similar to: DTMF issue --help

I'm connecting to an NEC t1 card via t100p (working great so far!) however I'm having problems dialing from the NEC system to an asterisk extension (sip-grandstream). If I hit the trunck line and dial REAL quick 103 I get the sip extension ringing; if I don't I get an invalad selection message from asterisk - and I can see on the console only one or two digits arrived. How can I

Hello, I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test,

Thank you Peter for you reply, I realize this problem occur because I take the CVS head (maybe a bugs get introduce), because when I rebuild using the checkout of the latest stable version (cvs checkout -r v1-0), I don't have the problem of detection of DTMF over a direct ISDN pri link. Hopefully this problem with be fixed before the next release of asterisk. Sylvain. -----Original

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

OK, this is probably a dumb question for a lot of you, but I have no experience with digital lines outside of a tiny bit of ISDN, so I'll just bite the bullet and ask some newbie questions. I am attempting to plan an asterisk installation with about 20 SIP phones and the following incoming lines: 1) At least 6 (as many as 10) lines for voice to the SIP phones 2) 2 incoming/outgoing fax 3)

Has anyone noticed a delay in sending DTMF to zaptel devices? I have a T100P connecting to an NACT telecom switch. All calls are sent just fine, but there are 4 seconds of delay between when the channel goes off-hook & the digits are collected by the NACT switch. I also have some cisco 5300 boxes connected, but there is no delay. Any ideas? -g

Eric Wrote: ----------- The trick is not to use options you don't understand. "show application dial" will show you what the t and T options are for. Most people use the transfer feature of their phone, rather than using the T/t hack on the Dial line. Sounds like you are using CVS-HEAD and so will have to configure stuff in /etc/asterisk/features.conf. /Snip/ Eric, Thanks for

Please point me in the right direction as to where to increase the length of time I am sending DTMF tones on my T100P. I can't remember where/how I set it last time, and I need to do a fresh install. I'd sure appreciate it :) Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977

I'm having a weird problem with IAX2 in today's CVS HEAD. I have two boxes with T100P cards connected via IAX2. Calls between them work fine, but when I press a key at one end, it comes out the other end as a click, with no tone. I've tested the DTMF on the T1 using SendDTMF with an outgoing call, and it sounds fine in that case; it seems to be only IAX2 that has the problem.

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

I'm having the same kind of problem too. Links to other excel workbooks are really breaking badly. One of them so bad that if I highlight the file through a windows explorer view; it kills the window immediately. I'm using RH 7.2 and Samba 2.2.4 -1 Please let me know if someone has some suggestions .... So far I've tried: veto oplock files = /*.mdb/*.MDB/*.ldb/*.LDB/*.xls/*.XLS

I'm looking to buy a channel bank. I'll be using this to basically dual a Toshiba PBX and Asterisk so that we can deploy Asterisk in small steps and have the Toshiba as a fallback in case something goes wrong. What I need is a channel bank that will allow me to take in 12 lines using the FXO ports in the CB and then output those lines to the Digium T100P. Has anyone had experience

Hi. Last week I installed asterisk 1.2.7.1 on a machine running Fedora core 5 with kernel 2.6.15 Today we installed a t100p digium card and I installed libpri, zaptel and rebuilt asterisk with make all and make install. Now after rebuilding asterisk, there is no audio coming back from any IVR apps. We have tried background, agent logins and voicemail and nothing works. Audio works fine

I'm experiencing rather severe problems with 1.0.9 (we've had to backrev to our last version we know works (1.0.5). We are running a single PRI line with a T100P card. After about 10 hours of asterisk running and the modules loaded we start hearing noise and "stuttering" on any call that passes over the PRI line. I've tried this with echo cancellation on and off with

I have two asterisk servers. One is at location 1 and the other is at location 2. What I am trying to do seems straightforward. I want the Asterisk server at location 2 to send all it outbound calls to the Asterisk Server at location 1. Both asterisk servers can dial each other using extensions without a problem, but when users on Asterisk server 2, dial 9XXX-XXX-XXXX the call never reaches

I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime

Hello All, I?m on the technology committee for a fraternity at the University of Illinois. We?re looking into moving from our current ?party line? (one line shared between every two rooms) system to a PBX with voicemail in an effort to lower our monthly phone bill and provide better communication services. We?ve pretty much settled on Asterisk as we do not wish to rewire all of our pots

Hi, Maybe someone has seen this before... I've installed a new T100P, but it doesn't seem to work. I've attached the T100P to an Adtran 750 using a crossover cable. The Adtran shows a red alarm on the T1 interface. The Adtran has been set to factory defaults with FXS cards in 1-3 and an FXO card in 5. The T100P shows no signs of life--the leds are not lit. Should I be seeing

my target is to produce a linux router with T100p to Fract T1 24chan to ethernet and TDM 4channel FXO. I have not been able to get nethdlc to work, just get zaptel networking not configured if I set bchan and dchan the card comes to life, but I don't believe this is correct for what I want to accomplish. according to markster@ this is the way they do it. using (multiple distro versions

List, I have a TE410P (T1 mode, all PRI) and a T100P (fxoks, for fxs channel bank). I cannot seem to get the T100P to send any clock to the channel bank. I prefer that it use the same clock source as the TE410P, but it doesn't matter if it's not in sync just as long as it's there. The TE410P is configured 3x pri_cpe, 1x pri_net. The three cpe go to XO Sonus switch, the net to