similar to: Caller ID from Call Pickup

Hi, I'm using the Pickup() application for direct call pickup having the following line in the dialplan: exten => _*88XX,1,Pickup(${EXTEN:2}) It works OK, though I would like to have to get the original caller ID number forwarded to the phone where I do the pickup and have it displayed during the call. Currently the string *88xx remains on the screen of the phone I do the pickup. It

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

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

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

> -----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

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:

I'd like to programmatically do the equivalent of a *8 using either the Manager or AGI interfaces or some other Asterisk interface. In this scenario, a line is ringing, the "pickup" program executes with an argument that is the extension to connect the incoming call to. That extension starts to ring and, when answered, the incoming call is automatically connected to the extension.

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

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:

Hi, I'm trying to achieve the following feature that's common in Avaya systems: A user page all extensions in a full duplex mode- he can hear all, and all can hear him via their phones' speaker. When one of the extensions picks up the handset, the call is bridged between the pager and the person who picked up. All the rest are disconnected. Does anyone have an idea or ever

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

We have calls that originate click-to-dial apps that use the manager interface. As most of you know these apps first ring your handset so that you pickup the handset and then place the outbound call once you have picked up. When they first ring my handset (before me picking up the handset) the call shows as being "from asterisk". Is there any way to change this "from" name to

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

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

Since caller ID does not work with my FXO card, I am wondering if Asterisk supports the following extensions functionality. When a call comes in, I'd like to give the caller an opportunity to enter an extension if he/she knows it; if not, Asterisk will dial one or more default handsets. I know Asterisk can do this, but is it possible to change the default "caller id" when the call

Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get