similar to: Transfer of queue call

Hi All, I was wondering, in my effort to show how Asterisk can replace Call Manager, if there is support for call transfers/forwarding from the users Cisco 7940 SIP phone to either another SIP client or through the AS5300 on to the PSTN. I do see some stuff in the docs but seems to be specific to a local PRI board in the PC of which I don't have. Any experiences/comments most appreciated.

The process for transfering a call with the Bugetone is as follows.. 1. Press "transfer", you will get a dial tone.. 2. Dial in the extension to wish to transfer to.. 3. Press the "Redial".. (on the newer phones this is the "send" button) You don't need the "t" option on your dial string to do transfers with a budgetone.. Later.. > With the right

I am just studying Asterisk now and have a question. Is it possible to force anyone who enters a queue into voice mail after they have been in the queue for 30 seconds? /************************************************************** Ken Alker ken@impulse.net ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo,

On Tue, 2002-10-29 at 05:01, Hielke Christian Braun wrote: > i am trying to use dovecot with pam and radius. My users have names > in the format joe at somedomain.com. When i have pam configured to use > the normal passwd/shadow files it works fine. With radius it does not. > I see at the radius server that the domain part of my usernames > is always replaced with the same domain

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and

Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to

I have a caller screen queue setup. Basically a caller calls in, goes through a IVR, and before that caller is put into the queue, they get a sub ran on them first asking for them to say there name. That gets saved and they are entered into the queue using Queue(mainqueue,,,,300). In the queues.conf i have a list of members these are local/extension at external-default, there are two

Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working?

Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

Hi Folks, Since there isn't a grandstream forum AFAIK I guess someone here may be able to shed some light on this. Apologies if this is viewed as offtopic.. I think I may have killed the firmware on my Grandstream Budgetone 101. I found a source for the 1.0.5.30 firmware and made the files available over tftp. The phone downloaded the files but now doesn't boot and hangs with a blank

Hello all, Does anyone know how to successfully transfer a call using the GrandStream Budgetone 100 phones? I've read multiple posts talking about hitting flash, the dialing, then flash again, etc. Some posts talk about using the transfer button, then dial, then flash. Anyways, it seems that I am able to put the caller on hold (whereas they hear hold music) by pressing flash or

I have a couple of Budgetones that I am playing with trying to get them to work with * from a remote network over the Internet (yes NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded (UDP) to my public address (through a Cisco PIX). Internally, I can setup my budgetone, it registers and works great. I then have a Linksys router connected to another Internet

Hi. I was wondering if there's a way to do supervised transfers on the budgetone 102 . Blind works ok, but can't do supervised. I thought that with the flash button that could be possible, but seems I'm wrong .... matteo.

> No, the 1.2beta1 preprocessor works rather well. I have a different opinion, but glad you like it anyway :-) > In fact, I very much welcomed the change that the preprocessor in > 1.2beta1 now starts out with the assumption of there being no speech, > and only activating and adapting AGC when where there is new activity > detected above that. As a result, with 1.2beta1, there is

Hello. Can You provide example, how can i run specific extension after incoming call going into queue and answered (but not hangup). (i want to use System(echo .....) after member of specific queue answered a call); Thank You. -- Vasiliy G Tolstov <v.tolstov at selfip.ru> Selfip.Ru

Can Asterisk light the message waiting light on a Grandstream BudgeTone phone? If so please reply with any related configlets :-) Thanks!

I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone

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