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 --
the subject says it all. Technical details: - Asterisk 18.104.22.168 - 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 http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ).
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…
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…
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?
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…
Hi, We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are: ERROR res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks WARNING app_meetme.c: Unable to write frame to channel
Right now, dahdi in our setup uses the software timer (with res_timing_dahdi.so which gives much better results compared to res_timing_timerfd.so). 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…
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
I appreciate any kind of help. Thanks! Regards,
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: firstname.lastname@example.org [mailto:email@example.com] 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 (xmpp:firstname.lastname@example.org
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…