similar to: DTMF problems on phonecell

Displaying 20 results from an estimated 400 matches similar to: "DTMF problems on phonecell"

2004 Oct 05
2
Dialing a # in phone number?
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
2004 Dec 22
1
Phonecell + wildcard FXO (DTMF problems)
Hi, I purchashed a Telular Phonecell Fixed Cellular Terminal. I hooked it up to my wildcard fxo card. I can receive calls and these calls are passed on to the Asterisk Calling Card application. My problem is that i can't get DTMF to work properly. If a pin number is 484443543639 i get 4844444333544336639. how can i sort out this problem. Please would like ur urgent assistance.
2004 Dec 10
5
Granstream phones message button
To all: (newbie) I have setup a BT 100 phone and mostly everthing is working pretty good except for the message button. I have place value in the appropiate field in the web configuration but nothing seems to work. When I press the button the speakerphone led goes on but the phone does nothing else (no dialtone, no sip request to *). Does anyone have this buttton working? I would like to
2004 Dec 07
1
How to play messeage when user picks up the phone
Is it possible to play a message, when user pickups a phone. For example: press 1 to use this provider, press 2 to use this ... etc.. Thanks
2005 Sep 15
1
USB ISDN (OT question)
Derek, could you give me some details regarding the solar power supply you're using for your installation? Thanks! J?rg > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Derek Conniffe > Sent: Thursday, September 15, 2005 12:28 PM > To: Asterisk Users Mailing List -
2004 Oct 04
2
Off Topic: Dead GS BudgeTone-100
Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't
2005 Sep 14
2
STUN vs NAT Helper
I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional
2005 Mar 02
3
Multiple lines
Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Feb 04
2
How to Create customized audio file to use with ASTCC??
Hello all, Can anyone help me out with this issue ?? I got ASTCC running, but the audios doesn't match my needs (currency, etc.). is there any way to create my own audios and replace the current one?? Thanks. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Sep 10
4
Fritz, mISDN, Help
A plea to all! Has anyone had any success with two or more avm fritz pci cards with either misdn, chan_misdn, or chan_capi, and any version of linux 2.6.x? I have managed to get misdn to load under 2.6.13 and detect two cards using misdn-capi and chan-capi (using capiinfo and capi info under asterisk) - but the second card/controller doesn't answer or dial calls. But if I try misdn
2005 Sep 13
2
Nat & Sip & Pain
Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",
2005 May 23
1
ZyXEL Prestige 2000W - cant make a call?
Hi All, Today I got a couple of ZyXEL Prestige 2000W WiFi phones. I'm having a problem making SIP calls although I can receive calls just fine. When I try to make a call the phone makes some sound (like "bup bup bup bup bup bup beep beep") and then I just hear hissing background noise (not too loud - like comfort noise). I upgraded to the latest firmware on the phone - Wj.00.10
2005 May 08
4
Cellsocket help needed
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 --------------
2003 Jun 03
3
Fixed Cellular adapters/terminals
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
2004 Dec 10
0
SS7 to E1 & CPC
Has anyone worked out a way to transfer the Calling Party's Category codes to Asterisk through E1 / T1 connections? I know this is normally available on SS7 interconnects but is it also available to asterisk on the ISDN signalling channels? (I kind of doubt that it is......) Thanks, Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201
2005 Feb 10
0
7940 VM DTMF not detecting
Hi all, I have a 7940 running the latest SIP firmware (V7 - thanks Doug Lytle for the tip on the V7 firmware upgrade!). Its almost working perfectly - I can make calls either though my local PSTN or over VOIP but for some reason if I dial my voicemail (which is mapped fine to the VM button on the telephone) it doesn't detect my DTML keypresses so when I press 1 for new messages it just
2005 Mar 01
1
Cisco 7940, Voicemail & DTMF
Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823
2010 Apr 12
2
daemon() function is unimplemented
Hi all, I''m a newbie... so be patient pliz ;-) I''m starting to use Rails on my machine (OS: Win 7) but when I''m trying to connect to http://0.0.0.0:3000 I get nothing. So I launch rails in this way: ruby script/server -d this is the answer: " C:\Ruby19\rubyApps\bookmark_manager>ruby script/server -d => Booting WEBrick => Rails 2.3.5 application
2005 Feb 02
1
Cisco 7940 [SIP], DTMF and Voicemail
Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). The problem I'm having is that when I connect to voicemail the DTMF key presses dont seem to work
2004 Oct 01
0
S100U / wcusb Zaptel driver / Crash / Kernel problem maybe?
Hi Everyone, I've been using Asterisk now for a few months for my small office (which is mostly just me while other guys are always on the road so we rely heavily on telephones) - I'm very excited with Asterisk as it can do everything I've ever wanted to do with a PBX. I'm having a problem with an S100U USB --> Telephone interface. I haven't actually made it work yet