similar to: PRI Dialing failure?

I've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information

I must be doing something stupid, but I can't figure it out. I have three PRI lines connected to Asterisk, one from the phone company, and two more connected to PBXs. Asterisk uses the incoming DID information to decide which PBX to route the call to. Should be simple. Asterisk is clearly getting the caller id info from the phone company: -- Accepting call from '512345xxxx'

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

Anyone have a suggestion on where I can get a decent new MB with 5v capable PCI slots. It seems like every decent server MB on the market has 3.3V slots only. Is diving into the junkbin my only choice if I can't afford to replace the 5v quad-T1 wildcard? Thanks Mark Farver

I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows...... < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000 EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 025 P/F: 1 < 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

I have a pair of sites tied together with a T1 line running zaptel PPP on either end. The system works, but I keep getting these messages in the kernel logs, and users are reporting connection problems (TCP timeouts, and hangs) especially under high usage. --snip-- HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

I need to check if a parameter is set so that I can build some information for my application, but No matter how I format my if statement in the file " Application.rb" it return the following error. You have a nil object when you didn''t expect it! You might have expected an instance of Array. The error occurred while evaluating nil.[] Here is the line if params[:day] So we

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110621/764a6fa9/attachment.htm>

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

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 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

I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user

Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. Sorry for asking here. Siemens-related websites use "salesperson language". There is no technical information.