similar to: Voicemail limit?

Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts "403, Forbidden" . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I

Hi, How can I retrieve those voicemails using my ip phone? and how will i confiugre it on asterisk? Please help I'm very new in asterisk. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Do?a Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with "j" letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of

Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

Hi guys, </br> This is my first post. I'm configuring asterisk together with SER. I'm testing the voicemail feature of asterisk. When I dial the extension it supposed to play first a voice prompt before recording starts. But I can't hear any sounds from Asterisk. Here are my configs: </br> </br>sip.conf </br>[general] </br>port = 5060

Hi All, The situation: When I dial into VoiceMailMain(@context), put in my VM # 1001 and Password 1001, no problem, but at the voicemail main audio prompt (Alison), when I ?press 3 for advanced options? then ?press 5 to leave a message? I put in a mailbox number 1002 within the same [context], but VoiceMailMain looks for the mailbox in the [default] context and will not recognize the mailbox I?m

At the moment I'm out of the office, but when I return I'll be certain to do that. Note that my solution is different from what you are working on with regexten, though I suspect some of the challenges that I've faced and overcome are not. I'm actually using UltraMonkey for load-balancing and failover of the Asterisk boxes, and my dialplan is set up so that it need not be changed

I remember seeing something somewhere that described how I could get SIP header information with Asterisk. It was a command or a variable. Anyone know what it is? Thanks. Doug.

Polycom 601, hands down. - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Warren Sent: Thursday, June 15, 2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk? If you had approx $300 per phone as a budget and needed to buy

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James

Im having a problem where I've just switched from static configs to "realtime" configs stored in mysql It's all working fine (in terms of it reading the configs and loading them as it should), except my problem is that if a user changes there voicemail password via the "Advanced Options (0)" in the Voicemail menu via there SIP phone, the password doesn't get

Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060

If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all goes well until the second time I hit forward (to join the caller with the extension); then, the caller's MoH goes away (making them think they've been hung up on), and the server spits out: asterisk-cw*CLI> <-- SIP read from 10.20.2.16:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP

Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally

Hello all, I am working on setting up a basic Asterisk system (using Digium FXS and FXO cards, Polycom IP phones). I've got a few questions in regards to voicemail and was hoping that someone could give me some ideas... Right now we have three incoming POTS lines. There are times when all three lines are used. When the lines busy out, the incoming call is sent to a voicemail

Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a >reload extension is performed, those dynamically created extensions in the regcontext vanish. Now

Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, "show dialplan xxx" reveals no change. And yes, I have also read and checked bug 7144; if I go down that route and no

Hello everyone, I was just looking over some logs, and it appears that there have been less than 3,000 downloads for my native Asterisk sounds packages (all formats combined). What gives ;)? In my humble opinion, EVERYONE (unless you have your own in a different voice/language) that uses Asterisk should be using these prompts. How about a direct link this time: