similar to: Weird Zaptel sound after anwser calls

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

I have a PRI line and I am having problems setting the ringtimeout on the dial application to more than 29. If I set ringtimeout to 29 on the dial application call and I do not answer the ringing phone then I correctly get DIALSTATUS set to NOANSWER. If I set ringtimeout to any value over 29 on the dial application call and I do not answer the ringing phone then I go to extension h and have

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

I've been using my Asterisk (At my house - 2 modem-type fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and I've been really happy with it, but I'm still having an echo problem that I've exhausted google with, and can't get straight... I think I've determined that because I'm using $7 voice modem clones for my FXOs that bad echo is going to

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

Hi all, I will a TDM card with FXO modules on it. Below is the dial plan. When someone can 9123456, CLI will show dialing to 123456 and answered. After answered, the call hangup. I would like to know what will cause the case to happen. Anyone can give me some advice to solve it? exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT}) exten => _9X.,n,Hangup zapata.conf

Hello all, Recently I bought a TDM02B digium card to conect to the PSTN. We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try to start asterisk, the box hangs. Someone have the same card running with asterisk in a similar machine? Could you tell me your box hardware details? Thanks for your time, Ismael Gil.

I have server with a total of 6 Analog ports...using TDM04B and TDM02B. I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have worked through getting the DIDs to work and route to the extensions...now what I need to do is when Extension 1111 picks up the phone to dial, I would like them to use their DID analog line first, unless someone has called in on it and they are trying to

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

hi folks, I use Dial(Local/extension@context) to make calls received on my DID number to ring a local extension. I notice that on 8 out of 10 calls, the audio is NOT working in the incoming direction (DID provider to asterisk). Local extension 2055 maps to SIP destination "homephone", and if i replace the Dial(Local/2055@local) with Dial(SIP/homephone), it works fine 100% of the

In Australia, Telstra, the local telco provides isdn modem for isdn connection. The modem has 2 analogue telephone jacks and a serial port for connection to dialup internet. My question is that will it be possible to use Zaptel TDM02B to connect to the analogue jack instead of getting a fritz card to do the telephony. Will there be less feature if doing so? -- David Kwok, CISSP Tel: 612

Hello all, Anyone expecialized with call center monitoring and reporting solution based on asterisk. A client of us, want to install a call center reporting solution for an asterisk server but I do not know which could be the best tool for that. I need a tool for reporting queue calls, agent calls, and disconnect cause. Any clue will be appreciated. Thanks in advance. VoipCrazy

Guys, I need to use Asterisk to call out PSTN numbers via an analogue line. I understand Digium manufactures these kinds of cards, but can someone tell me which model number it is. I really only need a card with one or 2 analogue ports max. Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good day , I have configured TDM410P (asterisk 1.6.x) on Cent OS 5, but dahdi take some time to hangup the call when playing the IVR..(it will send the hangup signal after finishing the IVR promt..) is there any specific setting to avoid such incidents ? iam using busycount as 3, signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes

Has anyone had any echo issues with the Digium TDM02B FXO card? I purchased a clone wildcard on E-Bay for $8.00 and have had horrible echo issues.... I'm assuming it is because of impedance issues? Just wondered what people's takes were on the TDM02B FXO and echo?

Any help on this would be great I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune

Hi all, I am looking for phones witch support POE, with a good relation between quality and price to work with asterisk. I just see the Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones gave you the best results in a productivity enviroment? Thanks in advance. VoipCrazy. -------------- next part -------------- An HTML attachment was scrubbed... URL: