similar to: Sending digits from SIP to Asterisk's VoiceMailMain

Hello, I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones. Every once and a while I have problems with either dropped calls between Asterisk and my provider, or invalid RTP audio streams with phones behind NAT. I have had a few Asterisk developers look into my installation and even my provider check my setup but still am having problems. They tell me that I need to

I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: "I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The

Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. Thank you, Steve Maroney

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

List, Okay, here's one that has me stumped, and it might just be something simple. Currently, we are setup so that when someone calls in and tries to reach the operator / front desk, it rings several different phones in sequence. (i.e. it rings the front desk for 15 seconds, then a guy down the hall from it for 15 seconds, then my desk for 15 seconds, and as a last resort, my cordless

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom

I have been getting that message also. I have been using various versions of CVS head since Feb. 2005. -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jeremy Gault Sent: Friday, October 14, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a

Hello I try to set up voicemails for extension. When VoicemailMain gets called, it prompts for mailbox and password. It seems not able to read from the phone. So the authentication always fails. I desparately need help to understand what is wrong. Here is a part of my extensions.conf: exten => _8500, 1, Wait(2) exten => _8500, 2, VoicemailMain(${CALLERIDNUM}) exten => _8500, 3, Hangup

Hi All, i dont what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS

I have noticed a problem with the Cisco 7940/7960 phones where if you put your voice-mail box on hold using soft keys and come back you can no longer navigate. I am curious if anyone else can duplicate this problem. Happens reliably for me with the 7940 phones. I use rfc2833 for DTMF. I would think it was a Cisco bug, but for the fact that this did not happen with older version of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am running into some problems here with PrivacyManager. We used to use it without any issue, but now there seems to be several problems. We are currently running Asterisk 1.2.4. First, it seems that if the user does not press the pound (#) key after entering their number, PrivacyManager will fail. I have the minlength set to 10, and

Hi there. Currently, our network design has two ISP lines and 3 subnets for LAN. Below are some details :- eth0 - isp1 eth1 - isp2 eth2 - subnet1 eth3 - subnet2 eth4 - subnet3 What i wanted to do is to assign incoming port 80 to our local squid server running on the firewall itself and assigned it to eth0(ISP1). I think it shouldnt be a problem as /etc/shorewall/rules provides a sample of the

I have a extension 981 setup for entering VoiceMailMain: exten => 981,1,VoiceMailMain,([mailbox]@usvm) exten => 981,2,HangUp() I want to pass the calling extension to the context (extension and mailbox numbers are the same). This dosen't seem to work. I get this in the console: Asterisk Ready. *CLI> -- Executing VoiceMailMain("SIP/2504-ba66",

Hi everybody, I'm in a middle of a Asterisk learning period. I am at a very good point except I'm not able to use VoiceMailMain. This Is my simple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

Hello guys Is there anyway for me to change the sounds that are presented in VoicemailMain? For instance, instead of it saying "mailbox", I would like it to say something like "please enter your mailbox number now". Is there a way for me to do this? I also noticed that when in some of the menus, even if I select one of the announced options it simply repeats the same menu

Hello, I'm experiencing a weird problem when using the VoiceMailMain application. If I use the application after dialing a Local channel, there's strange beep just after asterisk answers the call and before the first locution. The extensions.conf I'm using is: Ruido extra?o al llamar a la aplicaci?n VoiceMailMain [default] exten => _X.,1,Dial(Local/${EXTEN}@test) [test] exten