similar to: Xten & Digum TDP FXO card: No sound

Hi All, Does anyone know if there has been any developments in asterisk to use LineJacks to dial out (connect to the PSTN)? The card works perfectly with virtually anything else but asterisk. Maybe the CVS versions have some work on it? Cheers, -Z -- __________________________________________________________ Sign-up for your own FREE Personalized E-mail at Mail.com

Signed-off-by: Karol Herbst <karolherbst at gmail.com> --- drm/nouveau/nvkm/subdev/bios/vpstate.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/drm/nouveau/nvkm/subdev/bios/vpstate.c b/drm/nouveau/nvkm/subdev/bios/vpstate.c index 20b6fc82..71524548 100644 --- a/drm/nouveau/nvkm/subdev/bios/vpstate.c +++ b/drm/nouveau/nvkm/subdev/bios/vpstate.c @@ -58,8

Does the voicetronix card work with Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of woody+asterisk@solutionsfirst.com.au Sent: Monday, February 02, 2004 11:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum > -----Original Message-----

On 17/11/17 02:04, Karol Herbst wrote: Please add here something like this: If the entry size is too small, default to invalid values for both boost_id and tdp_id, so as to default to the base clock in both cases. > Signed-off-by: Karol Herbst <karolherbst at gmail.com> With the better commit message, this patch is: Signed-off-by: Martin Peres <martin.peres at free.fr> > ---

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

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium. Usign exclusively digium hardware. 3 TDM400P cards. 1 4xFXO 1 4xFXS 1 1xFX0 & 3xFXS When * is attending FXO calls, bridged to FXS calls, natively ofcourse, at a random time, the call hangus up. Also, for example, if a call is done, and an other extension hangup, there are some probability that the other extension

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

I've been looking for a good iax or sip <==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office:

Hi All, I have the following scenario, it may already have been answered elsewhere, but I cant find the solution. I already have a PBX and would like to start implementing asterisk. I have ordered a 4 port card from the asterisk store (2 port FXS and 2 port FXO) and am waiting for it to arrive. I do not want to plug my incoming lines into my FXO ports yet as not all the desks have IP phones

Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on

Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D

Hello Asterisk List, My company is running a bunch of Asterisk servers behind a Kamailio (openSER) SIP proxy gateway. Calls come in from our PTSN to VOIP service to Kamailio, which then randomly chooses an Asterisk server to handle the call. All Asterisk servers are 1.6.0.9, but the issue I'm about to describe exists in 1.6.1.5-rc1 as well. Ultimately what I want to do is cap each

I'm using a Te201p card, with unichan, I want to know if my channels are "ready" or in alarm... but uc show channel o uc show channels, doesn't show me anything... Any Ideas? thanks.

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

Hi all. I`ve installed a X100P on my box and is working well with incoming and outgoing calls as a trunk with one PTSN line. I want to connect the X100P to my Panasonic 616 EASA-PHONE as an internal extension to permit to users to make calls to SIP devices from analog phones, the problem is when I dial the ext number where the X100P is connected I get busy tone. What config I need to change to

This indicate that the connection with the local provider PTSN it is ok? : -- Attempting call on Zap/10 for s@inbound:1 (Retry 2) -- Making new call for cr 32781 > Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 2 (reference 13/0xD) (Originator) > Message type: SETUP (5) > Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) >

Hi, I am new in asterisk. Is it possible to use it with common dialup modem to connect ptsn to the server? Thanks Regards, Soragan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040123/8e77b5cf/attachment.htm