similar to: Multiple lines

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

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

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 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

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

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 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",

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

We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2

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

Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)

Hello! Do somebody knows how to compile meetme2 with 1.0.6. I readed wiki, applied patches, but no luck ;-( Me be someone can give me working meetme2.c ? :-)

>The combination of applications CBMysql and MeetMe2 seem to >address our goals. I have MeetMe2 working. CBMysql is >another story, the code looks simple enough and has been >modified to leverage MeetMe2, but * restarts everytime it >tries to launch CBMysql. I cannot find any examples of how >to launch it from the dial plan, nor have I been able to >get any meaningful

Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having

I've had 50+ people download the web components, and other than reports of compile issues, I have not heard if this collection has worked for anyone. I do plan to keep updating the * applications and the web pages, but I have almost meet all of our internal requirements and wonder if anyone else is finding it usefull. My focus has been and will likely stay on the user interface, since I have

I have Meetme2 (as well as the web ui) installed and am having some difficulty with the admin features. I've set up two extensions pointing to the same conference, one with the admin flag (1234|Maps) and another with (1234|Mmps). My issues: If the admin presses the * key, it goes to an endless loop of "enter the conf no. followed by pound key" If the user presses the * key, the

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

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