VOIP-info.org Page Changes

Asterisk config extensions.conf

Sat, 05 Jun 2010 05:38:30 GMT

Diff from version 133 to 134 (current)

@@ -287,6 +287,7 @@

----

!!Example files on the net

+* [http://www.beardy.se/2010/06/05/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx|How to set up a SIP trunk in the Asterisk PBX] - Basic setup How-To/tutorial, SIP trunk and dialplan, to dial out, and in. With full example configuration files.

* John Todd, Loligo.com: [http://www.loligo.com/asterisk/current/extensions.conf]

* Tilghman Lesher: [http://asterisk.drunkcoder.com/extensions.conf]

User	Comment	Time	Version
beardy		2010-06-05 05:38	134

Asterisk config sip.conf

Sat, 05 Jun 2010 05:35:25 GMT

Diff from version 164 to 165 (current)

@@ -461,6 +461,7 @@

!!See also

* Asterisk 1.4 comes with a new adaptive general jitter buffer also for chan_sip. More details [http://www.asterisk.org/node/48317|here] by Russel.

* ((Asterisk sip channels)): More documentation on SIP.conf

* ((SIP)): Session Initiation Protocol

User	Comment	Time	Version
beardy		2010-06-05 05:35	165

Asterisk system vendors

Sat, 05 Jun 2010 02:32:10 GMT

Diff from version 1147 to 1148 (current)

@@ -644,6 +644,7 @@

* [http://www.altitude.com | Altitude Software] : The leading contact center solution running on asterisk (in 15 countries, 50 customers)

* [http://www.3rdwavebiz.com] 3rd Wave Consulting Services (Washington, DC)

* [http://www.abacus-telecom.com/ |Abacus Telecom (USA, Spain, Panama, Colombia, Nicaragua) - Asterisk powered PBX Servers, Asterisk consulting, Digium Hardware]

+* [http://www.adcapnet.com | Adcap Network Systems] - Design, installation, sales and support of Digium Switchvox, custom Asterisk in Atlanta, Georgia, and Miami, Florida.

* [http://www.asteriskpbxs.com | Asterisk PBXS - A PBX in Excess] : Asterisk, Trixbox, Elastix, Other distros with Asterisk pre-installed in select pre-tested servers certified to just work! Integrations. (Puerto Rico, USA)

* [http://www.customvoicesystems.com |CustomVOICE Systems - Converged Communications] IP/PBX - Voice & Data Integration, Calling Card Systems, VoIP Billing

* [http://pbx.active-sites.com |Active-Sites, Inc. (Barberton, OH, US)]

@@ -765,6 +766,7 @@

* [http://www.zerabox.com |Zeracom, sales and service of IP-PBX systems, including Zerabox, an Asterisk based IP-PBX (Greenville, South Carolina)]

* [http://www.zone24x7.com | Zone24x7(San Jose, CA)]

* [http://www.jacons.net | J&A Consulting (Tijuana/San Diego)]: Consulting in Linux, Asterisk, Predictive Dialer and Open Source Software.

+

User	Comment	Time	Version
rversluis		2010-06-05 02:32	1148

Asterisk h extension

Fri, 04 Jun 2010 21:38:50 GMT

Diff from version 8 to 13 (current)

@@ -4,6 +4,10 @@

Use ((Asterisk cmd DeadAGI|DeadAGI)) instead of AGI here.

Do not call ((Asterisk cmd Hangup|Hangup)) from within the h extension.

+

+When the 'h' extension is running, the call legs have already been torn down. There is no way to delay this happening, and you can't do anything in the 'h' extension that needs to read audio from the channel (since no audio will appear, the first time it tries to read audio it will abort). Thus ((Asterisk cmd Playback|Playback))() or ((Asterisk cmd Background|Background))(), for example, does not work. Essentially, the only things that make sense to use in the 'h' extension are those that don't have anything to do with the external channel that was involved before the hangup. No audio, no DTMF, etc.

+

+__June 2010:__ ((Asterisk cmd Wait|Wait))() could be modified to work on a dead channel, but as it stands right now the code assumes the channel is alive, and to ensure that things work properly, it has to read incoming media from the channel so it can be discarded... and if at any time waiting for or reading media from the channel fails, it exits, because there's no point in continuing to wait since the call is gone. Instead of Wait() use something like "System(sleep 3)".

!!Note

Use with great care: Apparently some channel variables get destroyed when the call is hung up, and those variables aren't available anymore (or have inconsistent values) when the h extension is being called. Therefore you shouldn't rely on this special extension for billing/calling card purposes.

User	Time	Version
JustRumours	2010-06-04 21:38	13
JustRumours	2010-06-04 16:30	12
JustRumours	2010-06-04 16:29	11
JustRumours	2010-06-04 12:55	10
JustRumours	2010-06-04 12:03	9

voip-info.org

Fri, 04 Jun 2010 20:45:29 GMT

Diff from version 12444 to 12454 (current)

@@ -22,11 +22,12 @@

{/COMMENT}

!!NEWS

-* 2010-06-02 - OnSIP launches [http://www.bestvoipcalling.com |mobile voip app review program] dedicated to SIP Apps for Business. [http://www.prweb.com/releases/2010/06/prweb4078624.htm|Release]

+* 2010-06-04 - [http://a2billing2asterisk.googlepages.com/|New A2Billing v1.7.1 Automated - Hassle Free - Install Script for Trixbox, Piaf, Elastix, AsteriskNow, or Vanilla Asterisk released!]

+* 2010-06-02 - OnSIP launches [http://onsip.com/mobile-voip |mobile voip app review program] dedicated to SIP Apps for Business. [http://www.prweb.com/releases/2010/06/prweb4078624.htm|Release]

* 2010-06-02 - [http://www.i6net.com/2010/06/02/new-vxi-voicexml-browser-5-0-released/|New VXI* VoiceXML browser 5.0 for Asterisk released!] - by I6NET

* 2010-06-01 - [http://www.Asterisk4arab.com/|Asterisk4arab] The Revolution arrived the Arabic World, Opening the first Asterisk forum in Arabic language.

-* 2010-06-01 -[http://www.openvox.cn/aboutus/show.php?itemid=64&lang=2|Industry 1st Embedded Asterisk Motherboard Released by OpenVox]

-* 2010-05-31 -[http://itunes.apple.com/us/app/asteriskc2dpro/id287004096?mt=8|AsteriskC2DPro for iphone] New version released. Click to call via your AsteriskPBX and iphone.

+* 2010-06-01 - [http://www.openvox.cn/aboutus/show.php?itemid=64&lang=2|Industry 1st Embedded Asterisk Motherboard Released by OpenVox]

+* 2010-05-31 - [http://itunes.apple.com/us/app/asteriskc2dpro/id287004096?mt=8|AsteriskC2DPro for iphone] New version released. Click to call via your AsteriskPBX and iphone.

* 2010-05-31 - [http://voiptoday.org/index.php?option=com_content&view=article&id=425:exclusive-interview-for-voip-today-magazine-with-daniel-constantin-mierla-the-co-founder-and-core-developer-of-kamailio-openser&catid=905:general | Exclusive interview for VoIP Today magazine with Daniel-Constantin Mierla the co-founder and core developer of Kamailio (OpenSER).]

* 2010-05-29 - [http://www.officesip.com|OfficeSIP Communications] makes its TURN server officially [https://sourceforge.net/projects/turnservernet/|open-source].

* 2010-05-28 - ESCAUX FMC: [http://blog.escaux.com/|Nokia E-series handsfree over-the-air-provisioning (OTAP) of WiFi and SIP settings]

@@ -38,6 +39,7 @@

* 2010-05-26 - [http://astercc.org/tips/2010/05/astercc-and-astercc-box-released-014-beta.html|asterCC] released 0.14 beta, added new agi script for more features and start support asterisk 1.6.

* 2010-05-26 - [http://tinyurl.com/36nzfkj|Special on Asterisk Training in Colombia] click for more info (in spanish).

* 2010-05-26 - [http://www.2n.cz/en/news-and-events/3g-wireless-router-now-with-data-traffic-control/|3G wireless router with SIP client now with data traffic control ]

+

!!News Resources

User	Comment	Time	Version
techsupport		2010-06-04 20:45	12454
richboy360		2010-06-04 20:05	12453
a2b2b		2010-06-04 16:52	12452
spamblock	Oh, please. Stop with the polite comment. Don't post your website, period.	2010-06-04 12:36	12451
spamblocker	guys....is there something wrong withy this? I wish to have this link please	2010-06-04 12:11	12450
spamblockerblocker	Come on guys, it's not that hard to do it the same way as the others...	2010-06-04 11:58	12449
spamblock		2010-06-04 11:48	12448
spamblock	Hell no	2010-06-04 11:26	12447
spamblocker		2010-06-04 09:09	12446
spamblocker		2010-06-04 09:08	12445