similar to: How to Forcing Call Disconnect?

Hi, Please take a look at the following extensions.conf:- exten => _11XXXX,1,NoCDR() exten => _11XXXX,2,Dial(SIP/${EXTEN},10) exten => _11XXXX,3,VoiceMail() I'm already using realtime for some extensions/users/voicemail. Is there any way to do the following at point 3?:- Lookup the realtime users db and read the MailBox column for that buddy. If the mailbox column is empty, play

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

Hi, I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes? [client1_phones] include => client1_internal include => client1_outgoing_calls include => test_calls include => parkedcalls [client2_phones] include => client2_internal include =>

Hi, I think ive found a bug but need someone to double check. Whenever I issue a "reload" in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed...

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

Hi, I'd like to be able to create contexts in real-time when I add new clients to my asterisk box. Currently, I have to create a blank context in extensions.conf and add:- switch => Realtime/@ Is there any way to avoid the step of creating the blank context and simply include all the entries from the database? Thanks Dan -------------- next part -------------- An HTML attachment was

The Asterisk Development Team has announced the release of Asterisk 13.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

FOR IMMEDIATE RELEASE Puerto Escondido, Mexico, April 1st, 2010: Digium launches Asterisk VCC (TM) - a new virtual communication platform for enterprises, the public sector and the home. =========================================================== Asterisk 1.8 will contain a stunning new technology for all Asterisk users world- wide - virtual communication clouds or VCC (TM). With this

I've added an introduction article about the ARA on my web site http://www.voip-forum.com/ The same text is now also added to CVS head as README.realtime. On the same site, you will also find the news item about how we used Asterisk for a call from an airline jet above Greenland to Stockholm, Sweden. The world is getting smaller and more connected every day! /Olle

As we are getting closer to release of CVS head as version 1.2, we're in need of your help. One of the cool new features in CVS head is the ability to store the actual voicemail messages in a database. This is not using ARA, the Asterisk Realtime Architecture, but directly interfaces with ODBC from app_voicemail. It stores both meta data and audio in the database, which will give you real

* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND VIDEO TO MICROBLOGGING! In a surprising move, Digium in partnership with Edvina today released a new channel driver for Asterisk, chan_tweet. The driver connects seamlessly to several microblogging platforms, including Twitter, Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of this new module is to

hello, i am trying to set up a asterisk server (version 1.2.26 by now) with realtime configuration but the user shouldnt register directly to the server, instead i have set up a ser registration proxy. Everything works fine so far, but i can?t use the hint feature. Its possible to subscribe to a given hint, but the status is allways unavailable and also i dont get a notify. Could someone

Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload

I'd love to understand what the function of the peer list returned by 'sip show peers' is for, especially when Realtime is used. If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As each peer (phone) registers, or a call is made to the peer, Asterisk adds them to the list returned by 'sip show peers'. Correct? Apparently Asterisk doesn't

I am using mysql sipfriends and can't seem to get the MWI to work. From what I've read it seems this is not supported with that dynamic system, and probably never will be. I was thinking of just setting a cron job or something to check every minute for voicemail and set our sip NOTIFY messages as needed. Also, the PAP2-NA has the ability to reboot via a sip notify and I would like to be

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To:

Dear Asterisk Community, Asterisk 1.0 was released at Astricon 2004, in September last year. It's been almost a year and we haven't been able to go ahead and release a new version. Now is the time to try to move forward again. As we've outlined before, the process is this: -------------------------------------------------------------------- * Code freeze: At this point, we'll

Hi folks, I used the ast2sql.pl script (found on www.voip-info.org) to put into the database a simple sip.conf. Among other entries, you could find: [general] context=sip-in ;incoming sip calls Well, the script put the comment into the database entry, and asterisk started complaining about a 'sip-in ;incoming sip calls' context not found in extensions.conf. IMHO the comments should be

I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname => mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called "sysname" in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something