similar to: Integrating Asterisk into our Legacy PBX <-- Newb (correction)

I have been successful in setting up asterisk and making workstation to workstation SIP calls. But I am lost when it comes to anything past that. We are trying to integrate this asterisk server into with our Executone (432?) PBX to allow us to make outbound SIP calls between our disparate locations. We have a T1 card in our PBX, and the Digium TE110P card in the Asterisk. We have the T1 card

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

Does anyone have experience with melding Asterisk with an older Executone PBX? I have a client whose existing voicemail server(repartee) has become bonkers and we need to stick a VM system in there asap. I thought asterisk would be a good thing to use. Does anyone have experience with the older Executone PBX's and asterisk? Any caveats, any tips, any things I should be aware of? Thanks,

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

Hi, This error I got it just when I gonfigure zaptel support isdneuro 31 channels. But if I configure zaptel to support T1 and just 24 channels I have no problem. ######################################################################## #### # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = it span=1,1,0,ccs,hdb3,crc4 bchan=1-23 dchan=16

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

Hello, I am trying to get a Wildcard TE110P to work in a relatively modern HP Proliant DL385p Gen8 server. Being a potent 12 core Opteron server I expected no problems. Much to my dismay the dahdi_test results are constantly terrible: # dahdi_test Opened pseudo dahdi interface, measuring accuracy... 89.101% 89.195% 89.142% 88.957% 88.953% 89.115% 89.089% 89.134% 89.066% 89.021% 88.933% 89.044%

Hi everyone I have a Digium card TE110P and after plug it, turn on the computer and configure, the LEDs don't light up in spite of what Digium FAQ says about the LEDs: When will the LED's light up on my TDM400P/TE110P/TE2XXP/TE4XXP? ... The TE110P LED's will light up RED when the span is configured and kernel module is loaded. If configured correctly and a circuit or channel bank is

I am having quality problems on SIP bound calls made over the Zap channels. All Sip only calls (Cisco phone through Asterisk to another Sip device sound fine). Our setup looks like this: User --> Executone PBX --> Asterisk Server --> Router --> Internet The user is using a legacy handset that works with the Executone PBX and accesses the server using a button that calls up the trunk

Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5 -- Executing [33220567 at local:1] Dial("SIP/4804-0000001a", "DAHDI/g11/33220567,,T") in new stack == Everyone is busy/congested at this time (1:0/1/0) -- Auto

Hi, I have been using asterisk for a couple of months now and for thee most part, I love it. However, I'm having a problem with the drivers of the Digium TE110P. I have tried both the Debian package and the CVS. I have tried several kernels, and am now at 2.6.11. This has been working before (with 2.6.8.1), but after a reboot it stopped working and I am not able to consistently make

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed

Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341400 at local:1] Dial("SIP/4804-00000000", "DAHDI/g11/21341400,,t") in

Thanks Gene. Here is my localnet: localnet=172.16.64.0/255.255.240.0 Which matches our subnets network address and subnet mask. Are you recommending that I make it more restrictive? Thanks, Geoff > -----Original Message----- > From: Gene Willingham [mailto:gwillingham@comcast.net] > Sent: Tuesday, June 14, 2005 9:13 PM > To: asterisk-users@lists.digium.com > Cc:

Hi, I configured an asterisk box with 1 Digium Wildcard TE110P T1/E1 Card 0 I setup the jumper in e1 position. my zaptel.conf : defaultzone=it loadzone=it #gestione PRI span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 bchan=17-31 # set this to 1-15,17-31 for E1 # Asterisk starts correctly, I see th 30 channels. Anyway I cannot put

Hi, I have to link an Asterisk Box with a PBX Matra 6501. System look like this : E1------Te110P Asterisk Te110P-----E1----Matra 6501-----Phones | | Ip Phones Incoming call from E1 will enter on asterisk, if incoming number is _800n then go to IP phones. In this case no problem. But if it's an another call i want to return call to my

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

As these problems are very time sensitive and frustrating, I suggest you document each change you make and do them one at a time so you can actually know what the problem was and not introduce new problems in the process. Find someone who is on the phone quite a bit and will give you an honest evaluation of the call dropping situation (unless you yourself are experiencing this issue too).

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