look-and-feel for the Adhearsion website.  We hope you like it.
So, with a small flourish and no small amount of relief, I'm pleased to announce the immediate availability of Adhearsion 2.0, the open source framework for the creation of voice applications.
Here are some highlights of the changes relative to the latest Adhearsion 1.x:

dahdi timing


Hi, We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are: ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks WARNING[22024] app_meetme.c: Unable to write frame to channel
Right now, dahdi in our setup uses the software timer (with which gives much better results compared to On an idle system (Centos 6, Asterisk 1.8.7, dahdi 2.5), dahdi_test results are pretty good at about %99.99. However, when loaded, the numbers fluctuate between %99.90 and %99.99 which seem to cause the…

Asterisk Users 3.7 years ago 0 Answers

Asterisk as register server through OpenSIPS


Hi all, I've been trying to register a SIP user agent to an Asterisk server using
OpenSIPS as SIP router. The functionality is working fine. However,
Asterisk uses the IP address of the OpenSIPS server as the peer IP address.
How can I use the original IP address of the peer without changing the
peer's nat=yes?
I appreciate any kind of help. Thanks! Regards,

Asterisk Users 3.8 years ago 1 Answer

Hint'ing with XMPP?


I have not ever done what you are talking about. However, I can tell you that our Openfire XMPP server has similar functionality because of their Asterisk-IM Plugin. From: [] On Behalf Of Jay R. Worthington
Sent: Saturday, December 03, 2011 8:11 AM
Subject: [asterisk-users] Hint'ing with XMPP? Hiya, can i use an XMPP Client to see the presence of a hint? I have configured asterisk in component-mode, seem's to work, but all users ( are online, even if 123 isn't a configured hint). Any good howto's out there, all the…

Asterisk Users 3.9 years ago 2 Answers



I am currently running Asterisk 1.4.8 and have been for quite a while, it
has served me well. Getting ready to build a new box to replace the existing installation of
Asterisk. My primary use of the Asterisk box is run queues. I am sure the queue
features and functionality have been updated, expanded since 1.4.8 and I am
wondering what version of Ast you guys would recommend. Looking for the
best version in terms of queue features, functionality. Also, an OS recommendation would be great. Been running on CentOS forever
and no reason…

Asterisk Users 3.9 years ago 7 Answers