similar to: Off Topic: Dead GS BudgeTone-100

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

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

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",

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 -

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

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

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

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

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:

hi all. was having problems with my phonecell connected to wildcard fxo port. i get problems with detecting DTMF. i have tried relaxDTMF but to no avail. i have asked this before but would like possible causes. is it to do with echo? problems with the GSM network? haven't updated my asterisk for a long time. could this be a problem that has been sorted out. please would appreciate ur input

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:

> -----Original Message----- > From: Gilad Ben-Yossef [mailto:gilad@codefidence.com] > I'm prbably stupid, but wont this do what you want? > > > exten => 1,1,Goto(bye,s,1) No, because I wanted to match on "D", not "1". Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't

Redhat 9 is running 100% cpu usage. I had a couple boxes doing this. upgraded to Fedora and its ok.

BR C.

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

Hello, I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21) and i experienced that the memory consumption of the asterisk-process started by the init.d-script raises continously. Now, after 3 hours of operation (on our testing-system we have 30 concurrent connections to another asterisk box using IAX2 and GSM codec) there is already 66MB allocated. I think this could be ok, but

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

> -----Original Message----- > From: David Brodbeck [mailto:DavidB@mail.interclean.com] > Okay, the problem appears to be that I'm tone deaf. ;) > > I finally thought to turn on debugging on the channel. The > PBX is sending > "D", not "*". The programmer of the previous voice mail system (whose > configuration I was cribbing from) seems to have

At time to time somebody is trying "their luck" and send me most likely a junk fax on my voice line. During normal working hours is not a problem I just pickup the line and hangup the call but after-hours my voice mailbox is intercepting the call and recording those "beeps" (waisting my CPU cycles). Is there a way to block call / issue hangup command if the incoming call is a

For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100