similar to: Migration to Asterisk - Running off of Merlin Legend system

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

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

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

Hi List, Merlin Magix hardware v02 I'm trying to get asterisk to act as a voicemail server for a lucent merlin magix PBX that we purchased used. We have 4 FXO channels between the two PBXs on a Sangoma A200 card. The 770 dialgroup is working properly, in that calls to 770 are answered by Asterisk. The magix is sending mode codes in the format #XX#XXX#, where the 2nd block of digits

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?

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

Management just approved purchase of a Digium T100P and a T1 card for our Merlin Legend Switch. I will appreciate comments from anyone performing this installation before: - Which T1 card did you use in the Merlin Legend? - Did you require any special interface? (CSU/DSU, etc) - Any items to watch during installation. Also we will need to remove one of our old analog Trunk cards to accomodate

I have a legacy pbx that I know nothing about. The PBX is a merlin legend system (now owned by avaya). I have found information on the list that I can use a 100D T1 card to bridge the PBX to an asterisk system. We are looking to head this route as well as replacing are AUDIX VM system with Asterisk. Has anyone done this that wouldn't mind sharing some of the merlin specific setup?

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

I was just about to replace 2 Merlin Legend systems (one in my house and one in my Parent's house) with * systems. I have 2 beefy linux systems ready to go and have enough Cisco 7960 and 7970 phones to replace all my MLX-10D, MLX-16D and MLX-20 phones. Then I discovered that * seems to be lacking a critical feature for replacing a legend (or any standard PBX) and that is an ability for a

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

swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, > visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message

Folks, I have a Merlin Legend R7 V10.0 with a 2 100D cards. I have 1 card in slot 4 going to CenturyTel, and the card in slot 10 going to a flip cable to a TE110P card in a Asterisk 1.6.x box. I have routing setup on the Merlin to send a block of numbers to the Asterisk. Currently the PSTN can dial the Asterisk Extensions. The Asterisk can dial Merlin Extensions. The Merlin can Dial Asterisk

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

I have asterisk running as a voicemail system off of our Merlin Legend switch. We replaced our old Audix Voice Power (when the power supply fan died and burned it up) with asterisk a week ago. Many thanks to those who provided information about integrated VMI on the legend. The Audix system would, after a mailbox was closed, wait a few seconds, then use that line to dial the switch and update