similar to: Channels Banks that support neon MWI

Hello all, long time lurker, first time poster. I have a situation that has been confounding me for the better part of a week. I run an internal mirror of a Centos mirror, mainly because I have lots of hosts and low bandwidth. For my 7.x hosts, since 7.5.1804 was released, when I do a ?yum update?, I get the following errors on a number of files. "Package does not match intended download.

Does anybody know if this is supposed to work and if so, what, if any, workaround is needed? I have other phones (Snom, Polycom) MWI working with this system fine. 6.0.0.19 (latest) Mitel SIP firmware is loaded. Thanks for your time, - Jesse -- Jesse Peterson <jesse.peterson@exbiblio.com>

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

Some days ago there was a subject regarding BLF (SIP Phone-receptionist Setup). We are the developers of a Price Verify Terminal for a French company. We have developed the hardware (small board based on a PPC 823e), working with Linux embedded (based on Wolfgang Denk's work). I think that it can be a good BLF. Probably it is possible to integrate the Nicolas's FOP or a new application.

My Setup looks like this: Mitel 200 SX (1st T1) -------- Bell South (2nd T1) | | | Digium TE110P Asterisk MITEL CONFIGURATION Primary Timing Source: 1st T1 Card Secondary Timing Source: 2nd T1 Card ASTERISK CONFIGURATION span=1,1,0,d4,ami (Look to the Span for timing) We are getting a lot of Frame and Slip errors.... Time Slip Frame 7:00 736 950 8:00 690 1200 9:00 437

Hi, To me it looks like you are getting incomplete content from your rsync source. Try to sync from another mirror. hth -- Sent from the Delta quadrant using Borg technology! Nux! www.nux.ro ----- Original Message ----- > From: "Eric Germann" <ekgermann at semperen.com> > To: "CentOS mailing list" <centos at centos.org> > Sent: Tuesday, 15 May, 2018

I've configured an analog/single line off my mitel sx-2000 to a x100p card in my *, however when the remote caller drops the line I get a dialtone back from my pbx and the x100p doesn't detect the end of the call. Any way I can tell * to drop? I've tried call progress but it doesn't seem to work...

We have several agents on queues, and want to indicate to them that they are logged in or logged out. We have tried several different ways, from changing the screen to presenting different service menus, but cannot get anything to be "in their face" (their words, not mine). One of our team has suggested, as the agents do not have voicemail, is to use the MWI on the 7940 phones to

I am still a little confused about how to activate MWI with PJSIP on Asterisk 13.9.1. I use realtime for configuration. So far I have tried setting both the mailboxes field on ps_endpoints and the mailboxes field in ps_aors but I cannot get the indicator lamp to blink on any of my phones (Digium, Aastra and Yealink). I have tried just the number of the mailbox and also adding the context.

I've had a (small) working asterisk with X100P and 4xFXS TDM card for a while now. I've been running the release from (I think) April 13 - asterisk 0.9.0, zap* 0.9.1. I tried 1.0 RC2, and my internal FXS line doesn't ring - call comes in the FXO, gets answered, and rings through to the FXS extension, but the phone doesn't ring - asterisk shows it ringing and eventually punts to

Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I

Our setup is: 9.6k Modem <-analog-> Mitel SX-200 <-(pri)-> Asterisk <-(pri) -> Telco The modem works fine with the Mitel directly connected to the Telco, but once we add Asterisk in between connections start failing. I suspect the issue is caused by the echo canceller, since I believe the issue appear about the time we turned echo cancellation on (for the IAX users). We

Hi all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards: a TE420 w/Octasic and pri_net

We have a customer with an obsolete Centigram voicemail system who would like to replace it with Asterisk. Any one with experience doing this or information on the signalling and trunking used to connect the Mitel SX-2000 to the Centigram server? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

Hello all, The wiki has a fairly detailed description of the the issues involved with encryption of Asterisk calls: http://www.voip-info.org/wiki/view/Asterisk+encryption I'm interested in hearing what is working for people today. I think the ideal solution would be a hard phone that could be plugged in almost anywhere (dsl/cable modem, hotel, etc) and connect securely to a remote

Thanks. Is there a way I can log into the Merlin Magix to determine that? How else do I tell? ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Germann Sent: Monday, February 05, 2007 8:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users]

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

Eric "ManxPower" Wieling wrote: >> >>> span=1,1,0,d4,ami >>> e&m=1-24 >>> > > Looks like you have told Asterisk to get it's timing from the Mitel. > I'll bet the Mitel is trying to get it's timing from Asterisk. > > Try span=1,0,0,d4,ami and run ztcfg -vvv > I just set this back. It was originally set to your

I just registered an Avaya 9620 set to my Astlinux system (0.47 - Asterisk 1.2.22), using Avaya SIP Firmware version 2.0.1.34. Set mailbox=ext at default in the sip.conf Found MWI worked immediately. Turned off as expected. Have Fun! Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: