similar to: Legacy PBX Integration and Zaptel.conf Timing Source

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

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

I have a problem which I need to resolve. We are trying to put an asterisk between a Mitel PBX and the world. We are adding Voip service via Asterisk. Here is are config files for the settings but our problem is the following. We are able to send calls to the Mitel pbx and it's the T1 connections is green saying it's ok. The support department from Mitel said that they use e&M and

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

When I start Asterisk, I receive these errors: Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on peer 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'zoom_trunk' without zaptel

Say I had a crossover cable that connected a Mitel SX200 to a TE110P and the pinout was done as such: 1 - 4 2 - 5 5 - 1 4 - 2 (the 5 and 4 are transposed on the left side) Instead of the proper way of: 1 - 4 2 - 5 4 - 1 5 - 2 What would the results be? We have had the former as our cabling for a few months and the connection has been fine. Slip errors here and there. But we have had major

How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration? We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P. Twice today (first time in over a month) we received a Yellow Alarm on the TE110P. I have been able to clear it easily by restarting zaptel. Thanks in advance! -------------- next part -------------- An HTML attachment was

We are trying to debug a connection between Asterisk and a legacy PBX (Mitel SX200). We turned on the Zaptel debugging and we get the following message quite frequently: Oct 12 07:14:09 localhost kernel: T1: Lost our place, resyncing ( 28 ) Oct 12 07:14:09 localhost last message repeated 3 times -- Oct 12 07:14:11 localhost kernel: T1: Lost our place, resyncing ( 28 ) Oct 12 07:14:11 localhost

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

I'm setting up an asterisk server to extend several extensions from a mitel pbx. I'd like to display the caller id that I receive from t he mitel pbx on the sip phone. The mitel PBX person has setup the PBX to send be callerid, but I don't see it. I've set chan_dahdi up with usecallerid=yes cidstart=ring cidsignalling=bell callerid = asreceived cid_rxgain = 0.0 Are there

Correction: The hardware is a Wildcard T100P (not a TE110P) Thanks! > -----Original Message----- > From: Geoff Manning [mailto:gmanning@zoom.com] > Sent: Wednesday, May 18, 2005 9:07 AM > To: Asterisk Users (E-mail) > Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX > <--Newb > > > I have been successful in setting up asterisk and making >

Hi, All! Hopefully someone can point me in the right direction. I have a Mitel SX2000 Light that I am interfacing with my Asterisk box over a T1. I can receive calls on the Asterisk box without any problems, but when I call an extension on the Mitel, I heard the ringing from the IP phone, but I don't see anything on the Mitel, other than channel 1 of the T1 showing as "Locked

I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72

Hi everyone, We have been working with Asterisk for a while now and would really like to expand its capabilities by fully integrating it with our Definity g3si and are wondering about other peoples experiences with similar setups. Thus far we have only been able to achieve a partial 1 way integration, but ultimately would love to route inbound voip calls to the asterisk out of the lucent.

Hi, A potential client (hotel) has a Property Management System that talks the "Mitel" protocol to their current Mitel PBX in order to receive CDRs (which end up being rated by the PMS system and charged back to guests). Does anyone know of any (free or otherwise) docs on this protocol, or better still have experience interfacing asterisk in a hotel situation like this? The PMS

Dear all, I am having a data for 2 different treatments with different time points. So, I used the following code to plot the boxplot and also to do anova. T11 <- c(280, 336, 249, 277, 429) T12 <- c(400, 397, 285, 407, 313) T13 <- c(725, 373, 364, 706, 249) T21 <- c(589, 257, 466, 248, 913) T22 <- c(519, 424, 512, 298, 907) T23 <- c(529, 479, 634, 354, 1015) obs <- c(T11,

I have a small government department that wants me to implement a Asterisk installation, however, they connect to the Government PBX, a Mitel SX200, and want to keep the ability to do that. I know there is no chance to connect the digital extension lines, but would it be possible to have the pbx admins send analogue extensions over and have those lines interface through an FXO interface? Or

We have had quite a bit of success with our T100P and TE410P cards interfacing to Nortel Meridian PBXes and also to a Livingston Portmaster 3 using ESF/B8ZS and various combinations of E&M wink and ISDN PRI (usually in 5ESS mode). In the near future, I may also need to interface to a Definity via a T1. I was planning to use PRI--is that an option for you on your switch? -----Original

Hi all, I have a user with an old Mitel PBX connected to a couple of SPA112's. The user is reporting that their phones ring several times a day and when they answer the "call," all they hear is dial tone or busy signal. Their PBX guy says that the SPA112's aren't providing line supervision and the PBX requires it. Does anyone know how to fix this? I'd also like to

I see from the archives that Siggi Langauf was wanting to do exactly what I want to do back in November 2003. Here is what he asked: I would like to do a pilot with some legacy gear, however. Accordingly, I'd like to be able to have * dial 1000X where X is the box that has a new voicemail message and 1001X when the user of mb X deletes the new message(s). The dialing should occur