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Wed 01 of Aug, 2007 [09:12 UTC]

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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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voip-info.org suggestions

09 March 2007
Hi, i would like to update the information for my company's products on the page at http://www.voip-info.org/wiki-VOIP+Phones, but the instructions on the homepage indicate not to edit pages directly, rather to create a new page with your company's products. I am not sure how to get that page to update with our products if not by editing it directly. Any insight would be very appreciated. Thanks, my login is 'vishaljain'.


02 October 2006

Is it possible to clean up among the "Comments", some of the comments doesn't have anything to do with the pages, example, comments on the start page.


17 August 2006

Proposing a new section on Voip security. plenty of info out there on this subject which is becoming more and more relevant. See fron instance this voip blog lots more out there


24 July 2006

It there anyway for example to add the Skype page already here to the Navigation menu
and would that be OK? Thanks

11 April 2006

There doesn't seem to be any section for direct VoIP connections, or connections that don't include a VoIP or SIP service. I understand that this can be done but can find no exlanation of it.


I would like to propose a section for UMTS products.

Dec 23, 2005

I would like to propose that there is a license statement on the first page.

And what type of Common Creative License or ... is less of my concern but something that both contributors and readers can refere to.


Nov 15, 2005

With the popularity of RSS feeds, it would be nice if we had a section where we could select our favourite RSS feeds for our News Readers.

i.e. Asterisk RSS Feed XML



Mar 10,2005

I've noticed an increase in interest on our company Web Site from people searching for information on 3G IMS (IP Multimedia Subsystem) and Security Gateway accplication for interconnecting UMTS security domains. How about a section on 3G and IMS in particular?



Mar 7, 2005

When viewing the history of a page, the Version (Ver) column seems to be off by one for everything but the most recent revision. For example, a page with 24 versions goes from 2-24 with 24 being shown twice (current and second most current). I haven't really checked other wikis to determine if this is the norm or not.



I second the suggestion for an archive for SHOUT Box info. Net.Com ShoutIP, ShoutGate, etc. would be a great resource!

Answer:
Click on the "Read more" link at the bottom of the SHOUT Box sidebar.



How to use Asterisk as a B2BUA ?

Asterisk is not really designed for this.
See: Asterisk is not a SIP proxy
However, you might be able to simulate some of the functionality of a B2BUA depending on what your requirements are.


How to build a Chinese version?

jackon kang

See here:
http://tikiwiki.org/TikiWikiInternationalizationAndLanguages



An autodiscovery tag in the home page and recent changes page for the RSS feed would be nice.
example: <link rel="alternate" type="application/rss+xml" title="RSS" href="url/to/rss/file">
This goes in the page header.
- ben




Sept 20, 2004
Please add an RSS feed and register with syndic8.
Done: There is already an RSS feed: http://www.voip-info.org/tiki-wiki_rss.php?ver=2&css=y
There is a button for it at the very bottom of each page, but its not too obvious.
The RSS feed has already been submitted to syndic8 and is in their review queue.
Thanks.

6-July-2004
How do you add a new page to this site? Perhaps a button or a description somewhere would be helpful, thanks.
Done: How to add information to this wiki

4-July-2004
Please add a section for VoIP Billing And Calling Card Issues.
Done: VOIP Billing

3/05/04
Shout Box archive
It would be nice if there was a archive for the shout box

12-Feb-2004
Fixed link problem with Asterisk How to connect with CAPI
Jim


There appears to be a linking problem from Asterisk tips and tricks to
Asterisk How to connect with CAPI (this link seems to get the same problem - the page does exist if you search for capi and click the relevant link) - perhaps because of the colon, can it be renamed? Thx
12-Feb-2004

Is anyone else having problems with the site in Firebird(Mozilla)? The site works fine in IE but is
firebird on two different machines(XP) here is an alignment issue. The best way to show this is to
log in with a bad user and pass. The error message shows up but you have to scroll right to see it.
kent
1/18/2004

ivelin: How about adding a page for local VoIP User Groups?

01/12/04

The wonderful thing about a wiki is that you can do this yourself.
Please feel free to add this page and links to it where you think appropiate.

Jim
01/12/04



oej: Please add search capability!

I added the search box. I think this wiki uses the native Mysql Full text search, which has a few strange quirks.
For example, if you search for a popular word that appears on over 1/2 the pages, then it will find nothing.
Search for: Asterisk and get nothing,
but search for: dialplan and find pages that mention dialplan.

Jim 9/8/03

oej: Weird... I agree, it's broken.
mose: I confirm full-text is used in 1.7. but search engine is rewritten in 1.8, and rc2 maybe could be stable enough for voip-info.org as you use only wiki feature.

oej: Jim, could you add a "Recent changes" page - or make it a link in the menu. I only find the RSS feed in the page bottom, but's not made for browsing... Thank you!

Click on the :: just in front of "Wiki" in the Menu box.
I'll see if there is a way to make this more obvious.
Jim 9/8/03

I changed the menu blocks around.
Let me know if better
Thanks.
Jim 9/8/03

Oej 9/10/03: Much better! I like it.

Oej: 9/15/03: Jim, I would like you to edit the RSS feed for Recent changes. The title is not
"voip-info.org" - it's just Tiki news or something. Is it possible to change the name of the
RSS feed?

Will the search be improved. Any reports from tiki.org?

Try the RSS now.
Let me know if OK.


I saw a discussion thread on the tiki mailing list on the search, but didn't look like anything was going to happen soon. Maybe I should set up an external search engine to index the pages. Wouldn't be real-time like the Tiki search though. i.e. it would only update the index nightly. As the number of pages expands the built in Tiki search should work better as there will be fewer cases where the search words appear on over 1/2 the pages and are thus invisible to the search.
Jim 9/15/03

Oej 9/15/03: The RSS title works now. Thank you! Yes, the web grows. Added some stuff on SIP and on SER and new contributors keep showing up in the "Recent Changes" list. Great!




Oej 9/20/03: Jim, another suggestion: Google likes real URLs, without the ?page=. Most Wikis I've used is able to - with the help of apache rewrites - handle URLs like http://www.voip-info.org/wiki/Asterisk
It would help the Wiki being indexed by search engines.

Oej 9/21/03: I think the Mysql database has to few available connections. I get "Tiki can't get connection to database" sometimes. Maybe we're getting some users in here after all...

I changed the URLs. Had to change the Tiki source to get it work correctly. Let me know if you find anything broken.

Not sure what was happening with Mysql. I just set PHP to not use persistent DB connections and restarted Mysql and the web server. Maybe that will help.
Let me know if you get any more errors.

According to the log analysis program (which I'm not totally sure is reliable) we've been averaging around 50 unique visitors a day to the site.

Thanks.
Jim 9/21/03



Moved to a new server
Jim 12/5/04



Enabled the user page watch feature that automatically notifies users when pages they mark are changed.
To use click on the "eye" icon in top right corner of page.
The "eye" icon only appears if you are logged in.

Jim 12/13/03



Installed: http://turck-mmcache.sourceforge.net/
Hopefully will improve performance

Jim 12/14/03

--
VON Spring 2004 Report
--

Please put new entries at TOP of this page!

Created by oej, Last modification by Vishal Jain on Fri 09 of Mar, 2007 [20:37 UTC]

Comments Filter

VoIP Thailand Service provider

by pariwat_p on Saturday 24 of February, 2007 [05:15:44 UTC]
XebecInfo Technologies Co., Ltd.

The company to provide service in Thailand, who would like to get more product solution please visit http://www.xebecinfo.com
If you have any question about Asterisk please let me know,

Best Regards,
PS

SIP command to ISDN suspend/resume

by Luis Filipe Ribeiro on Wednesday 20 of December, 2006 [10:58:23 UTC]
Hi I'm looking for a SIP command to be mapped to ISDN suspend/resume.
There are one feature in ISDN PBX, for ACD, which use ISDN suspend/resume. I'm looking for a SIP commands to be mapped to ISDN suspend/resume. Is that possible? The scenario is:
SIP phone->SIP PBX->Cisco VoIP gateway->ISDN net->ISDN PBX.
thk
Regards.

Warning[4608] chan_sip.c:1109 __sip_xmit

by Le Son Phat on Friday 27 of October, 2006 [12:51:09 UTC]
for me ask
this warning:

 Warning4608 chan_sip.c:1109 __sip_xmit of 0x8c779c & (len 479) to 10.0.0.10:5061 returned -1: bad file descriptor
 Warning4608 chan_sip.c:1082 __sip_xmit of 0x8c779c & (len 479) to 10.0.0.10:5061 returned -1: bad file descriptor

what do they mean????
please reply for me at herowater1@yahoo.com

Linksys Wip-330

by Erica on Saturday 20 of May, 2006 [19:48:34 UTC]
Hi
Since the Linksys Wip-330 has been finally released (and I've purchased one), why don't add a page about this phone?
Wip-300 already has a page ;)

Bye!

azacall200 settings?

by offline_thk on Monday 17 of October, 2005 [20:34:08 UTC]
Hi I have a azacall200 from azatel and is trying to configure it for a new ip-telephone provider. But of course my old one doesn’t give support if I'm not using them and my new one doesn’t give support if I don't have one off there boxes. Have someone configured this box for some provider and can help me with what everything means. Worth mention is that I live in Sweden and don’t have that much to go on what to enter. I have set tit to factory settings, supervisor logon. I have some info on configuring other boxes for my new provider such as D-link DVG-1120S and Cisco ATA 186. But these don't have half off all the settings so no luck getting it to work. Please please.. Hope that someone can help! Regards
Edit

Open Source PBX

by Anonymous on Sunday 26 of December, 2004 [14:56:25 UTC]
(:biggrin:) For people looking for a true SIP PBX, you need to go to SIPfoundry.org. They now have a great single download with install script that provides a top of the line PBX. This is great and since I am running it now, I hope this site begins to cover and talk about it.

KS
Edit

Service provider ratings

by Anonymous on Wednesday 24 of November, 2004 [02:43:59 UTC]
Some way of rating the various service providers would be useful. I've tried three so far and haven't been impressed with call quality, uptime and/or technical support with any of them.

Small Business Newbie

by tbixler on Wednesday 27 of October, 2004 [19:10:12 UTC]
(:frown:)
I would like a laundry list of things I would need to set up a Asterix system for a small business that has 3 incoming analog lines and 12 extensions. Would VoIP be used for the internal calling and the handling of voicemail?? Basically what would I need to set this up and then connect it with the PSTN for incoming and outgoing calls to be allowed.
Any help would be greatly appreciated. Thanks in Advance
Tony Bixler
Edit

Web/program to VOIP

by Anonymous on Sunday 10 of October, 2004 [17:00:26 UTC]
HI all,

Is there already a program or Java interface that can be used to call voip phones?(:idea:)
Edit

Information for novice VIOPers

by Anonymous on Monday 16 of August, 2004 [18:53:13 UTC]
There are those who know nothing about VIOP (like me) It would be nice to have as one of your first links a link to a page explaining just what VIOP is and how to use it.

Secondly, a link to a simple how-to page that walks someone through the steps necessary to establish their first VOIP connection.

Thanks
wcstuart@aol.com

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