similar to: More on"Callprogress"

Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B.

I have following setup Asterisk - T100P -> Adtran TSU600 P + FXOcard -> PSTN line When PSTN line is plugged directly in to analog X100P caller id is received by Asterisk but when I plug it into adtran I'm not getting caller id. Any ideas what kind of setup Adtran TSU600 requires to pass caller id to T100P ??? regards m.

I am sending a call down a E&M wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=>799179,1,Dial(zap/g2,20,D(9179)) exten=>799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered.

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

I have just received a T100P and an Adtran TSU 600 in the mail. I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel => 1-22 ... signalling=fxs_ks ... channel => 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In

People on the list tend to think you can't make many cards work on a regular desktop. If you're willing to wait a couple of week I might have an answer for you. _____ From: Robert Augustyn [mailto:augustynr@yahoo.com] Sent: Saturday, December 11, 2004 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Will Adtran TSU 600 work with *?

Hello, I have just received Adtran TSU 600 with 24 FXS ports. I have installed sucessfuly T100P card. Adtran is connected to t100p with crossover T1 cable. On T100P card I have a green light and on Adtran I do not get any errors or alarms. But I do not get dialtone on FXS ports. Adtran is configured: For Network Timing, fxs ports ore fxs_ls on Adtran. In zaptel.conf: span=1,1,0,esf,b8zs

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

I'm running Asterisk 1.6.2.1 with DAHDI 2.2.1 using a TDM410P. I have callprogress=yes in chan_dahdi.conf because, from everything I've read, it is needed when using call files over PSTN, which I DO use occasionally. I know that callprogress=yes is "experimental" and causes some issues. We've never experienced any problems when making local calls over PSTN with callprogress

Hi, Would anyone happen to have experience with the callprogress option? What I'm trying to do is use a couple normal POTS lines in a queue setup where it will call the queue members to pass the call to them. Of course the problem I'm running into is that under normal conditions the lines register as answered immediately and the caller gets transferred to the ringing line which

Location = US asterisk/zaptel from CVS. Updated last week some time. Currently rebuilding with todays checkout. I have 2 fxo channels hooked up to outside standard Bell South phone lines. If I configure as so [channels] context=pstn group = 1 signalling = fxs_ks callprogress = yes channel => 4,3 Then any call routed from asterisk to the outside line will ring, and can be picked up, but *

It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030820/4a9e4608/attachment.htm

>From what I've been reading about the callprogress option, it seems like it will work properly only with a T1 or PRI in the US. Is that correct or are there still issues with call progress detection even if those qualifications are met? Thanks, Shaun Tierney

The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort

Hi, I have little problem and it is so embracing when u r talking to some one and line get hang-up. When some one calls from out of state or out of country my calls gets randomly hang-up with in few seconds and it happens with most of the calls. It's happening randomly I got few calls, which worked. I have also observer that with quite few call I can't hear the person and person can

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled

I'm specing out a new box to act as a tandem switch. It will have a TE410P with 4 x PRI and support IAX connections to four other boxes using predominantly ilbc and/or gsm. It also has 3 IAX trunks to Teliax for call routing also using gsm. No extensions actually terminate on the tandem, they're all switched to other boxes (highly distributed). On the PRI card, one goes to Embarq, the

Question for other snom 200 users: 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the

i wrote previous about a setup i thought might work with asterisk and the tsu-600. no one replied, so i thought i would ask if anyone is using a tsu-600 with asterisk and if so how do you have it setup ??