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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

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 an IP PBX or control over the
remote end.
2.use asterisk to terminate one end of the trunk by using it to log in to a
sip trunk and then define the peering.

I know the first one is easy, but the second may be the way to go.
What does asterisk need to supply the remote end of the sip trunk? I realize
that it is based on the remote end, I just haven’t seen any example for the
most popular IP PBXs, like Cisco UC or Avaya CM/SES.

Ids this is all trust, I could see the SIP trunk peer entry and trusted and
maybe bypassing the peer username/password in the definition.

Has anyone done this in a proof of concept lab or in production?

T.38 Incoming Fax Problem


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

In the sip_general_custom.conf file, I have added:


I am sure something else is missing. The IP PBX simply does not detect
the tones nor tell the switch to implement T.38. If you have some ideas,
please let me know. I sure appreciate the help.

AsteriskNOW 1.7.1


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.

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 http://downloads.asterisk.org/pub/telephony/asterisk/releases

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 these vulnerabilities, please read the security advisories AST-2011-013 and AST-2011-014, which were released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:


Security advisory AST-2011-013 is available at:

* http://downloads.asterisk.org/pub/security/AST-2011-013.pdf

Security advisory AST-2011-014 is available at:

* http://downloads.asterisk.org/pub/security/AST-2011-013.pdf

Thank you for your continued support of Asterisk!

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 a phone system for a small business (see the “Brief History” section for the juicy details) but in the decade since it was originally released it has grown into a universal tool for building communications applications.  Today Asterisk powers not only IP PBX systems but also VoIP gateways, call center systems, conference bridges, voicemail servers and all kinds of other applications that involve real-time communications.

Asterisk is not a PBX but is the engine that powers PBXs.  Asterisk is not an IVR but is the engine that powers IVRs.  Asterisk is not a call center ACD but is the engine that powers ACD/queueing systems.

Asterisk is to communications applications what the Apache web server is to web applications.  Apache is a web server.  Asterisk is a communication server.  Apache handles all the low-level details of sending and receiving data using the HTTP protocol.  Asterisk handles all the low level details of sending and receiving data using lots of different communication protocols.  When you install Apache, you have a web server but its up to you to create the web applications.  When you install Asterisk, you have a communications server but its up to you to create the communications applications.

Web applications are built out of HTML pages, CSS style sheets, server-side processing scripts, images, databases, web services, etc.  Asterisk communications applications are built out Dialplan scripts, configuration files, audio recordings, databases, web services, etc.  For a web application to work, you need the web server connected to the Internet.  For a communications application to work, you need the communications server connected to communication services (VoIP or PSTN).  For people to be able to access your web site you need to register a domain name and set up DNS entries that point “www.yourdomain.com” to your server.  For people to access your communications system you need phone numbers or VoIP URIs that send calls to your server.

In both cases the server is the plumbing that makes your application work.  The server handles the low-level complexities and allows you, the application developer, to concentrate on the application logic and presentation.  You don’t have to be an expert on HTTP to create powerful web applications, and you don’t have to be an expert on SIP or Q.931 to create powerful communications applications.

For deep details about Asterisk, please visit the Oficial Documentation

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 JIRA cases, but did not find anything like that. Any help would be appreciated.
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maybe this is the problem you have:


best regards