similar to: Cellsocket update

I recently purchased a Cellsocket, which is a cradle that holds some older Nokia GSM/PCS phones and converts them to an FXS interface. My test phone is a Nokia 5190 on the T-Mobile GSM network. After going through the ordeal of unlocking the phone (T-Mobile provided unlock codes that didn't work), I finally got it up and running. The good: - Disconnect supervision works - Outstanding sound

I am posting this in case someone need help.. ========================================================= YOU THA MAN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! No sure how I will repay you, but anything you need, just let me know!!!!!!!!!!!!!!!!!!!!!!!!!!!!! Thank you, thank you, thank you!!!!!!!! -- Executing GotoIf("SIP/2007-12c7", "0?4") in new stack

I need help from someone who has a working cellsocket, I have received couple email of people who wanted to help, but they just think they know how it supposed to work, but they don't have a working units, and they confused more...I need someone with a working solution to get my cellsocket going.. Thanks!!! Write offlits @ mawise (AT) mail.com -------------- next part --------------

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson

Hi all, I've just connected a cellsocket to my * server so that I can keep call costs low for outgoing calls to mobile phones. I'm using an old fashioned 5110 cellsocket (because I have a few of these phones and I think that they are very reliable). For those that don't know the cellsockets they convert a mobile phone into a POTS line. The cellsocket has a funny dialtone but that

FYI. http://www.telular.com/products/index.asp These look like the right solution for any one wanting a cellular FXO device for * that interface with the digital cellular networks (GSM ,CDMA etc) similar to the www.cellsocket.com or the mystery FCT's http://www.ericsson.com/products/products_az.shtml#F or those old Motorola Bag phone adapters -------------- next part -------------- An HTML

Hi, I'm looking for a reliable, reasonably-priced, single-channel interface between * and US GSM. The VOIP GSM Gateways listed at http://www.voip-info.org/wiki-VOIP+GSM+Gateways (VoiceBlue, QUTEX) are multichannel systems, very expensive ($2500 or more). Next step down, there are various Fixed Cellular Terminal (FCT) or Fixed Wireless Terminal (FWT) devices. These typically have an FXS

Does anyone have the iRiver iFP-790 player (or maybe any other one of their iFP-7xx series)? I bought this player very recently, and I have been having problems with Ogg Vorbis playback. It skips so much that you only hear music 20% of the time. Is my unit defective or does the player really have problems with ogg vorbis? I think mine even chokes on wma's encoded at standard settings. Any

Hello, I was wondering if anyone had any ideas to a problem I am having. I have asterisk installed, and running. All extension phones are analog on FXS ports. The phones call each other and even transmit caller ID information but they DO NOT RING. Here is my setup: T400P Quad span T1 card currently connected to a CAC I w/ 12 channels of FXO for incoming lines and 12 channels of FXS for

Many of you are familiar with how lousy Festival sounds. AT&T has a product, NaturalVoices, that sounds much better. There are male & female voice fonts for US/UK/Indian English, French, Spanish, and German. I am considering offering a linux-based text-to-speech engine based on the NaturalVoices runtime. An asterisk module would also be provided, making it easy to add natural sounding

---------- Forwarded message ---------- Date: Tue, 4 Jan 2005 17:12:32 -0800 From: Bill D'Anjou <danjou@rocketinternet.net> To: steve@daviesfam.org Subject: FW: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server Can you post this to the group for me? I've joined & I am receiving e-mails from the group but when I try to post I get a bounce saying "waiting for

Layne Fowler writes: > On Mon, Dec 9, 2019 at 12:58 AM Marco Walther > <<URL:mailto:marco at sonic.net>marco at sonic.net> wrote: > > Lite information I can help with, let me know. > > We had that discussion about the SMART1500LCDT  & USB a long time ago:-( I Ditto. > bought mine in 2013 (still going strong on the second set of batteries) >

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

Hi, I have not been successful in working out how to dial a # within a phone number. EG: exten => _12345,1,Dial(Zap/1/0868563823#,5,t) or exten => _08XXXXXXXX,1,Dial(Zap/1/${EXTEN}#) I'm trying to append a # character so that I can use a cellsocket (mobile phone to pots adapter) connected to an x100p. I think that asterisk is simply ignoring the # character. The docs on

I've just made an interesting observation that I'd like to share with you all: the popular Sipura SPA-2100 just doesn't seem to be as great as I'd hoped. I've been trying to get inbound AND outbound faxing working via Asterisk and at least one of my termination services: Voicepulse or Sixtel. In general, inbound has been working flawlessly but outbound has been pretty

Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.

Hello everyone, I am facing a strange problem on my asterisk box (using isdn lines with pri card installed on it). Normal incoming/outgoing calls are working perfectly fine. When a user dial a wrong/out-of-service number they don't hear back any such message like "The number is wrong or user is switched off" in some cases, and it's just a silence for the user. Now while

Greetings, Using Xen 3.1 and Fedora 7, is it possible to export the serial interface such that I can access external devices on COM1 or COM2? I am uncertain if PCI pass-thru is needed or if there''s nothing to configure for this to work in a fully virtualized Linux guest. Any help would be greatly appreciated. Regards, Tony Johnson _______________________________________________

-----Original Message----- From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of John R Pierce Sent: Thursday, January 19, 2017 1:01 PM To: centos at centos.org Subject: Re: [CentOS] [OT] VOIP On 1/19/2017 4:41 AM, TE Dukes wrote: > I lost > the ability to use the DSL as a FAX line. Analog traditional FAX may not work very well over VOIP. Just sayin'. > So, I bought an

Hi all, I'm playing with Asterisk and I've already configured all needed .conf files. It works quite well, but now I need your help to tune the system: when I place a call from a softphone to the PSTN, I can't hear directly Telco's tones and I can't use its services, e.g. a mobile's answering machine. I don't know if I have to modify the dialplan or if it depends on my