VOIP-info.org Page Changes

Asterisk Consultants Switzerland

Sat, 06 Mar 2010 09:58:43 GMT

Diff from version 62 to 63 (current)

@@ -27,6 +27,15 @@

Post your Project Requirements today at

Web: [http://www.asterisk-consultant.com|www.asterisk-consultant.com]

+!!Asterisk Schweiz

+Consulting, Installation, Support and Customer Application Development.Worldwide Remote Support.

+

+Regensbergstrasse 242a

+8050 Z rich

+Switzerland

+__Voice:__ +41 43 317 9440

+__Mail:__ [mailto:info@asterisk-schweiz.ch]

+__Web:__ [http://www.asterisk-schweiz.ch]

!!Byll Technology SA

Development, deployment and Integration of Asterisk, SER and other open source products Solutions for PME.

**__Telephone:__ +41 78 751 85 51

User	Comment	Time	Version
newconnection		2010-03-06 09:58	63

Asterisk system vendors

Sat, 06 Mar 2010 09:50:23 GMT

Diff from version 1113 to 1114 (current)

@@ -415,6 +415,7 @@

* [http://www.wx3.se | wx3 telecoms-resources ab (Kista, Sweden)]

!!!Switzerland

+* [http://www.NewConnection.ch|New Connection GmbH] offers 3 Types of Asterisk PBX systems. From a low cost pre-installed version for the SOHO market (~6 users), over mid-size Appliance (~20 users), up to a full-size solution. Our service includes Consulting, Installation, Support, Upgrades and Customer Applications.

* [http://www.pbxplus.com|PBX Plus] offers advanced virtual PBX services such as speech recognition ivr, voicemail, call forwarding (landline, mobile, Skype, Digital VoIP), audio conference, (with transcription support), internet fax and on hold music in 40 different countries. For a limited time, PBX Plus is offering __free pbx__ accessible from 40 countries as a virtual extension with free Skype and VoIP forwarding.

*[http://invox.com| Invox] - Drag and drop designer, Invox, provides an enhanced IVR Solutions with Virtual PBX features. The designer won Best of Show during IT Expo. Out of box applications include payment over phone, surveys (type your questions and choose answer type), store locator, geographical routing, IM integration, CRM capture lead (Salesforce, Sugar, Zoho), CRM by Phone (with speech recognition), phone appointment scheduling and reminders, outbound automation, Skype/VoIP forwarding, disaster messaging, database integration and others.

* [http://voip.alunys.com | A l u n y s] (Belgique) Installation de votre central t ? l ? phonique IP (IPBX) cl ? sur porte et int ? gration de solutions Asterisk sur mesure.

@@ -438,6 +439,7 @@

* [http://wildix.com.ua| Wildix Ukraine]

* [http://startel.od.ua | StarTel Ltd (Ukraine)]

* [http://www.spectravoice.com | SpectraVoice Communications (Russia, Ukraine)]

+

User	Comment	Time	Version
newconnection		2010-03-06 09:50	1114

Asterisk system vendors INDIA

Sat, 06 Mar 2010 09:05:06 GMT

Diff from version 480 to 485 (current)

@@ -95,6 +95,59 @@

+^!!*astTECS - THE IP-PBX PEOPLE^

+We strive to enable Asterisk, the world's telephony engine,power your business. *astTECS is a telecom centric young enterprise headquartered in Bangalore, the Silicon Valley of India. We provide you complete telephony solutions: solutions with the standard/traditional features.

+

+!!!Enterprise IP PBX:

+Our IP PBX version provides you with the call handling capabilities you have come to expect from closed-source PBXs.

+

+!!!!Features -

+ 100% Wi-Fi solution

+ SMS Integration

+ Auto Monitoring

+ Receptionist console

+ Voicemail system

+

+!!!!Additional Features -

+ GSM Integration

+ CRM

+ Voice Logger

+

+!!!Call Center Solutions:

+The Call center version is a comprehensive customer contact platform that seamlessly integrates with your existing voice and data systems, whether VoIP, PRI and GSM.

+Compatible with PRI and GSM interface.

+

+!!!!Features -

+Our packages deliver world class applications like

+ Predictive Dialing

+ Progressive Dialing

+ GSM Integration

+ CRM

+ V Logger

+ Multi-party Conference with Talker Optimization

+ Voicemail System

+ CDR

+ SMS Integration

+ Auto Monitoring

+ IVR

+

+!!!GSM Card for Asterisk:

+*astTECS offers variety of solutions, based on the *ast g4 GSM Card for asterisk which can cater from SMEs to large organizations.

+

+!!!!Features-

+ Rapid Deployment.

+ Extreme cost Reduction.

+ CDR

+ ROI less than 6 Months.

