similar to: Paging with Pickup

Hello Maybe a silly question, but after some searching couldn't find answer. Is there a number I can dial to pickup and listen to my voicemail messages on my SIP phone? I am used to eg dialling *17 to pickup my voicemail messages on Avaya system? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to duplicate a behavior we had with our old avaya system, and I've come across Auto Answer (Ring Answer). However, its not quite the same yet. Right now, when I dial **5053, it will add the SIP header for Ring Answer and it will call 5053. The phone auto pickups just fine. However, we need that call to be muted. If you were to call into a meeting, we wouldn't want them to

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. the customer is currently using a paging system (small receivers which display a callback number and a base station (transmitter) with several antennas at the site) the problem is, that the currently operative base station uses 4 ISDN BRI interfaces. But the protocol is old germany 1TR6

Hi, > Does anyone of you has a working configuration with SNOM phones that are > able to pickup a call from a flasing LED? Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and therefore I don't think any config changes will fix it. We've been told to roll back to our previous 1.4.13 installation. It also seems to manifest itself in "ghost

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Paul Madley wrote > >Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 >release, and therefore I don't think any config changes will fix it. >We've been told to roll back to our previous 1.4.13 installation. It >also seems to manifest itself in "ghost ringing" as I've called it; >place a call to a SIP extension, then put down the

I was wondering if someone could help me with this, I've searched high and low to find more info but with no success. When I want to transfer a call from another ringing sip phone to my sip handset I dial *8. This works but the caller ID shows up as *8 on my handset. What I want to do is be able to have the original caller id come up on my SIP phone rather than *8. There was a page on

I spoke the other day about my preliminary tests with office-wide paging with Cisco phones using the new SIP 6.1 image which supports auto-answer. I've got a small and crude recipe for those of you who want to experiment and hopefully create some better and more complete examples than the one I've thrown together below. Create a new line on each of the Cisco phones, and put the

Reason. I have a fax/ans phone with handset, that lets you monitor the caller, so if you wish, you can pickup the call. The asterisk is undergoing testing, it will then be online tested at the house so I can get more familiar in setting components up, e.g. sip phones, voicemail, transfers etc. But, I really need the monitoring of the Voicemall being left, with the ability to pick up the

Has anyone had any luck getting the Avaya IP SoftPhone to work in wine? I get these errors on Ubuntu 8 Code: preloader: Warning: failed to reserve range 00000000-60000000 wine: creating configuration directory '/home/microchp/.wine'... preloader: Warning: failed to reserve range 00000000-60000000 preloader: Warning: failed to reserve range 00000000-60000000 err:dosmem:setup_dos_mem

Hi, I'm wondering if it's possible to get Caller ID information from a Call Pickup. specifically on a Cisco 79xx SIP handset. I've setup a speed dial line on my 7940 to dial *8 so I can quickly pickup a call, but because the 7940 initiates the call the information on the screen is "To: *8" I gather because of the call setup it's probably not possible to

> -----Original Message----- > From: Jerry Geis [mailto:geisj@pagestation.com] > Sent: September 8, 2004 2:19 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] T100P calls with playback starts speaking > before pickup > > > Hi > > I am using a T100P connected to a panasonic phone switch using T1 and the > switch has 4 analog lines

Hi, As you can see in this log the problem is very new to me.. Connected to Asterisk 1.2.5 currently running on volcano (pid = 7874) volcano*CLI> set verbose 4 Verbosity was 0 and is now 4 -- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "SIP/180|60") in new stack -- Called 180 -- SIP/180-d11c is ringing -- SIP/180-d11c

I have an analogue trunk to an AT&T Definity. It has a DISA context defined. From a Definity handset call the analogue port extension 1008 and wait for dial tone from asterisk. It takes between 3&4 rings. Likewise from Asterisk SIP handset <PBX Access No><PBX Extn> takes nearly 10 secs to ring. Is this configurable? Ian Cowley -----Original Message----- From:

Is there a way to configure voicemail to do reminder paging? I would like to configure some voicemail boxes to send an e-mail message to a pager every 10 minutes until the message is retrieved.

I have a simple dial plan to cascade calls when the first phone does not answer: exten => 100,1,Dial(SIP/1000,10,tr) exten => 100,2,Dial(SIP/1000&SIP/1001,10,tr) exten => 100,3,Dial(SIP/1000&SIP/1001&SIP/1002,10,tr) exten => 100,4,Voicemail(u100) Problem is that the once the call goes onto the second and subsequent steps exten 1000 cannot answer the call. When the user

Hi, I've been experiencing trouble with my DAHDI channels for some time and have finally decided to try and resolve the issue. Essentially, the problem I am having is that when a call comes in, and my DAHDI phones therefore ring, Asterisk thinks that one of the handsets has picked up to answer the incoming call - whereas in actual fact it is still on hook. The call then gets instantly