similar to: Voicemail2 - auto fill the dialing extension?

When I drop my file into the outgoing folder, the call is completed but the 'Context' entry is not respected. Instead, it drops into the default context. It does drop "properly" into the default context and function as would be expected. I looked through the source but didn't see any reason it would be completely ignoring the context. Call file: (where

I guess the subject says it all.. :) I am running the CVS from right now.. +- 12:25 GMT MySQL CDR logging is installed and working.. Anyone got any ideas?

We currently have a Merlin Legend system. The voicemail is falling apart (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system locked up and refused to take calls; the official solution is to change the system time back to a year with a matching calendar). We are in the process of preparing the network infrastructure to support a VoIP system with Asterisk, but won't be

I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets "stuck" off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being

Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What

How can I loop through something x number of times in the dialplan? i.e. if I get an invalid extension I want to re-play the menu, but not forever. Maybe 3 tries or something. I'm pretty sure that I've seen it before, where you can increment a variable and do "Gotos" based on it. But I've searched the Asterisk handbook, searched the user archives, and Googled for it,

I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/4c2821af/attachment.htm

I am wondering if it's possible to use a bunch of cards in a PCI backplane instead of going out to the extensions with T1 and then and adapter. How are people connecting to large amounts of extensions?

It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030820/4a9e4608/attachment.htm

Hello. I've seen some mentions of asterisk possibly being used as an inexpensive voicemail attachment to a commercial PBX etc. Does anyone here, have experience of using it in this fashion ? What commercial systems have been successfully attached too ? How is the attachment made ? Analog, digital ? If anyone has successfully accomplished this, I would like to hear the make and model of

hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind

Greetings Asterisk Users, I'm looking for some friendly advice on setting up a asterisk PBX for our small business. I've played with Asterisk and setup a soft-phone open323, though even on my ethernet network this showed very poor performance. Got a phone call through to digium, but had a difficult time either hearing (low volume) or understanding (line breaking up). Hoping a hardware

>> Wouldn't that break everybody's dialplans where they would have to >> replace all occurrences of Voicemail2 with Voicemail and all >> occurrences of Voicemailmain2 with Voicemailmain? > > No, we would register with both names. Is it necessary (with reasonably current cvs) to make any changes in the *.conf files to use Voicemail2, or is that happening

I'm using the stable branch. Is voicemail or voicemail2 deprecated? TKS Paul pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040502/1b0ab572/attachment.htm

Hi all: Somebody knows the mysql table structure for VoiceMail2 application? Thanks in advance, Gus

I figured out what was going on with the lack of/stuck on stuttered dial tone. Apparently, there are two voicemail directories being referenced: /var/spool/asterisk/voicemail/default, and /var/spool/asterisk/voicemail/local. The sip phones were using /var/spool/asterisk/voicemail/local to dump VM messages into, yet the MWI looks at /var/spool/asterisk/voicemail/default. Does anyone know why

Here are a few things I would like to see .. 1. In addition to time/date stamps, store/read the caller id info with the voicemail messages. 2. Have the ability to configure the system to ignore and delete messages left by a caller that are 3 seconds or less (maybe make this configurable) Not sure but that would cut alot of hangup calls out of your voicemail box. I can't think of much more

Does anybody have any reason why I should *not* permenantly replace app_voicemail with app_voicemail2? If so, speak now or forever cvs update -D "8/18/2003". Mark

Hi, I am looking at using the softfax that Steve Underwood has developed. It's very straight forward when you assign an extension for the fax. A function that several pbx's has is that they listen for the 'faxtone' for 5 seconds after 'answer' in the menu where you can enter your local extension number, it's normally done in parallel with the DTMF detection. I think

Asterisk Users: I've been working hard on app_voicemail2 which is an enhanced scalability version of app_voicemail. Specifically, its features are: * Highly improved internal architecture (maybe someone else can actually code on it) * Foot print for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies