similar to: Granstream phones message button

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

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

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

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 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

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:

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 have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James -------------- next part -------------- An HTML attachment was scrubbed...

Hi all, I'm using VOIPJET to make international calls with an IAX2 connection between my local asterisk server and their server(s). At times I seem to have a problem if 5 or more international calls are made at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the asterisk server uses this DSL line). Today I switched the codec from ulaw to ilbc in an attempt to lower

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

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

To all: I am having an echo problem with X-Lite and iaxComm. I am using the monitor speakers and a desktop microphone. My problem is that the sound from the speakers is repeated by the microphone causing an echo that is annoying. Is this correctable? I have searched throught this archives and played with various settings but have not been able to fix this. I have purchased a couple if

Hi Derek, I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ?? Thanks. Daniel. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Conniffe Sent: lundi 7 f?vrier 2005 11:59 To: Asterisk

I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11. It's relatively stable, and the last thing I want

hi all, I known taht this ML is about *, but lot of you are using BT telepnones. I'm using FW 1.0.6.7 for all phone. This firmware support VLAN tagging for QoS on the Layer2. I use it to separe the PHONE network form the PC network, which are in PC connector in the BT. And I also use the TFTP server for provisioning (software, configuration). The problem is, if you set the VLAN for QOS.

I recently posted a question RE the Sipura 941 and using different ring tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's didn't seem to do the trick?? Has anyone one had any luck on this topic? Also haven't been able to find any info on an auto-answer for the GXP 2000, again, I have succeeded in doing so with the