similar to: For Sale - (10) Dialogic D/240SC-T1 REV2

I'm trying to connect an Asterisk T1 port to a Dialogic card. The Dialogic side is an external VMS. I setup for ISDN-PRI between systems and have green lights on both card/ports. Zttool shows connection is good also. However, when I try attempt terminate or originate a call to either system, nothing appears of CLI. How can I monitor the "D" channel? zaptel.conf: # Span 8:

Anyone in need of some of these? Garrett Smith Sales Executive garrett.smith@b2llc.com B2 Technologies 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix,

Garrett Smith Sales Executive garrett.smith@b2llc.com B2 Technologies 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix, Carrier Access, Digium, Zultys,

On Fri, 2019-08-16 at 21:29 +0000, Patchwork wrote: > == Series Details == > > Series: drm/connector: Allow max possible encoders to attach to a > connector (rev2) > URL : https://patchwork.freedesktop.org/series/62743/ > State : warning > > == Summary == > > $ dim sparse origin/drm-tip > Sparse version: v0.6.0 > Commit: drm/connector: Allow max possible

Hi, I am having problems configuring my Centos 4 server as an ADS domain member of our 2003 AD. I've followed the instructions on samba.org and did quite a bit of Google'ing and haven't found an answer to the problems. Basically I used the configuration illustrated in this section of the howto, and of course a number of other suggestions I've found along the way:

Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html

My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of service. They were deployed for about 6 months. These include the AC power adapter and station license. We also have some other related equipment. If someone is reading this and is interested, shoot me an email candrews@b2llc.com Thanks Cory Andrews ***************** b2 Technologies

Don't want the opening line of my post to be misconstrued. "Anyone familiar with the BrookTrout TR1034 Fax Board, supports T.38 and PSTN fax in an Asterisk environment?" That's a question, not a statement. Curious if anyone has played with one of these as an extension of an Asterisk server for T.38 faxing, did not mean to imply it could, should or would play well with

Just got a call from a company in Warren, MI . They recently had an Asterisk system put in by a vendor, and are having issues which need analysis and correction. They have a tremendous sense of urgency. They have about (40) users, and need DID's assigned to extensions and are having some echo issues at the site. If anyone is in the Warren, MI area, and is interested in some cavalry work,

Hello Everyone, I have two never-used, still in the static-bag T400P Cards that I bought a long while back that I'd like to get rid of. Before I ever got a chance to use them, I bought Sangoma A400 Cards instead, and now I definately don't need them any longer. They have the Dallas DS21Q352 Chip on them (A4 Revision). I'll be happy to test them before I ship them if you like.

I have a lot of folks asking me about an auto-negotiating WLAN phone supposedly being brought to market by Hop-On, which is touted to carry an MSRP of $40 Press photos (stock art) of the device shows it looks almost identical to devices from Zyxel and UTStarCom. I am trying to explain to folks there is no way in hell you are going to be able to buy these phones, hardware only, for $40

I'm hosting some folks today from Sangoma for a sales training. If anyone on the list has any specific questions regarding current or upcoming Sangoma products, please shoot me an email in the next hour or so and I'll do my best to get you the answer. Thanks -- Cory J Andrews Partner / Purchasing +++++++++++++++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo,

Have a customer running a 3rd party PBX implementation based on Asterisk, not utilizing SIP inbound and outbound calls I believe are coming through a Digium TDM2402B. They are utilizing Polycom phones. They are experiencing frequent static on the line, and overall insufficient volume on conversations. They are in a bit of a rural area, I was curious if anyone thinks it could be an issue with

Lumenvox recently put out a press release regarding integration of their technology with Trixbox http://www.lumenvox.com/news/lumenvoxNews/2006/092506.aspx If they have a TTS engine as part of this integration, you should be able to create an IVR with Trixbox to grab DTMF input from a user, query a local or remotely accessible database, and run the query result through a TTS engine and output

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Thanks Cory Andrews ++++++++++++++++++ VoIPSupply.com PBXSelect.com ++++++++++++++++++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax -

---------- Original Message ---------------------------------- From: Louis-David Mitterrand <vindex@apartia.org> Reply-To: asterisk-users@lists.digium.com Date: Mon, 22 Sep 2003 22:28:40 +0200 >On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote: >> My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of >> service. They were deployed for about 6

I am currently developing an application in which two users exchange goods and money in an escrow like environment. Buyer has some money and wants an item that Seller has. Buyer does not want to pay Seller until Seller ships the item. I am unsure of how to best handle the payment part of my application. Buyer''s money eventually has to end up at Seller and my company would like to

Spanish/Espa?ol: Hola a Tod at s, Estoy en la implementaci?n de una soluci?n a medida en la cual requerimos conectarnos v?a SIP a un IMG de DIalogic y salir hacia otras centrales (MSC) v?a SS7, la conexi?n entre Asterisk y el IMG es SIP, pero como requerimiento necesito enviar un NOA (Nature of Address) con valor 8 por requerimientos normativos, hasta ahora no he podido encontrar como puedo

I am looking at some lower cost phone to use with Asterisk. What is the ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find any information on it. Adi