similar to: T1 vs PRI

Hi, I just got a T1 interface for a AT&T (became Lucent) System75 (uses same cards as Definity). I would take a crack at making a cable but can't determine the pinout for a cable and it is not apparent from the board. Asking you guys makes sense as one of you may have one of these systems. The cable has a amphenol male D50 connector on one end and probably a rj45 on the other. I also

Hi, We are looking at consolidating our lines with PRI. This will allow the elimination of many fax lines. Some of them will be replaced with this type of config ... PRI * IAX * Channel-Bank FAX We will have daggressor suppressor enabled. Is anyone doing this and should I expect smooth operation? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.

I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten=> definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD

I have a machine that crashes every so often. I believe the following macro is responsible (gotoif,$[${ARG3}] in particular). The macro works as expected: if ARG3 is defined - hop over assignment. But my hunch is that it gradually chews up memory. ; This macro is puts voicemail in an alternate mailbox (if ARG3 defined - otherwise Mailbox matches extension). [macro-stdexten] exten =>

I just called 6610 from Zap/1 exten => 6610,1,Dial,Zap/1&Zap/2 (Don't try this with the handset pressed up against your ear) ...is this the call waiting indicator? Is it a file, can it be turned down - and does it get used anywhere else? These are both fxs ports. In zapata.conf changinging to callwaiting=no stops this. The text implies that this affects fxo ports. John This e-mail

Hi, I'm looking for 4 line phones that will see some heavy usage (they need to be tough) for office secretary/receptionists. Any recommendations? One additional consideration (not particularly for my current needs) is regarding the page/intercom feature that can be mapped to one of the analog ports - does anyone know of a 4 line phone that will do this? Thanks, John This e-mail was

Hi, I am setting up a hylafax server. From what I've read so far, hylafax supports CID numbers and names but currently does not support DID. I assume I can do something like this... [40faxDIDs] exten => _87[5-8]X,1,SetVar(CALLERIDNAME=${EXTEN}) exten => _87[5-8]X,2,Dial(Zap/g${hylafaxMODEMGROUP}) ...and use the CIDName variable in hylafax to route the faxes to the appropriate

Here is something I wrote years ago. I expect you can adjust it for your needs # cat remove_blank_vmail #!/bin/bash # remove_blank_vmail takes arguments as voicemail boxes and removes messages with audio files shorter then MINSIZE (in bytes) #---------------------------------------------------------------------- # Description: # Author: John Harragin Monroe-Woodbury CSD # Created at: Thu Nov 6

Hi, Can I make e&m wink start lines just wait for digits - instead of going to default? Someone else cleared a similar problem (as described below) on an fxo port with "usecallerid => no" but it is not doing the trick for me. In this case the line when straight to default which would be ok also. John I posted the stuff below about a week ago... I set up a t1 from my sys75 to

I would like to access VoiceMailMain2 skipping extension and password prompting if calling from a resource that has a mailbox defined. What variables can I use to retrieve the calling channel & calling extension (if it exists)? Here is what I'm trying to accomplish (of course ${CallingResourse.MailBox} is not a real way to retrieve this info)... exten =>

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.

Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I

I think the solution may exist. The compatibility of mysql-connector-odbc with maria may just means the driver can access the mariadb - but my experience suggests not live on the same host. maria has its own connector: https://downloads.mariadb.org/connector-odbc/ it does not look like this is in the sig, so I'll have to turn to the maria repo. I'll try replacing the driver first, then

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 with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

I've been curious how IAX has been working for users running without a jitterbuffer and no echo cancellation. In my particular installation, IAX2 traffic remains within a pretty high bandwith WAN ( - but busy as it is linking buildings in a school district). It seems that at most times, things could run quite nicely in this mode. The ability to dynamically disable echo cancellation when end

Thanks for each of your inputs. It was not a configuration issue as odbcinst.ini does not reference the mysql subdirectory. Rather than use Alexander's url, I ran: yum -y install centos-release-openstack-ocata yum -y install mariadb-server ...the cloud repository provides the properly pathed file. # repoquery -l mariadb-server mariadb-libs | grep lib64 | grep libmysqlclient

Are there any digital phones that run on asterisk yet? I'm talking about non-IP phones here... John

I have a PRI comming into each of 2 buildings. How do I redirect an incomming call on PRI_A of particular DIDs to arrive at PRI_B instead? Thanks, John

callgroups & pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John

Out of necessity, a friend of mine and I have created a patch for syslinux to allow it to modify the mbr and partition table in windows. It adds a -m option to syslinux. If given, it will overwrite the MBR of the drive specified with the mbr.bin provided in syslinux, and if the bootable flag is not set on the partition being syslinuxed, it will set it. This was done to distribute with my USB