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...it may concern, When dialing out an 800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This is not due to a problem on your Asterisk or your line- the bad sound effect occurs in VoicePulse. (just spend lots of time finding that out) Assaf Benharoosh MCP, MCSA, MCSE ab@franticllc.com <mailto:ab@franticllc.com> Frantic, LLC. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/4620202d/attachment.htm

Hi all, Does anyone know how to get the channel ID on the other side of the call? For example: When SIP/50 calls SIP/21, and the call is answered by SIP/21 I get: SIP/21-6735 answered SIP/50-b456 ${CHANNEL} will show me SIP/50-b456. Is there a parameter or a workaround to get the SIP/21-6735 part? Thanks. Assaf Benharoosh -------------- next part -------------- An HTML attachment was

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1" 200 9438

Hi All, Did anyone came across a 411 XML service I can feed to the "service" button with XML? Some other feed I can manipulate to XML query? Assaf Benharoosh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/dbf552ac/attachment.htm

I found a message from you to the asterisk users mailing list from 2001. I was wondering if you got (or still have) an asterisk system working with the linejack? If so, would you be willing to assist me with mine? I seem to have things working, and * says that caller ID is coming in, but I can't get * to actually answer the call. Thanks, Greg -- NetIO.org

...erisk-users@lists.digium.com My apologies, I realized too late that what I thought was someone's email address appears to go to the list. --__--__-- Message: 4 Subject: RE: [Asterisk-Users] Asterisk and Linejacks Date: Thu, 22 Jul 2004 14:24:09 -0400 From: "Assaf Benharoosh" <ab@franticllc.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com Hi, I actually gave up on the LineJack. I'm using Digium 4 FXO card- which does the job pretty well.=20 Assaf Benharoosh MCP, MCSA, MCSE ab@franticllc.com Frantic, LLC. 246 West 38th Street 2nd Floo...

Hi all, Does anyone know if it's possible to continue talking to one member of an initiated conference call on Cisco 79xx ? In other words- disconnect one of the parties. Thanks. Assaf -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040821/525e26b0/attachment.htm

Hi, I'm trying to achieve the following feature that's common in Avaya systems: A user page all extensions in a full duplex mode- he can hear all, and all can hear him via their phones' speaker. When one of the extensions picks up the handset, the call is bridged between the pager and the person who picked up. All the rest are disconnected. Does anyone have an idea or ever

For some reason I can't get the Dial command 'C' flag to work. The calls are recorded in the CDR with the 'C' on. Does anyone have an idea? extensions.conf: exten => 114,1,Dial(SIP/114,,C) It shows in the lastapp: cdr: | 2004-10-05 13:16:02 | "112" | | 114 | intern | SIP/112-3fb6 | SIP/114-0e7a | Dial | SIP/114||c | 6 |