Re: [Asterisk-Users] transfer & other features

On Sunday 04 June 2006 11:46, Ronald Wiplinger wrote: > *CLI> show features > Builtin Feature Default Current > --- --- --- > Pickup*8 *8 > Blind Transfer# ## > Attended Transfer *2 > One Touch Monito

Re: [Asterisk-Users] transfer & other features

Ronald Wiplinger wrote: What do I miss ??? Your current blind transfer setting is ##, so try ## 632 instead. cYa, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +6

[Asterisk-Users] transfer & other features

*CLI> show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call

Re: [Asterisk-Users] Transfer extensions processing control to Manager

I would have it invoke an AGI script. [incoming_extensions] exten => _X.,1,AGI(ManagerControl) You could have the AGI script have it then jump out to some other context,extension, or priority in the dialplan or have it handle the call itself. ---johann Álvaro Palma wrote: I'm developing an a

Re: [Asterisk-Users] Transfer extensions processing control to Manager

this is what i have in my event driven router. exten => X.,1,Answer() exten => X.,2,MAGI() exten => X.,3,Hangup() look in google for info about MAGI patch regards On 5/23/06, Álvaro Palma <[EMAIL PROTECTED]> wrote: I'm developing an application that monitors the state of the incoming calls us

Re: [Asterisk-Users] Transfer extensions processing control to Manager

no, i think there isn't. But definitely you don't need it. Your incoming calls have to go somewhere, at least in a queue. Your manager app will always be able to redirect or drop the channels as needed. 2006/5/23, Álvaro Palma <[EMAIL PROTECTED]>: I'm developing an application that monitors the sta

[Asterisk-Users] Transfer extensions processing control to Manager

I'm developing an application that monitors the state of the incoming calls using Manager events. So, as a part of it, I need to "override" the control of the extensions by the dialplan itself. The problem is that, if I don't declare the incoming extension, Asterisk hangs up the call by default

Re: [Asterisk-Users] transfer outside of a call?

Juraj Bednar wrote: > I would like to ask, if there's a way to transfer a call from some > external program? I would like to build something like Asterisk Flash > Operator Panel, with the ability to transfer a call using drag and drop. > So I would like to connect to asterisk command line interfa

[Asterisk-Users] transfer outside of a call?

Hello, I would like to ask, if there's a way to transfer a call from some external program? I would like to build something like Asterisk Flash Operator Panel, with the ability to transfer a call using drag and drop. So I would like to connect to asterisk command line interface and transfer one s

Re: [Asterisk-Users] transfer variables

I believe that app_transfer will set this variable, and that is what the OP wanted. On 5/8/06, Moises Silva <[EMAIL PROTECTED]> wrote: yep, that works for BLINDTRANSFER, but not for Attended transfer, may be the variable name should be homologated to a single name. On 5/8/06, C F <[EMAIL PROTEC

Re: [Asterisk-Users] transfer variables

yep, that works for BLINDTRANSFER, but not for Attended transfer, may be the variable name should be homologated to a single name. On 5/8/06, C F <[EMAIL PROTECTED]> wrote: http://www.voip-info.org/wiki/view/BLINDTRANSFER On 5/8/06, David L. West <[EMAIL PROTECTED]> wrote: > Can I get the chann

Re: [Asterisk-Users] transfer variables

http://www.voip-info.org/wiki/view/BLINDTRANSFER On 5/8/06, David L. West <[EMAIL PROTECTED]> wrote: Can I get the channel name (and variables) of the call the iniated the transfer? It looks like the Transfer() application makes a new "local" channel and goes back to the context of the SIP clie

Re: [Asterisk-Users] transfer variables

The variables can be inherited prefixing the variables names with underscore(s) check: http://www.voip-info.org/wiki/index.php?page=Asterisk+variables The "Inheritance of Channel Variables" for the second issue ( get the channel name of the call that initiated the transfer ) please check this:

[Asterisk-Users] transfer variables

Can I get the channel name (and variables) of the call the iniated the transfer? It looks like the Transfer() application makes a new "local" channel and goes back to the context of the SIP client, and in there I need to do things differently based on who started the transfer. _

