similar to: Pictures of new multiport FXO/FXS from digum

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, I'm relatively new to the subject - so pleace don't punish me for idiotic questions. ;-) Can Asterisk act like a normal Sip phone and e.g. connect to another sip-gateway? Background: There is a new german company at: http://www.sipgate.de (sorry German only page) They offer a a gateway between a real telephone number and

I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 "bridging" using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium

I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error

*This message was transferred with a trial version of CommuniGate(tm) Pro* In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware/software. I need a 416/647 area code number. In looking at FXO cards

Hello, Hacker I install VoiceTronix OpenSwitch 12 port PCI Telephone Card, and setting vpb.conf, extensions.conf following My problem is: When i dial to fxo(channel 9-12), it is ok, but when i continue press exten 102, the channel crach with error messages following exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872 Do i ignore some setting for VoiceTronix OpenSwitch12

Hey Everyone, I am playing around with my * box, and I have a few different phones hanging off it it right now. I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco ATA186 with a Panasonic cordless phone attached to it, I have a Digum IAXy with a dumb analog phone attached to it, and I have a Linksys PAP2-NA with an AT&T 959 analog phone attached to it. I also have several

Hi, I thought I remember seeing somewhere on the Asterisk website a card that had 16 ports fxo or fxs, that was user selectable with straps on the card. Am I going crazy, I can't seem to find it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/86c8ae69/attachment.htm

Hi. I need help with my brand new Voicetronix OpenLine4 board that I installed into Asterisk. After building the Linux device driver and inserting the module, I modified the /usr/src/asterisk/channels/chan_vpb.c file to uncomment the US settings and comment out the Austrailian ones. I made the appropriate entries for routing in vpb.conf and extensions.conf.... All appears to be well, except

Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without

hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part --------------

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up the handset on your desktop phone and the call is automatically moved there, hanging up the cellphone

I have a tdm 3xfxs and 1xfxo, aslo I have a setting with 1 snom 190 and 2 xten line. I can call from the snom to the ptsn line at the fxo port ok. I can call from the ptsn to the xten lite phone. I can call from the xten lite to snom but what I CAN`T do is; Call from xten to ptsn. When I dial from the xten, I can hear the dialed party, but he cannot hear me... Tips? Help? What I'm

Hello! I've been following this list for several weeks now and would like to purchase some hardware for a VOIP/ voice-mail solution. This card appears strikingly similar to a modem. Is it? Is there a product to bring in more than one POTS line short of a full T1? It just seems silly that the technology hasn't advanced any further than to have a single line per card. Any thought

Hi all, i'm trying to make AMD, Answer Machine Detection, to work on my outbound context but i can't get it to work, just on inbound context like whe i use the application Answer before AMD, but i need to make AMD to do the detection on an outbound predictive dialer integration. Follow are the inbound and outbound examples. My current environment is Asterisk 1.4beta3 and a Digum

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

I have installed a 6VPCI card from voicetronix's but i can't get astersik to use it! Now looking at the loaded modules the chan_vpb is not loaded- so I assume that is why it is not working. Now I modified my vpb.conf file and extensions.conf, have I missed something Has anyone a installation guide as I am very new to this!! I have had asterisk working with SIP extensions. by dowloading

Hi there, I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO line. The reason being is that Voicetronix sends out the DTMF too fast even before the line is fully established with the carrier. Usually when dialing an 8 digit number, only 7 digits are actually successfully heard by the carrier. Currently, my dial plan is: exten => _9.,1,Dial(vpb/1-1/${EXTEN:1}) Daniel

Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. -- Jim

Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn