similar to: Analog Telephone Adapter

My office is investigating using an Asterisk PBX and also going to a VOIP provider for our main phone connections, but one of the tricky things is that we need to have outbound and inbound modem calls (fax too). I see a lot of talk about faxes but no mention of modems on this list. I get the impression that modems will be a problem for us. Is that true?

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

Ok I F@%& up. I didn't realize the card is 3.3 volts and my new computer is 5V. Can anyone point me to a PCI to PCI bridge. Any suggestions? Mike

I'm looking for an inexpensive way to connect 20 analog phones to asterisk. I could get a bunch of Linksys or Sipura boxes but was wondering if there is a more cost effective way? I came across the Mediatrix 1104 and even the Mediatrix 1124 but that comes out to be almost $100/port. I might as well buy inexpensive IP phone. Does anyone have any suggestions? Thanks, Waldo

Hello All, I am looking for a IAX2 analog telephone adapter, just want to ask your views on which ones are bad, good and the best. Thanks in advance, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh@imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg

Searching through wiki and google. http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html but there are also other products on the market. --- Wondering if its possible to connect as follows: Extension -> Asterisk -> ZyxelAnalogTelephoneAdapter -> GSM gateway. The best way would be to make the ZyxelAnalog.. to be a channel. But I don't think that is doable.. or ? ---- So i

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the

Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off. I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work: http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html But it is still not working. Does anyone know how to get echo

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

Hi All, It's broken !! (drat) Asterisk if failing to load with the following error (taken from end of /var/log/asterisk/full) after adding bristuff. Can anyone help please? Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so] =>

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

Hello, New subscriber here. I noticed that the FTS index is not used in compound searches. Is this expected? Tested in 2.0.0 and 2.0.8: . search BODY "waldo" * SEARCH . OK Search completed (0.000 secs). . SEARCH CHARSET UTF-8 OR SUBJECT "waldo" FROM "waldo" * SEARCH . OK Search completed (1.768 secs). . SEARCH CHARSET UTF-8 OR SUBJECT "waldo" BODY

Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? Thanks, Waldo

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1

Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28

I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running ztcfg and asterisk fails. [root@asterisk asterisk]# ztcfg Notice: Configuration file is /etc/zaptel.conf line 3: Unable to open master device '/dev/zap/ctl'

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

I'm planning on setting up some remote agents and before doing so, I did some simple PING tests to measure latency. The average latency I got was 250ms. Does anyone have experience in terms of quality of calls when there is such high latency? Can anyone comment? Thanks, Waldo