search for: tsu600

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.

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

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...

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

Here is some more stuff to add to the confusion about the "callprogress" option. I currently have my * system operating with a T100P talking to an ADTRAN TSU600 channel bank with 8 FXO ports connecting to the outside world and Grandstream SIP phones as handset extensions. At first I naively set "callprogress=yes" in zapata.conf. The results were typical of what many people have reported in the lists: I could receive incoming calls (to the S...

Hi folks, I'm still having the following problem, maybe someone can help me out of it. Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *) communicate through IAX2. Everything works ok on machine 1. On machine 2, if I try to use 4 fxo's from a TDM400 card, sound gets lousy. If I manually destroy one of the zap channels (e.g. zap destroy channel 4), sound gets good again.

We are using Voicemail 2 and use the e-mail with WAV file attachment option. When we record a message from an internal extension, the WAV file sounds fine. However, if a caller from outside leaves a message, the volume on the WAV file is so low it is almost useless. When we access the voicemail through an extension, however, the volume sounds fine. Has anyone had this problem or any idea of a

Great idea Dean, "I would also suggest that maybe instead of looking to set up within the disaster zones that you consider setting up in the relocation zone (eg where people are being sent to) yes they have payphones there but having a bank of 20 grandstream phones connected a T1 is a far smarter and more effective solution and probably more meaningful." > -----Original Message-----

Everyone-- Here's a cost analysis, rather crude and inspecific, of using Asterisk to implement a phone system. I'm really quite naive and new to all this, so I'd appreciate any corrections, tips, pointers, etc, from those in the community, who have far greater experience, knowledge, skill, etc. than I. Am I forgetting something important? Am I way off in my estimates? The Minimum

Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN

...$500 Asus A7V133, 900MHz, 256MB,60GB 2) Two Nics $50 3) Digium T100P $495 4) 20 GS Budgetone @$65 $1300 5) Adtran TSU600 Channel Bank (e-bay + patience) $99 6) 2 Dual FXO plugins for TSU @$100 (used) $200 7) 1 Dual FXS Plug in for TSU @$100 (used) $100 8) Asterisk and Linux $0 9) APC Smart-UPS 70...