similar to: Playing a sound file on handset pickup

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

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:

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

Does anybody know how to build a handset to sound card adapter ? Thanks! --------------------------------------- Norberto Harmath Centro de Comunicacion Cientifica Universidad de Buenos Aires

Hi, I've recently bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1. I just noticed something, which I first blamed on Asterisk and NATs (a 2 second silence at the beginning of a call). Something I've noticed also on my old phone (which is having the same problem now, but its also been upgraded). My keys are sticky. Simple as that. Sometimes I press a number

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've tested the S100U unit in two different computers both with different USB subsystems. The one it worked the best in used the usb-uhci USB driver. The other system uses the usb-ohci USB driver. To make sure it was not the installation, I used the same hard drive that had asterisk in both systems. And I moved the X101P and S100U between both

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

Hi, I'm trying to do the following but can't quite get it right: 1) Callers rings DID number 2) Asterisk rings the appropriate channel for 30 second, if no answer sends to voicemail (no problem up to here, of course) 3) IF the channel is answered Asterisk plays an audio file 4) Asterisk connects caller with me I need to do this to "cover up" the delay within the first few

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, > 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

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

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 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

At one point I did have this kit working but since upgrading to the latest Asterisk, it no longer seems to. I had the following problems after several reinstallations: - USB adaptor had a proper dialtone, asterisk recognised the pickup, but pressing keys on the handset had no effect - USB adaptor produced a strange horrible tone, not a dialtone; pressing keys has no effect - USB

Even better you can set your firefly softphone to auto answer so that you don't even need to be near the pc to answer. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Linn Boyd Sent: Friday, March 04, 2005 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

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

https://bugzilla.samba.org/show_bug.cgi?id=8456 Summary: improve --link-dest bahaviour Product: rsync Version: 3.0.8 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P5 Component: core AssignedTo: wayned at samba.org ReportedBy: toralf.foerster at gmx.de