+

+!!!Contact: Vaibhavi Talaulikar

+!!!Mobile: +91-9731311422

+!!!email: [t.vaibhavi@asttecs.com]

+!!!web: [http://www.asttecs.com]

+

^!__Venus Tele-Links__^

@@ -510,59 +563,7 @@

visit our website www.indianasterisk.com

-^!!*astTECS - THE IP-PBX PEOPLE^

-We strive to enable Asterisk, the world's telephony engine,power your business. *astTECS is a telecom centric young enterprise headquartered in Bangalore, the Silicon Valley of India. We provide you complete telephony solutions: solutions with the standard/traditional features.

-

-!!!Enterprise IP PBX:

-Our IP PBX version provides you with the call handling capabilities you have come to expect from closed-source PBXs.

-

-!!!!Features -

-100% Wi-Fi solution

-SMS Integration

-Auto Monitoring

-Receptionist console

-Voicemail system

-

-!!!!Additional Features -

-GSM Integration

-CRM

-Voice Logger

-

-!!!Call Center Solutions:

-The Call center version is a comprehensive customer contact platform that seamlessly integrates with your existing voice and data systems, whether VoIP, PRI and GSM.

-Compatible with PRI and GSM interface.

-

-!!!!Features -

-Our packages deliver world class applications like

-Predictive Dialing

-Progressive Dialing

-GSM Integration

-CRM

-V Logger

-Multi-party Conference with Talker Optimization

-Voicemail System

-CDR

-SMS Integration

-Auto Monitoring

-IVR

-

-!!!GSM Card for Asterisk:

-*astTECS offers variety of solutions, based on the *ast g4 GSM Card for asterisk which can cater from SMEs to large organizations.

-

-!!!!Features-

-Rapid Deployment.

-Extreme cost Reduction.

-Customization.

-IVRs

-Multiparty Conference.

-CDR

-ROI less than 6 Months.

-

-!!!Contact: Vaibhavi Talaulikar

-!!!Mobile: +91-9731311422

-!!!email: [t.vaibhavi@asttecs.com]

-!!!web: [http://www.asttecs.com]

-

+^

User	Time	Version
subbu	2010-03-06 09:05	485
v.sunil	2010-03-06 05:16	484
subbu	2010-03-05 15:18	483
v.sunil	2010-03-05 10:56	482
v.sunil	2010-03-05 10:54	481

voip-info.org

Sat, 06 Mar 2010 07:17:45 GMT

Diff from version 11853 to 11857 (current)

@@ -26,6 +26,9 @@

{/COMMENT}

!!NEWS

+* 2010-03-06 - [http://www.nomado.eu | Nomado Telecom, first global VoIP provider offering 24/7 LiveChat support]

+* 2010-03-05 - [http://www.comparebusinessproducts.com/blog/entryid/202/cdw-survey-reveals-increase-in-non-voice-usage-within-uc-solutions.aspx | New Unified Communications Survey shows rise in multimedia and IM and email use at the expense of the telephone]

+* 2010-03-05 - [http://www.ucexpo.co.uk/Register/?WT.mc_id=UC10422ENVOC | SIP Trunking "Getting it right 1st time" from The SIP School] Don't forget to register for this seminar @ UCExpo - next week, London - Olympia

* 2010-03-04 - [http://www.comparebusinessproducts.com/blog/entryid/200/new-cloud-based-voip-from-megapath.aspx | Another new cloud-based VoIP service - from Megapath]

* 2010-03-04 - [http://atlaug.com/drupal/fest2010|4th Annual Atlanta Asterisk VOIP Conference ] - Saturday, April 17, 9AM - Register here for [http://atlaug.com/drupal/festreg| FREE]

*2010-03-04 - [http://www.kamailio.org/mos/view/Present-and-Future-of-SIP-Routing-2010-London/|Present and Future of SIP Routing] - free event on Tuesday, March 9, 2010, London, UK

@@ -37,6 +40,7 @@

*2010-02-28 - [http://voiptoday.org/index.php?option=com_content&view=article&id=332:100-open-source-web-conferencing-system&catid=35:general&Itemid=136|100% open source web conferencing system.]

*2010-02-27 - [http://www.evilspurv.net/blog/2010/02/ari-asterisk-rickroll-improved/| Asterisk rickrolling improved] - A short example on lua usage in asterisk.

*2010-02-26 - [http://truvoipbuzz.com/2010/02/gvmax-most-comprehensive-google-voice-notification-system-tutorial/| Most comprehensive Google Voice notification system]

+

!!News Resources

* ((New Software Releases|Software Releases)) - Check here for recent VoIP-related software releases.

User	Time	Version
nomado	2010-03-06 07:17	11857
alanlindsay	2010-03-05 18:56	11856
vocale	2010-03-05 11:41	11855
vocale	2010-03-05 11:40	11854

Asterisk v1.2 upgrade to v1.4 gotchas

Sat, 06 Mar 2010 03:14:04 GMT

Diff from version 16 to 28 (current)

@@ -1,12 +1,28 @@

-The upgrade is fairly smooth, and there's lots of new features. It's the deprecated stuff that bites you.

