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...6956 This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies.

...6956 This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies.

Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history

Hi, I am trying to read data from Microsoft SQL. I tried to use odbcConnect (in RODBC) but failed. What should the "dsn" be? Does anyone know how to do it? I will appreciate it. Thank you! Xiaorong ________________________________________________________________________ The preceding e-mail message (including any attachments) contains information that may be confidential, be

Hello, I have been running Asterisk for the past 5+ years on RedHat and I never upgraded it before. All my Asterisk software is of the following release: 1) Asterisk 1.4.21.2 2) Libpri-1.4.4 3) Zaptel-1.4.11 I would like to move the OS to CentOS and then I thought I can at the same time ponder about upgrading Asterisk releases. However, I am bewildered by the myriad of different releases like 1.6,

Polycom 601, hands down. - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Warren Sent: Thursday, June 15, 2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk? If you had approx $300 per phone as a budget and needed to buy

...____ This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies.

...____ This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies.

Hi, I have noticed it for a while but I just thought about confirming this with the Asterisk community. As the compilation of DAHDI will need to reference Kernel-devel, does it mean that after DAHDI is installed, we should not yum update kernel because it will affect the operation of DAHDI? Thanks. The contents of this e-mail are intended for the named addressee only. It contains information that

I remember seeing something somewhere that described how I could get SIP header information with Asterisk. It was a command or a variable. Anyone know what it is? Thanks. Doug.

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James

Guys, I am planning on building a small SIP PBX with a single ISDN line. Currently I am looking into the specs of a very tiny barebone system that has an option Colognechip base ISDN controller. The only thing is that the ISDN module that comes with this barebone hooks up to the motherboard using USB. My intention is to allow incoming and outgoing calls from SIP to ISDN. Is this setup in any way

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060

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Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a >reload extension is performed, those dynamically created extensions in the regcontext vanish. Now

Do you mean the peristence of connecting a specific phone to a specific server? If so, then it's relatively easy. The ldirectord has a persistence setting that does that. If I'm misunderstanding you, then could you explain further what you mean? Regards, - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

Hi Kristian, Anywhere I can read about this Soekris/AstLinux project? ... Regards, Hans -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Kristian Kielhofner Sent: Thursday, February 24, 2005 6:02 AM To: jim@vanmeggelen.ca; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]