similar to: List extension in directory without mailbox?

Anybody ever put a Sangoma and a Digium card in the same server? Specifically a four port card from each company? -bill wlloyd@slap.net

Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid this ? Jonas -------------- next part --------------

I'm in search of useful ACD type statistics from the queues. Ie talk time, ratio's, dropped calls etc. The flat file queue_log is nice, but more useful would be the data in Postgres or Mysql. Unfortunately the queue module does not yet support ODBC DB logging (yet). In the meantime this quick and dirty hack gets the job done. Replace the flat file with a unix named pipe.

I too have been having SIP problems the last couple of days, I get the same message as Louis-David but in this setup only: PSTN >-ISDN-> AS5300 >-SIP-> * I can make outbound calls no problem but inbound calls seem to stall, according to 'sip debug' it just says 'Ignoring this request' but I cant establish why .... > -----Original Message----- > From: William

I've got a Zhone 24 port FXS to configure. The configuration is beyond stupid. The people that designed this unit should be chased down and fired. I'm going around in circles frigging with all the options. Does anyone have a config file for this unit that I can use as a starting point? -bill wlloyd@slap.net

I'm trying to get a patch tested for inclusion in CVS. Anyone that is running E&M on a T1 and had to fool around with emdigitwait could you please try this. This patch removes the need for the emdigitwait parameter and speeds up dialing. This situation is mostly interfacing a legacy PBX/Key system with Asterisk. I've been running a couple of systems with this patch for

Scenario is as follows. Caller comes in over ZAP channel connects to handset on another ZAP channel. Call is bridged. I'd like the callee to be able to hangup on the caller and then be presented with a agi application. Basically the agent that answered the call has to enter a few responses to questions asterisk asks. On some ACD phone systems this is called a "wrap code".

I've been trying to work with the dynamic feature support.. IE adding codes like *2 to features.conf that can trigger a dialplan application to run. I've been unable to get "goto" to work properly. "AGI" also seems to not function correctly if called as a feature. Anyone else playing around with this feature might have some insight? -bill wlloyd@slap.net

The cell companies are "doing it" like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give "4-10 rings" of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but

The cell companies are "doing it" like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give "4-10 rings" of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

Hi, Is there a way in Asterisk 11 to use a single voicemailbox for multiple extensions while still hearing each extension's individual greeting? Use case: someone has 2 numbers and wants all voicemail messages for both numbers to end up in one mailbox. So when dialing 1234 and NOANSWER you would hear "the person at extension 1234 is unavailable" and the message would be stored

I am sure this has been addressed somewhere else, but I haven't found it. Is there a way to make multiple extensions have their MWI light flash, all for the same common voicemailbox? And to make it even trickier, what if it's a mix of SIP and ZAP channels? I'd like to be at my server-room desk and see MWI from my office mailbox.. -------------- next part -------------- An

Hi I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. The problem happens: - With both software and hardware phones. - With calls going out through the ZAP channel and to internal SIP extensions. - After I have transferred an

hi community, I'm new to this list & asterisk in general, so let me first say thx to everybody involved in providing such great tools & ressources!! I'm currently trying to implement a simple voicebox-system. for demonstration purposes, I've successfully connected my cellphone via bluetooth using the current chan_cellphone-patch on the current SVN-version of asterisk.

Hello, We have a situation where a call comes in, users are notified via an external process (curl request to web service), and we can't answer the call until a callee can call in and pickup the call. How can we implement this functionality? We tried using : [caller-inbound-leg] ; code to send the CALL_UUID information to users. exten =>

Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. from my dial plan: [incoming] exten => s,1,Dial(SIP/somebody1|60|tTrR) [internal] include => outbound-local include => parkedcalls

Hello list, I try to pick up a ringing extension but nothing works. To be clear, I'm trying to pick up extension 10. [Jun 14 17:37:34] -- Executing [**10 at from-TESTCORP:4] Pickup("SIP/testcorp3-00000041", "10 at 123456") in new stack [Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No target channel found for 10. [Jun 14 17:37:34] --

Anyone know how (or if it's possible) to get a logo on the screen of a Polycom 501? I've been looking around for hints on how to do it but so far nothing would indicate Polycom supports doing it. -bill

How is a callgroup used in the dial plan?? I can't see an example in the extensions.conf.. > See configs/sip.conf.samples search for pickupgroup > > RTFSC (Sample Configs) > > > Jeremy McNamara > > > > > WipeOut . wrote: > > >Is there a way to pickup a call whn using a SIP phone? > > > > > > > >> From