similar to: Blank Voicemails/Vonage Problem

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

Hi! I am currently working on setting up an Asterisk system, and I was wondering if anyone has worked on an alternate mp3 player to mpg123. We have a library of MP3 files that we would like people to be able to select and play over the phone -- and this will require pause & resume, as well as fast forward / reverse (jump forward / jump back). It doesn't seem like mpg123 can do this. Is

The options are not seperated by commas. exten => s,1,Dial(SIP/50,23,r,d) should be exten => s,1,Dial(SIP/50,23,rd) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Klimek Sent: Thursday, June 22, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

We've been running an Asterisk-based phone system here in our office for a year and a half, and it's pretty much been running smoothly. One employee who works out of the office has a problem that she can't make outgoing calls on a temporary basis every so often (a few times a day). No one else has this problem, her settings are fine, and she regains the capability spontaneously with no

Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx >> I> AGI Rx << #!/usr/bin/php5 -q AGI Tx >> 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping

If I understand this correctly, this is a user outside your firewall dialing in to your office over the Internet. Always, inbound calls work, but sometimes, outbound calls do not work. So if you have replaced the hardware totally, and you still have the same problem, it could be a routing issue with an upstream ISP. The way to test for this is to do a traceroute from her LAN to your office. Then,

Hi, For a few weeks now, our asterisk server has been experiencing something very odd. From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707

X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: yusuf at ecntelecoms.com X-Spam-Status: No My third try, humph! Yusuf wrote: > Hi, > > I am trying to understand the RTCP stats in Asterisk. > > 1. I am using the 'h' exten to store the RTCP records in

In article <1251.165.146.69.140.1144596935.squirrel@www.ecntelecoms.com>, yusuf@ecntelecoms.com says... > Hi, > > I have had the exact same problem last week. I have not yet solved it. > So instead I am using ooh323, but would prefer to use oh323. Can anyone > help? I'm glad that I'm not the only one :)) Hopefully we'll find solution to this problem. --

Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I

Hi, Pretty sure I'm missing something simple, but I've seen references to this feature but not found documentation for it: I have a queue set up so that many people are contacted (ringall) when a call comes in. I would like the answering party to confirm that he is a human being rather than cellphone voicemaill by pressing a digit. This is somewhat similar to the 2nd macro example

I've noticed that some products, like Fonality's HUD, allow live monitoring of a VoIP call (not just Zap Barge). The Asterisk {client | manager} command set only seems to allow recording to a file without the use of a meetme room. Does anyone have a good solution for this? What I'd like to implement, ideally, is that once an incoming call is transferred to a particular operator,

Hi, I've been trying to follow the documentation that I am finding, but am running into trouble getting things set up correctly for postfix + virtual domains (using ldap) with dovecot LDA. I can get it to work without LDA, but I'm running into permissions problems when I try to run with LDA. I am wondering if anyone has any good examples of configuring this. I basically have a

I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten => s,1,Dial(SIP/50,23,r) exten => s,2,VoiceMail(u50@default) exten => s,3,Playback(vm-goodbye) exten => s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are

Anyone out there using ActiveLDAP have an idea how I can accomplish creating an object one level below a known base where we have a variable item in the middle? That first sentence doesn''t even make sense to me. Here''s what I want to do: I have a user class that I use for managing users. Each user gets a ou called addressbook (which in turn will contain sub-entries, but

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Jerry

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

Dear List, I need to print out each of 'k' levels of a factor 'n' times each, where 'n' is the number of elements belonging to each factor. I know that this can normally be done using the gl() command, but in my case, each level 'k' has an unequal number of elements. Example with code is as below: vc<-read.table("P:\\Transit\\CORRECT

I am using Trixbox 1.0(Asterisk 1.2.7.1)at a customer site. They whishes to change the default Voicemail behavior. Standard behavior No answer/Busy -> send to Voicemail Requested behavior No answer/Busy -> message that if you press 9 you will instead be cent to reception -> send to Voicemail or Reception if 9 pressed. I want this to always happen when Voicemail is invoked. How

I setup a vonage account last year, and cancelled it last night when I put my asterisk box together and signed up for a Broadvoice account to use with it. Now I would like to use my Linksys router as an MTA, but realize it is still programmed with all of vonage's proprietary information and I do not know how to clear it. I understand that just pushing the reset button will not do it.