similar to: Outbound ZAP Dialing

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 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 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 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'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

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.

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

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

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?

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

I'm using a T100p to interface to a channel bank and from there to analog PSTN lines. Because of my particular setup I have to do post-connect inband DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming 0.5/sec per digit (ie. using "zap/g1/31|5|D(6045551212)". Even with an 'outside transfer' voice prompt before commencing dialing my users are getting

> -----Original Message----- > From: Adam Goryachev [mailto:mailinglists@websitemanagers.com.au] > Sent: September 7, 2004 8:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Monitored outbound dialing via Zap > interface? {clip} > Have you considered adding the r option to the Dial command, so they > might hear ringing

Kevin, Are you in the US? If so then you've probably got several carriers to choose from. In my experience analog lines have a flat expense of $20-$25 per month. That equates to about $140-$175 per month in flat fees, plus you have usage on top of that. (Your experience may vary.) I am currently experimenting with a company out of NY called Digizip (www.digizip.com) that sold me a Qwest

Hi all, I have done a major upgrade, first I have bought a Digium TDM400P (TDM02B) 2-Port FXO to replace x100p cards, than moved from an old version Asterisk to a new centos 4 box, compiled asterisk from Asterisk CVS-HEAD built on 2005-06-11. Seems that my iaxy is now "incompatible" with the TDM400P. When I receive a call from the TDM02B, the caller can't hear my voice (I am using

I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2. There is any work around or different setting to avoid

We have 4 analog line and 2 analog trunks. On the trunks we have all the DIDs coming into the current phone system. Trying to get everything moved over to Asterisk but having issues picking up the calls on the analog trunk. We can receive calls on the plain analog lines and we can call out on all analog lines and analog trunks. When a call comes in on the trunk line the ZAP channels don't

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels and the first

Hi, Can someone point me to a zapata.conf example that will create a single DIAL OUT group including all 4 spans on a TE4XXP? One friend says to change the group number all to "1" on all 4 spans. Another suggestions says it is possible to have these unique groups (1-4) and to combine all 4 into a single group "5". I like the second suggestion best. Can you guide me to the

Greetings list- I have a box with a single FXO card in it. I'm able to dial out ZAP/1 with no problems and as expected. However, I would like inbound calls on that POTS line to go unanswered by Asterisk since I have other equipment on the line. I've setup zapata.conf for the channel without a context but the line is still answered. I've also setup a blank context with the same result.