similar to: DID from an analog phone

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 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 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 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

Of course I am not a kernel expert, so .. please be patient. I am investigating on my zaptel/zapata problem. As the main error message asterisk quits on mentions <'/dev/zap/channel': No such file or directory> I went peeking over there. [Asterisk Verbose Error Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or

hi all, I have this anoying problem with Direct in Dial, when someone calls from an extern analog phone to my isdn number, Asterisk is not waiting for the DID numbers the caller dials after the main-number. Version: Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n, with junghanns quadbri zapata.conf: switchtype = euroisdn signalling = bri_cpe faxdetect=incoming pridialplan = unknown prilocaldialplan =

Hi all. I am seeing incoming calls from digital lines (mobiles e.g.) dialling my main number + 3-digit extension just fine ("Accepting voice call from '11234567' to '250' on channel 0/1, span 1"). The problem however is with calls from analog lines: "Accepting voice call from '13331846' to '25' on channel 0/1, span 1" * just sees 2 digits, not

When I try to start up zaptel, whilst running ztcfg, I get the following error: Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) My /etc/zaptel.conf is: fxsks=1 fxoks=2 loadzone = au defaultzone=au Channel 1 is a X101P card connected to the PSTN and channel 2 is a S100U box driving an analogue phone. The zaptel kernel module gets loaded OK as does the

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

Hi, FYI, Red Hat released an advisory today about a vulnerability in Ruby. So far it doesn't appear in the VuXML, but am I correct in presuming it will soon? https://rhn.redhat.com/errata/RHSA-2006-0604.html http://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2006-3694 cheers, -- Joel Hatton -- Infrastructure Manager | Hotline: +61 7 3365 4417 AusCERT - Australia's national

Hi everybody, i've a E1 connection with 30 phone numbers, i'm using freepbx (trixbox) with TE110P card, when i call my sell phone just the first number of E1 is always showed evenif i set up different trunk with different zapchannel (g0,1,2,3...) for example when i type 1MyPhonenumber , i get the Head of line 555-55555 (the trunk is g0) when i type 3Myphonenumber, i get also the same

Seems to me like the negotiation fails for some reason. Maybe you are trying to use a callerid that isn't allowed? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com genom Gary Richardson Skickat: on 2006-02-01 21:45 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: [Asterisk-Users] Re: CallerID Problem No, I'm not

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

Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk

Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continues in use, even after hanging the call up, then i need to run the "soft hangup Zap/<zapchannel>" in the asterisk CLI to release the channel. Here is my zapata.conf: [trunkgroups] [channels] language=pt_BR context=default usecallerid=yes

Hi, On behalf of one of our users, I installed R 1.8.1 on HP-UX11.0 by compiling the source using the gnu compilers and all was fine with make check On 1.9.0, I get an error from 'make check' when running reg-tests-1.R which gives error exit code 1. Commands used are: ./configure --prefix=/apps/global/Gnu_R/R-1.9.0 --with-x R_BROWSER=mozilla (all on one line, of course) make make

hello, I have a asterisk box (Slackware 9.1.0, Linux 2.4.31) connected to a Ericsson Businessphone PBX on the internal S0 bus with HFC-S card and zaphfc driver - point2point mode -. --------- | TELCO | | BRI | --------- | | PBX external S0 -------- | PBX | -------- | PBX internal P2P S0 NT Mode | | HFC-S Card P2P TE Mode

Hi, Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and

Hi all, I really hope that you guys can help, because I've been tearing my hair out for the past 5 hours on this one. I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel Meridian phone system. Phone calls from the Nortel to say MSN 510 are correctly being sent to the right SIP phone. When asterisk dials say Zap/g2/224 (a Nortel internal extension) the call goes