similar to: Voicetronix support

Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE

One of the suggestions I got for dealing with my analog FXO woes was to look into using a Voicetronix card instead of the Digium. They're more expensive but if it worked better it may be reasonable. They indicated they were compatable with the CVS HEAD version of *. Can someone share with me their experiences with Voicetronix OpenLine and/or OpenSwitch cards with *? I'm specifically

hi there, i've been able to successfully run asterisk with the Voicetronix OpenLine4 card, it can accept calls and function normally. The only problem I'm experiencing so far is getting the card to outdial to a third party. What I'm trying to achieve is basically call bridging, where the caller dials in to asterisk, some IVR plays and then attempts to perform a "transfer"

Any one has documented how-tos for making voicetronix openline 4 to work with Asterisk. I have been contacting Australian Digium resellers and Digium cards are not approved in Australia. So I suppose Australian users are interested into putting Voicetronix in use. Any expereience to share will be most appreciated. David Kwok -------------- next part -------------- A non-text attachment was

Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice

Seems strange to me that I should be having this problem but I'm stumped. I'm still playing with a Voicetronix OpenSwitch6 card and Asterisk (development CVS). I using a dialplan that's derived from one on another running installation that's using Digium FXS/FXO interfaces instead of the OpenSwitch card. It seems that as soon as a user on an FXS interface dials 7 digits, a call

Hello all - I'm trying to get Asterisk working with an Openswitch/6 card with all ports set to FXO for PSTN in/out, and 10 SIP phones on the LAN with the Asterisk server. The SIP phones are working properly to dial one another, but I can't seem to get a call to go out to the PSTN through the voicetronix, and I similarly can't get incoming calls to the lines hooked up to the

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I normally don't write emails like this, but the last couple of days have been very frustrating. We built a system for a customer with the following hardware: Asus Vintage (sis chipset) P4 2.8 512M Ram LSI SATA Raid card (2x80g drives) Digium TDM400 we had originally purchased a tdm2400 with echo cancellation but couldn't fit it in the

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

Hello all, I hope this is not off-topic, if it is please let me know. I'm currently playing with an Asterisk at home, in order to get to know it's ins and outs. Very very impressive indeed. I've got it hooked up to my home phone line via a Wildcard clone board (Intel modem with Ambient chipset), and it works like a charm. Zaptel picks up the card as a generic clone, and works with it

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

Hello, As I have problems getting transfert call working with my grandstream SIP Phones, I woul like to know if it is possible to do it with a proper dial plan in exten.conf. I haven't found any information about that in the docs. Regards, Daniel ANDRE -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com

Hello, Sorry for posting again my question about MGCP Phone and Asterisk But I can't use it. I'd like to know weather it is a pb of my confiuration (mgcp.conf), My IP Phone device or asterisk. I include my mgcp.conf file and may send some debug trace. Thank you for any feedback. Best regards, Daniel ANDRE ; ; MGCP Configuration for Asterisk ; [general] ;port = 2427 ;bindaddr =

Hello all, I wonder to have some feedback on using ISDN BRI Cards with Asterisk and the Echo problem. I have tried a simple BRI card with i4l driver and encounter huge echo problem. I have tried to solve it with a Sw chocanceller without success. What I'd like to know is wether some of you have used other BRI Cards (I have seen reference to Eicon cards on this list) and if the echo

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

Hello, I would like to have some comparison between E1 cards from Digium and those from Eicon for a VOIP - ISDN Gateway. How does they compare on the echo cancel point of view? Is the echocancellation code for E400 good enough for production environment? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz -

Hello, I have just received an MGCP Phone for test purpose and I can't place a call from my MGCP Phone. I can call my MGCP phone from a SIP Phone. Here is my mgcp.conf: ; ; MGCP Configuration for Asterisk ; [general] ;port = 2427 ;bindaddr = 0.0.0.0 ;[dlinkgw] ;host = 192.168.0.64 ;context = default ;line => aaln/2 ;line => aaln/1 [192.168.10.10] host = 192.168.10.10 context =

Hello, Now that I have a nearly working configuration for my IP10S with * I wonder if anyone has done call transfert with this Phone. In the IP10S documentation they talk about the 'service key' wich is the key with the white dot on it. With this Key, it should be possible to have a menu with call transfert entries. This menu should (accordingly to the documentation) depend on the

Hello, Dos anybody use these card with asterisk? If so I'd like to have some tips on how to configure them. Regards, Daniel ANDRE -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com