+The upgrade is fairly smooth, and there's lots of new features. It's the deprecated stuff that bites you during the migration process.

+

+Start the migration with reading __UPGRADE.txt__, and then look at the __CHANGES__ file for details that were introduced with 1.4.0.

+

Make sure to read the new xxx.conf.sample files. That way you may detect new features/options that not seldomly also fix potential security issues.

+

+!!asterisk.conf

+For sure you will want to have "internal_timing=yes"!

!!extensions.conf

Lots of static variables are now much more versatile functions.

* SIP_HEADER() with (Via) now needs to be written as (Via,1) in Asterisk 1.4

+

+Many similar changes for variables are described in ugprade.txt:

* change ${TIMESTAMP} variable to ~np~${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}~/np~ function

* change ${CALLERIDNUM} variable to ${CALLERID(num)} function

+

+!!sip.conf

+* In the ~np~[general]~/np~ section "port=" has been renamed to "bindport=" to prevent misunderstandings

+* The default for QoS settings has changed from the old TOS to the [http://svnview.digium.com/svn/asterisk/branches/1.4/doc/ip-tos.txt?view=co|new DiffServ method]. This also applies to iax.conf, by the way.

+!!!BLF and hints

+* you will need to set "call-limit=" to make hints (SIP SUBSCRIPTIONS) work in Asterisk 1.4

+* also look at the general setting "limitonpeers=yes" and "notifyringing=yes" etc.

+

!!iax.conf

The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up.

@@ -19,29 +35,37 @@

add this to iax.conf: calltokenoptional = 0.0.0.0/0.0.0.0

add this to the ~np~[guest]~/np~ user in iax.conf: requirecalltoken=no (many guests will be using old Asterisk boxes)

In future: Upgrade the IAX peers and provide call tokens!

+

+!!zaptel turns into dahdi

+During the summer 2008 and after the release of 1.4.17 (?) zaptel has been [http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/|renamed to dahdi]. Since zaptel/dahdi provide timing to MeetMe this also matters for users that do not have any zapte (Digium) hardware (ztdummy vs. dahdi_dummy). Also the zaphfc module for the HFC-S ISDN cards is affected. See [http://www.asterisk.org/zaptel-to-dahdi|UPGRADE.txt] in your dahdi-linux source directory for details how to migrate.

+

+Asterisk 1.4 will continue to have support for Zaptel, although it will be enhanced to also transparently support DAHDI instead, and the documentation (and default configuration files) will encourage new users to use DAHDI instead of Zaptel.

!!AgentCallBackLogin()

AgentCallBackLogin is deprecated. If you were using this functionality, it can be replaced using dialplan logic, but there appears to be no way to update hints (if you use them) dynamically.

-

!!Music on Hold

-Digium has adviced everyone to discard the previously shipped music-on-hold files due to potential license issues.

+Digium has [http://af-za.facebook.com/note.php?note_id=209640921417|strongly advised] everyone in Aug. 2009 to discard the previously shipped music-on-hold files due to potential license issues.

!!Incompatibilities

!!!The g726 codec mess

-...

+Look at "g726nonstandard=yes" and "allow=g726aal2" - this is mentioned in upgrade.txt, but maybe you have missed to understand that you will need to review every single case where you have g726 enabled. In particular make sure you test cases where Asterisk is supposed to transcode between call legs with standard g726 and g726nonstandard!

!!!Obtaining and compiling the ilbc codec

-...

+The download script in contrib/ forgets to copy the ilbc code to asterisk/codecs/ilbc

+!!!g723 removed

+codec_g723 was removed

!!!rfc2833compensate

-...

+In Asterisk 1.4 you will need to turn on rfc2833compensate if you want to cleanly interact on DTMF with an Asterisk 1.2 system.

!!!Patches that would have deserved to be in 1.4

+See [http://users.netplex.net/~andrew/asterisk/], and in particular:

+# Codec Negotiation mod

+# Remote CID update mod

+# G.722 Codec mod

+# FAX app (spandsp)

----

-!!Official Information for Upgrading From Previous Asterisk Releases

-

+!Official Information for Upgrading From Previous Asterisk Releases

!!Build Process (configure script):

User	Time	Version
JustRumours	2010-03-06 03:14	28
JustRumours	2010-03-06 03:03	27
JustRumours	2010-03-06 02:56	26
JustRumours	2010-03-06 02:43	25
JustRumours	2010-03-06 02:41	24
JustRumours	2010-03-06 02:40	23
JustRumours	2010-03-06 02:39	22
JustRumours	2010-03-06 02:27	21
JustRumours	2010-03-05 21:47	20
JustRumours	2010-03-05 18:54	19
JustRumours	2010-03-05 18:54	18
JustRumours	2010-03-05 15:44	17