similar to: Merlin Legend

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

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

I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of

Hello all. When I use outlook express and try see the printer, it says "print$ on Merlin - Paused", and I can't unpause it, I get: "You do hot have permission to modify the settings for this printer. If you need to change the settings, contact your network administrator." Merlin is the name of our samba server. Also, when I try click on properties, nothing happens

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

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

Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? I have been diligently investigating converting our small (Ontario, Canada) office to Asterisk, and ditching our Avaya PBX. However, our head office (New Jersey, USA) maintains our AUDIX system, and a) have no intentions of leaving it and b) some users rely upon AUDIX's ability to transfer

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

I am having a problem with the Vmail Web app and was hoping someone has some advice. I have been able to install vmail.cgi. I am able to bring up the vmail web page and login with out problems. However, I never see any vmails listed even when there are vmails present in the directory [root@linux-home INBOX]# ls -l total 40 -rwxrwxrwx 1 root root 255 Dec 2 14:45 msg0000.txt

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

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am stuck in a situation where I have two te110p's and really need a te210P. Anyone interested in a trade? They are in working condition guaranteed. If this is the wrong forum for this, I am sorry... I will take it off list. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird -

What and When is the next version of Asterisk? /Obelix

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have a 1650 running successfully in production mode with 2-4 PRI's? I want to make sure I don't have a motherboard compatibility problem before I buy one of these. We are going to be using a Digium TE210P to start off with and probably moving to the TE411P down the road aways? thanks, Sean -----BEGIN PGP SIGNATURE----- Version:

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ok... I am having a serious brain fart this evening. IIRC, the next sip draft addresses shared lines and I thought I remembered something on the list about support for it in the near future. I also thought that chan_sccp supported it with in asterisk. Am I loosing it here? Is there an implementation for shared line support in asterisk? I know that

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]

On 03/08/2017 06:21 PM, Alexander Dalloz wrote: > Am 08.03.2017 um 22:57 schrieb Mark Weaver: > > On 03/08/2017 11:00 AM, Paul Heinlein wrote: > >> On Wed, 8 Mar 2017, Mark Weaver wrote: > > >> I wrote this article years ago: > >> > >> https://www.madboa.com/geek/sendmail-auth/ > > > Hi Paul, > > > > I followed your guide to the

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