similar to: HasNewVoicemail not being called if user hang up after leaving VM ??

Hi I have been trying for a while to find a way to get an SMS send when I receive a voicemail into my asterisk system. I don't want to send an SMS if the caller doesn't leave a message. I have voicemail.conf set up to email and delete. 302 => 302,Website Sales,sip@example.com,,attach=yes|delete=yes However I can't seem to find a way to test is a message is left. I have tried

Hi all, Maybe I'm being thick here, but I've had a look through the mailing list and the Wiki, and I can't seem to see details of anybody else with this problem.... With the following line: exten => s,1,HasNewVoicemail(201) I am getting the following error: -- Executing HasNewVoicemail("SIP/201-2f1e", "201") in new stack Sep 2 12:41:09

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

Hi, I am trying to implement a dialplan in which the user is notified of a missed call, if no voicemail is left. Basically what I would like to achieve is something like ... exten => _0207XXXXXXX,1,DIAL(SIP/${EXTEN},15) exten => _0207XXXXXXX,2,HasNewVoicemail(${EXTEN:4}@default:INBOX|msgcount) exten => _0207XXXXXXX,3,Voicemail(u${EXTEN:4}) exten =>

Hi, for some weeks now I have been unable to make calls via my voicepulse connect IAX account? When I attempt the console looks like this:- rt*CLI> -- Executing Dial("SIP/2008-cf55", "IAX2/NBhXXXXXX:XXXXXXN82@gwiaxt01.voicepulse.com/12124565900") in new stack -- Called NBhXXXXX:XXXXN82@gwiaxt01.voicepulse.com/12124565900 -- Call accepted by 66.234.228.160

Hi all, I have a demo machine I'm running up on Lenny - it has the packaged Asterisk version installed (1.4.21.2+stuff). I'm trying to add an extension to leave a voicemail message, just with Voicemail(1234), which I've done before (on 1.2 at least), but it's saying "no application 'Voicemail' ". "module show like voi" shows

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"

I received this before, and it is because you are using the wrong context in the iax.conf. For example: The context must match the username in the register statement. Iax.conf... register => username:secret@iax.host.net [username] type=friend context=iax-in user=username secret=secret auth=plaintext host=iax.hust.net ----------------------------------------------------------------

Hi, Is there some script which can be called from a * extension to playback the recent incoming callers on a particular PSTN line? In the UK 1471 is a BT number which plays back the most recent callers number, it also gives you the option to call this number back (now charging you for this service too!). Is there anything similar in asterisk-land? thanks Mike

Hi, I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. I've setup a fwd account and added IAX capability to it via the website. I got the email saying it had been done with some welcome text and sample phone numbers to try, such as 10001 for the answer phone. I followed the setup example on the fwd site for configuring * to work with fwd's IAX.

Hi, I have a Cisco 7960G phone for which I know the pinout of the headset socket. I have a couple of standard telephone headsets which I do not know the pinout of. I'd like to connect the two. If I have the pinout of a normal/standard headset I can rewire the ones I have to match the cisco. Thanks Mike

OK, I have just discovered what may be a conceptual flaw in realtime extensions. DB schema as follows: uniqueid, context, exten_id, priority, application, application_data Primary key is uniqueid with an compound index of context and exten_id data looks something like this: xxxxx, sip, 5555551212, 1, SetVar,ext=${EXT} xxxxx, sip, 5555551212, 1, Goto, officevm|2|1 The database is MySQL

Hi, I've recently got * working (thanks Clive and list!) at home. We have 2 PSTN lines connected via X100P cards. I've got 3 x SIP phones (2 are Budgetone, the other is a Tecom SIP). One of the lines is our standard home line, the other a business line. Presently I've got * set so you dial 91<tel no> and 92 <tel no> to select which line to dial out on. I should probably

Hi, I recently upgraded the firmware on my Blackberry 8700 to 4.2, this seems to give it the ability to play wav files. I wondered if anybody out there had managed to get their BB to play the wav files as attached to the Asterisk voicemail emails? Mine seems to ignore the attachment. I am using BES 4.1 for sending these emails out via Exchange 2003 if that makes a difference. thanks Mike

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

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Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Hi, I've been setting up * recently and slowly getting to grips with it, however I'm getting rather confused with all the different configs for IAX calls, FWD calls iaxtel etc etc. What I think I need it a basic understanding or even a comparison of these different voip systems (if thats what they are?) I'd like to be able to make calls to other voip users, both in the UK and abroad