Asterisk as a SIP trunk termination point.


I have been looking online for a definitive how-to on using Asterisk as a
SIP trunk termination point . I am seeing conflicting messages and
methodologies for doing this. I am not going to use a commercial vendor for this trunk, it will be used in
testing out various customer scenarios and am looking at Asterisk as one
alternative. I see two ways to do from all my research
1.use two asterisk in VMs(or on bare metal) to originate and terminate a
peer trunk.
- this would be good if one doesn't have…

Asterisk Users 3.5 years ago 0 Answers

T.38 Incoming Fax Problem


Hello, I've installed the free (one user) Fax for Asterisk (FFA) license.
Outgoing faxes, using T.38, to the PSTN work quite well. However,
incoming faxes, do not seem to detect tones and certainly do not switch
to T.38. The call drops as soon as the fax answers. Since I am using FreePBX (, I am trying to modify the adjunct
files to make this work. That makes it a little more confusing for me to
grasp. In the sip_general_custom.conf file, I have added: t38pt_udptl=yes I am sure something else is missing. The IP PBX simply…

Asterisk Users 3.7 years ago 0 Answers

best softphone for 2012?


Just installed asterisknow 1.6. I can access freepbx. I need to test
system on my LAN. Which softphone is best to use? I'm running ubuntu
on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX
for incoming/outgoing calls. No video.

Asterisk Users 3.8 years ago 8 Answers

Voip: Asterisk Security Releases Available


The Asterisk Development Team has announced security releases for Asterisk 1.4, 1.6.2 and 1.8. The available security releases are released as versions 1.4.43, and These releases are available for immediate download at The release of Asterisk versions 1.4.43,, and resolves an issue with possible remote enumeration of SIP endpoints with differing NAT settings. The release of Asterisk versions and resolves a remote crash possibility with SIP when the "automon" feature is enabled. The issues and resolutions are described in the AST-2011-013 and AST-2011-014 security advisories. For more information about the details of…

VoIP News 3.9 years ago 0 Answers

Asterisk PBX


Asterisk is software that turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and more. It is used by small businesses, large businesses, call centers, carriers and governments worldwide. Asterisk is free and open source. Asterisk is sponsored by Digium People often tend to think of Asterisk as an "open source PBX" because that was the focus of the original development effort.  But calling Asterisk a PBX is both selling it short (it is much more) and overstating it (it can be much less).  It is true that Asterisk started out as…

General 4.1 years ago 0 Answers

No Audio after attended tranfer


Am 18.07.11 16:15, schrieb Alex Vishnev:
> I am wondering if anyone hit this case yet. I am using and doing an attended transfer. The transfer is going to an outbound number (normally AA on another IP PBX). the audio on the first transfer is fine. But if the user requests a transfer from AA to another department, I loose audio from Asterisk to the 2nd transfer. Based on the review of SIP packets, the second transfer issues ACK+SDP. Anyone have experience with that? it looks like ACK+SDP is not being handled properly by asterisk. I searched thru…

Asterisk Users 4.3 years ago 1 Answer

Trying to turn off TLS....


Hey all, I'm currently running Asterisk 1.8.3 with FreePBX It's tied to another IP-PBX via TLS. I have two problems going on.. 1.) Every so often (say roughly every 24 hours), Asterisk stops handing calls back to the second IP-PBX. The call rings indefinitely and Asterisk complains about the certificate like below:
a. [Mar 16 16:10:04] VERBOSE[2973] tcptls.c: SSL certificate ok
b. [Mar 16 16:10:05] VERBOSE[3688] tcptls.c: SSL certificate ok
c. [Mar 16 16:10:05] ERROR[3688] tcptls.c: Certificate did not verify: unable to get local issuer certificate So, it says it's okay and then within the same…

Asterisk Users 4.6 years ago 0 Answers