similar to: Voicemail to a commercial PBX/key phone system

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 want to tag categories to its menuname. i have a csv containing menu item name and in other csv i have a column containing some strings, i want to pick that strings from categories and look into menu items if any menu item containing that string i want to create a new column next to menu item name flagged as 1 otherwise 0 and the only condition is once a menu item flagged as 1 i don't need

Hey All, I'll be configuring an asterisk box to be the voicemail server to an old Merlin system which had an octel 100 voicemail server that is now dying. My question is simple: do I need to stick an FXO card in the asterisk box? My logic is that if the Merlin Magix system is actually generating electrical current, then I would need to have an fxo card. Is this correct?

Hi all, I know that this has been passed around before, and I know that it happens about every 3 months or so, but evertime the answers change, so I thought I would pass it around again. A company I work for has 3 incomming lines and 4 phones. They require voicemail and MOH. Their phone systems VM hard drive died today, they were quoted a $2000 to replace it. I started to talk to them about

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

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

Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways,

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

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

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

Hi List, I have a question about uninstalling and installing R on linux, which I am new to. > sessionInfo() R version 2.10.1 (2009-12-14) x86_64-unknown-linux-gnu locale: [1] LC_CTYPE=en_US.UTF-8 LC_NUMERIC=C [3] LC_TIME=en_US.UTF-8 LC_COLLATE=en_US.UTF-8 [5] LC_MONETARY=C LC_MESSAGES=en_US.UTF-8 [7] LC_PAPER=en_US.UTF-8 LC_NAME=C [9] LC_ADDRESS=C

List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is

I've got a piece of CPE equipment that has an FXS port that I have tied to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a standard telephone connected to it, I get a dialtone. If I dial a digit, and send a hookflash, the device will provide a dialtone back for the next available channel on the device. I'm trying to recreate this same behavior with Asterisk,

BellSouth (now AT&T) has a number you can dial and it will play back voice prompts with your calling number? It's used by their techs with a buttset in identifying analog 1FB lines... Eg Dial 704-210-3233, it answers "Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two" if your dialing from 704-559-2122 ----------------------------------------- Disclaimer: This e-mail

Hello, Actualy I have got a PBX conected to a BRI port on a Cisco Router witch route the calls. The router is using h323 with the other end. How could I replace the actual PBX and pass all the calls to the router, as now works. I tryed connecting an asterisk CAPI port against the same BRI Cisco port, but I can not make any calls, I get the following error from asterisk. Connected to

Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- While you don't greatly need the outside world, it's still very reassuring to know that it's still there.

Greetings. I am attempting to configure a system based on Asterisk 1.2.3 to be used as a backup should our aging voice mail/auto attendant system fail, which seems increasingly likely given its advanced years. The first part of this task is getting the auto attendant feature to work correctly, which I would have figured to be relatively easy. I have successfully built a menu structure, but cannot

Hi all, I upgrade my rsync from 2.3.1 to 2.6.6 today in the client host, everything seems not working after the migration. I didn't touch the rsync server. Herewith is the log from my rsync server. I looks like something related to IPV6, anyone has any idea? Rsync Server : RHEL4 with rsync 2.6.3 deamon running Rsync Client (merlin) : Solaris 8 with rsync 2.6.6 2006/05/09 16:18:40 [5332]

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

On 03/08/2017 11:00 AM, Paul Heinlein wrote: > On Wed, 8 Mar 2017, Mark Weaver wrote: > > > Hello all, > > > > I've been googling my brains out since yesterday looking for up-to-date > > information on this matter, and have found information that is > anywhere from > > 15 to 5 years old. I'd really like some information that much more > up to