[Asterisk-Users] Transfer - context/priority

Hi list! When I'm doing transfer, to what context/priority does that call goes? Can it be changed? Is it the same for blind_tr/att_tr/and for transfer that appears when phone replies with - 302 "Moved Temporarily"? The thing is that I'm trying to transfer incoming call from E1 interface back

Re: [Asterisk-Users] transfer call after advise

what you ask for is called "attended transfer". asterisk can do it, but i don't use the manager API so i have no idea how you can realize it. regards christian On Fri, 7 Apr 2006 16:44:41 +0200 nik600 <[EMAIL PROTECTED]> wrote: > i am developing a web application to manage callcenter, i will sh

[Asterisk-Users] transfer call after advise

i am developing a web application to manage callcenter, i will shortly release it on sf.net this is my problem: i will grant to users the possibility to transfer calls to other users using a web interface, for example if SIP/200 is talking with SIP/400 who wants to transfer the call to SIP/500 i

[Asterisk-Users] Transfer Calls - REFER

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. Here's the REFER that the phone at 2944093 sends directly to Asterisk: U 216.186.128.68:5060 -> 216.186.142.203:5060 REFER sip:[EMAIL PROTECTED] SIP/2.0. Via:

[Asterisk-Users] Transfer after group pick-up

I can't transfer call which was picked up with feature - group pick up. I'm running * 1.2.5. The problem is that asterisk doesn't "hear" that I have pressed #1 and doesn't play "transfer" sound for me. "Regular" phone calls I can transfer without problem. Can anybody check is this a BUG? -- T

Re: [Asterisk-Users] transfer calls via Manager Api

nik600 wrote: > nik600 wrote: > but...how can i know the channel used in a specific period from the called? > > for examle...i know SIP/200 but how can i know that the channel is > SIP/200sfhj3e ? either by following the events (NewChannel, Rename, ...) or by issuing a StatusAction that will retu

Re: [Asterisk-Users] transfer incoming call to VM without answering call

This is something that the phone you are using will have to support, and your case with Cisco phones it is NOT supported, the Polycoms support this. If you use a something like FOP (Flash Operators Panel) then you could define an extension that just goes to VM, and drag that call to VM. On 3/23/0

Re: [Asterisk-Users] transfer calls via Manager Api

nik600 wrote: > Is it possible to transfer an existing call from the extension ... > SIP/xxx to another extension in a specific context? you can do this with the redirect action: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect =Stefan ok thanks, it works! but...

[Asterisk-Users] transfer incoming call to VM without answering call

Hi, i'm a newbie running Asterisk 1.2.1 with Cisco 7940/7960 SIP version 7.4 phones. Is there any way in the dial plan or other mystical conf file to allow a user whose extension is presently ringing to press a button on their phone that would instantly send the incoming call to the called use

RE: [Asterisk-Users] transfer calls via Manager Api

: Thursday, March 23, 2006 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] transfer calls via Manager Api ok thanks, it works! but...how can i know the channel used in a specific period from the called? for examle...i know SIP/200 but how can i

Re: [Asterisk-Users] transfer calls via Manager Api

ok thanks, it works! but...how can i know the channel used in a specific period from the called? for examle...i know SIP/200 but how can i know that the channel is SIP/200sfhj3e ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Aster

Re: [Asterisk-Users] transfer calls via Manager Api

nik600 wrote: > Is it possible to transfer an existing call from the extension ... > SIP/xxx to another extension in a specific context? you can do this with the redirect action: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect =Stefan signature.asc Description

[Asterisk-Users] transfer calls via Manager Api

i've seen that opening a socket on the asterisk server i can originate a call from one extension to another in a specific context. Is it possible to transfer an existing call from the extension ... SIP/xxx to another extension in a specific context? thanks

[Asterisk-Users] Transfer to specific park number

HiI'd like to allow users to transfer a call to a specific park number.  This way, the receptionist can tranfer a call park for ext 100 at park number 7100 etc...It seems like this should be fairly simple using the Park(ext) app but it doesn't work for me.  No matter what I extension I use, the sys

[Asterisk-Users] Transfer problems revisited

Hey all, an odd update to my previous note about not being able to transfer if I'm the 'caller'... I set the dial command (in amp) to "T" only... now, If I call into the pbx from outside and reach an extension, I CAN hit # on the calling phone (outside, from-pstn) and get the 'transfer' message

[Asterisk-Users] Transfer (SIP REFER) - AccountCode not available?

I have a snom 320 connected to an Asterisk server. I do some weird things using the AccountCode as an identifier. When the snom makes a call, the AccountCode is used successfully in the dialplan as a variable ${ACCOUNTCODE}. When that same call is transferred using the button on the snom, I see a

Re: [Asterisk-Users] transfer, recording ...

Bartosz Piec wrote: Ronald Wiplinger wrote: exten => 600,1,Dial(${PHONE_LOCAL},60,tr) Type this: exten => 600,1,Dial(${PHONE_LOCAL},60,tTwWr) dial at 600 and see if this helps. If so, change all commands in that way (tT is for transfer, wW is for recording). You must also have sox install

Re: [Asterisk-Users] transfer, recording ...

Ronald Wiplinger wrote: exten => 600,1,Dial(${PHONE_LOCAL},60,tr) Type this: exten => 600,1,Dial(${PHONE_LOCAL},60,tTwWr) dial at 600 and see if this helps. If so, change all commands in that way (tT is for transfer, wW is for recording). You must also have sox installed for calls recordin

Re: [Asterisk-Users] transfer, recording ...

Ronald Wiplinger wrote: does still not do the trick! Show your Dial command from extensions.conf file. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update opti

Re: [Asterisk-Users] transfer, recording ...

Bartosz Piec wrote: Ronald Wiplinger wrote: I tried to transfer a call, pickupcall and onetouch recording, but have not got it to work! You must uncomment the lines in feature.conf (remove the ; character from the beggining). [featuremap] blindxfer => #1; Blind transfer ;disco

Re: [Asterisk-Users] transfer, recording ...

Ronald Wiplinger wrote: I tried to transfer a call, pickupcall and onetouch recording, but have not got it to work! You must uncomment the lines in feature.conf (remove the ; character from the beggining). -- Best regards, Bartosz Piec ___ --Bandwi

[Asterisk-Users] transfer, recording ...

I tried to transfer a call, pickupcall and onetouch recording, but have not got it to work! feature.conf: ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;blindxfer => #1; Blind transfer, default is # ;automon => *1 ; One Touc

[Asterisk-Users] transfer and zap

Hello, some problems with transfer and zap... one hfc-card in NT mode and one fritz isdn-card in server. there is one gigaset SX353 isdn phone on the hfc-card. anybody calls from external via capi and the call is bridged to the zap-device. if you want to transfer the call via R-button on the isdn

[Asterisk-Users] Transfer issue with a Cisco CCM/phone

Hello, We have a mixed environment here consisting of a number of Avaya PBX systems, a group of Cisco Call Managers, an H.323 gateway on a Cisco router, and an Asterisk server. The PBX land is connected to the VoIP land using the Cisco router/H.323 gateway. The Asterisk system is running code fr

RE: [Asterisk-Users] Transfer to meetme on different server

l join a meetme conference room 999 exten => 1002,1,Meetme(999) to choose a dynamic generated room exten => 1002,1,Meetme(|d) Hope that helps Diyanat From: "Steven Langley" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion T

Re: [Asterisk-Users] Transfer sounds - notifications

Tomislav Parcina wrote: When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear "transfer". I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back

Re: [Asterisk-Users] Transfer sounds - notifications

On Wed, January 11, 2006 12:46, Tomislav Parcina said: > When I try to make attendend transfer (*2) this what hapends. > I press *2 other person goes on hold and I hear "transfer". I press > extension number and that extension starts to ring but I don't hear > anything. If nobody picks up that phon

[Asterisk-Users] Transfer sounds - notifications

When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear "transfer". I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking

[Asterisk-Users] Transfer to meetme on different server

Title: Transfer to meetme on different server Hi there I am using IAX2 based phones and am wondering if the following is possible: 1.  User registers with Server 1 2.  User dials an extension on Server 1 3.  Extension transfers call to an extension on Server 2, which transfers

Re: [Asterisk-Users] transfer application

Bill Michaelson wrote: I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXX,n(nocid),transfer(1000) exten => _NXXN

[Asterisk-Users] transfer application

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXX,n(nocid),transfer(1000) exten => _NXXNXX,n,noop(boo,${TRAN

Re: [Asterisk-Users] Transfer

Victor Alvarez schrieb: Hi, I'm afraid I don't know how to use the command Transfer. I am also interested how the command "Transfer" should be used. I am aware of the possibility to add the option t or T to dial, so #33 transfers the call to extension 33. Is there any use of this command

Re: [Asterisk-Users] Transfer

Hi! > Can * transfer call if I use canreinvite=yes in sip.conf? > Can * start "automon" (recording) if I use canreinvite? > > If answers are no, then which one did you chouse for your configuration? > Do you use "canreinvite=yes" so you can't do those stuff or you don't > use this so you have h

Re: [Asterisk-Users] Transfer

Hi Tomislav, If you want to do recording and are worried about high processor load when keeping asterisk in the media path with SIP, you might check out http://www.oreka.org which is an open source voip recorder that can run on a separate machine altogether. Henri On 23/12/05, Tomislav Parcina <

Re: [Asterisk-Users] Transfer

Tomislav Parcina a écrit : Can * transfer call if I use canreinvite=yes in sip.conf? My understanding is that canreinvite only redirects the media path. Signaling and media are separate with SIP (which is what makes it so nice by the way). Can * start "automon" (recording) if I use canre

[Asterisk-Users] Transfer

Can * transfer call if I use canreinvite=yes in sip.conf? Can * start "automon" (recording) if I use canreinvite? If answers are no, then which one did you chouse for your configuration? Do you use "canreinvite=yes" so you can't do those stuff or you don't use this so you have high processor loa

Re: [Asterisk-Users] Transfer/take call to/from other phone

Hi, To pick up another persons phone that is ringing dial '*8' followed by their extension. To do an attended transfer dial '*2' followed by the extension... Hope that helps Denny Schierz wrote: hi, Quoting Chuck Bunn <[EMAIL PROTECTED]>: Push the '#' key followed by the extension for a

Re: [Asterisk-Users] Transfer/take call to/from other phone

This is what I use. You pre-pend a '4' to the extension number (I used that because that is how our old pbx worked). There is a number you can use that will pickup any ringing extension but I forgot what that is. It should be listed on the asterisk wiki for Pickup. exten => _4XXX,1,Pickup(${EX

Re: [Asterisk-Users] Transfer/take call to/from other phone

hi, Quoting Chuck Bunn <[EMAIL PROTECTED]>: Push the '#' key followed by the extension for a blind transfer. absolut perfect, thanks :-) .Is there also a shortcut, to take a phone call from other phones to me? cu denny This

Re: [Asterisk-Users] Transfer/take call to/from other phone

Hi, Push the '#' key followed by the extension for a blind transfer. Thanks Denny Schierz wrote: hi, my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)? Or when the phone 401 rings, bu

[Asterisk-Users] Transfer/take call to/from other phone

hi, my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)? Or when the phone 401 rings, but my boss is not there, how can i take the phonecall from 401 to 400? Do i need special options in my e

[Asterisk-Users] Transfer problem...

I am having issues with transferring calls. I can transfer outgoing calls, but not incoming calls. * 1.2 / BRIstuff 0.3 PRE1 / 1 HFC cards Connecting calls between the 2 cards (1 NT mode, 1 TE mode) The caller is always able to xfer the call, the callee only sometimes (have not yet been able to

[Asterisk-Users] Transfer call error

 When I call a internal Sip telephone, the calling transfer to Teleohone external, but Asterisk show this error: Executing Dial("SIP/201-1e2a", "ZAP/g1/3472543320|60") in new stack     -- Requested transfer capability: 0x00 - SPEECH     -- Called g1/3472543320 Nov 30 15:52:09 WARNING[1866]: chan_z

Re: [Asterisk-Users] Transfer directly to voicemail (blind transfer)?

Doug wrote: >Another method would be to prefix with a digit instead of suffix with an >"*". For us, all of our extensions are three digits and begin with a 5 >or a 6 (5xx or 6xx). To transfer to voice mail we stick an eight in >front of the extension (85xx or 86xx). It works well for us. >

Re: [Asterisk-Users] Transfer directly to voicemail (blind transfer)?

At 09:11 10/5/2005, Don Pobanz, wrote: >Doug wrote: >> Hi, >> >> Have looked around for info about this: >> >> >gium.com> >> >> >> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail >> >> If we are usi

Re: [Asterisk-Users] Transfer directly to voicemail (blind transfer)?

Doug wrote: Hi, Have looked around for info about this: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the exten

[Asterisk-Users] Transfer directly to voicemail (blind transfer)?

Hi, Have looked around for info about this: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can

Re: [Asterisk-Users] Transfer calls from cellphone

Brilliant, thanks for the response. Arnar >>> [EMAIL PROTECTED] 8.9.2005 13:26:52 >>> b is possible. See res_features.conf for more information on transferring via DTMF. c is not yet possible. This would require shared call appearances which isn't yet implemented. On 9/8/05, Arnar Birgiss

Re: [Asterisk-Users] Transfer calls from cellphone

 b is possible. See res_features.conf for more information on transferring via DTMF.    c is not yet possible. This would require shared call appearances which isn't yet implemented.  On 9/8/05, Arnar Birgisson <[EMAIL PROTECTED]> wrote: Hello,Avaya has a nice feature that allows you toa) ring bot

[Asterisk-Users] Transfer calls from cellphone

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up

[Asterisk-Users] Transfer a call from cell phone (pseudo-disa)

  I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service.  Is there a way

[Asterisk-Users] Transfer a call from cell phone (pseudo-disa)

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service.  Is there a way to

Re: [Asterisk-Users] Transfer to outside line.

I believe that on the digital receptionist configuration the last option (after extension/voicemail/) is for a customization command. And you can use it to send the user to a extension context you define (in "extension_custom.conf" if memory is good) with a destination number specified on this

Re: [Asterisk-Users] Transfer to outside line.

.   Ariel - Original Message - From: Tim King To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, August 03, 2005 5:24 PM Subject: RE: [Asterisk-Users] Transfer to outside line. I tried this solution, although ti ac

RE: [Asterisk-Users] Transfer to outside line.

t: Re: [Asterisk-Users] Transfer to outside line.   I think what you want is called DISA http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA DISA (Direct Inward System Access) Allows someone from outside the telephone switch (PBX) to obtain an "internal" system dialtone and

RE: [Asterisk-Users] Transfer to outside line.

: Wednesday, August 03, 2005 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Transfer to outside line.   This is simple since your using AMP, you can create a ring group to dial that number out for you.  First create your ring group lets put number

Re: [Asterisk-Users] Transfer to outside line.

Ariel - Original Message - From: Tim King To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, August 03, 2005 10:12 AM Subject: [Asterisk-Users] Transfer to outside line.Finally got everything up and run with the help of Manny Wise last

RE: [Asterisk-Users] Transfer to outside line.

: Wednesday, August 03, 2005 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Transfer to outside line.   I think what you want is called DISA http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA DISA (Direct Inward System Access) Allows

Re: [Asterisk-Users] Transfer to outside line.

August 03, 2005 10:12 AM Subject: [Asterisk-Users] Transfer to outside line. Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a

Re: [Asterisk-Users] Transfer to outside line.

I think what you want is called DISA http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA DISA (Direct Inward System Access) Allows someone from outside the telephone switch (PBX) to obtain an "internal" system dialtone and to place calls from it as if they were placing a call from

[Asterisk-Users] Transfer to outside line.

Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the c

[Asterisk-Users] Transfer and CDR's

Hello   I have a problem with bad CDR's after transfer of call.   This is an example:   I've called from 616222820 to 616222821. Next I've called from 616668020 to 060034 and then I've transfered the call. I think, that I should received two CDR's where:  in first CDR source=616222820 a

Re: [Asterisk-Users] Transfer

In your extension.conf 35,1Dial(SIP/33,Ttr) in order to transfert during a call  #33 Victor Alvarez a écrit : Hi,  I'm afraid I don't know how to use the command Transfer. I have a couple of SIP users in the system and although exten => 35,1,Dial(SIP/33) works fine, exten => 3

Re: [Asterisk-Users] Transfer

i know that there are extensive rework on the transfer in SIP at the moment. --On Tuesday, June 21, 2005 13:40:47 +0100 Victor Alvarez <[EMAIL PROTECTED]> wrote: Hi, I'm afraid I don't know how to use the command Transfer. I have a couple of SIP users in the system and although exten => 35,

RE: [Asterisk-Users] Transfer

> 35,1,Dial(SIP/33) would be a way to transfer the incoming call from 35 to > 33, but what I want to do is to get 33 dialplan, not to dial 33. I mean, > if 33 is 33,1,Voicemail that's what I would like to execute when calling > 35. > > Could anybody help me? Do you mean if you dial 35, you want

[Asterisk-Users] Transfer

Hi,  I'm afraid I don't know how to use the command Transfer. I have a couple of SIP users in the system and although exten => 35,1,Dial(SIP/33) works fine, exten => 35,1,Transfer(33) just don't work. All the description in the wiki is 'Transfer(exten)' without a single example.    35,1,Dial

Re: [Asterisk-Users] #(transfer) no longer working

On 20:55, Thu 16 Jun 05, Florian Overkamp wrote: > Hi Michiel, > > > -Original Message- > > Anyone who can help me with this ? > > I tried everything :( > > > > exten => s,4,Dial(Local/[EMAIL PROTECTED],5,tTr) > > > exten => s,5,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],10,tTr

RE: [Asterisk-Users] #(transfer) no longer working

Hi Michiel, > -Original Message- > Anyone who can help me with this ? > I tried everything :( > > exten => s,4,Dial(Local/[EMAIL PROTECTED],5,tTr) > > exten => s,5,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],10,tTr) Have you tried using the /n parameter for chan_local ? I've no

Re: [Asterisk-Users] #(transfer) no longer working

Anyone who can help me with this ? I tried everything :( On 14:26, Tue 14 Jun 05, Michiel van Baak wrote: > Hi list, > > For months everything worked super here in our setup. > This week I implemented some new idea in our webbased > calendar system. I thought it would be nice to have an > option

Re: [Asterisk-Users] Transfer differences between BudgeTone101 and Snom190

You can try the ${RDNIS} variable. On Tue, 2005-06-07 at 00:32 +0200, Elwin Andriol wrote: > Hello all, > > This email is intended rather informative than questioning. While > developing some script-generated dial plan, we figured out that there > are differences between Snom 190's and BudgeTo

[Asterisk-Users] Transfer differences between BudgeTone101 and Snom190

Hello all, This email is intended rather informative than questioning. While developing some script-generated dial plan, we figured out that there are differences between Snom 190's and BudgeTone 101's relating to transfers. It appeared that the 190's will have their own 'Caller ID' set as t

[Asterisk-Users] Transfer of Calls Between Legacy PBX and Asterisk

Hi, We have a scenario where we receive calls from 2 different places: 1- Avaya IP Office 2- CIC Interactive Intelligence PBX and the calls are transfer automatically to an Asterisk Box. The problem we are experiencing is that more that half of those calls come with Echo and Jitter. For outbounds

[Asterisk-Users] Transfer of Calls Between Legacy PBX and Asterisk

Hi, We have a scenario where we receive calls from 2 different places: 1- Avaya IP Office 2- CIC Interactive Intelligence PBX and the calls are transfer automatically to an Asterisk Box. The problem we are experiencing is that more that half of those calls come with Echo and Jitter. For outbounds

[Asterisk-Users] Transfer from/to a queue

Good Morning,   Does anyone know if it is possible to transfer a caller from queue 1 to queue 2?   Regards Jenn Hales ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-user

[Asterisk-Users] transfer queues agents

Good day all This is what i got off the net about queues and agents "Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination

Re: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

Joseph wrote: On Sat, 2005-04-30 at 13:23 -0400, Time Bandit wrote: I was wondering if there was a way to have incoming calls to my PSTN line be "transferred" to a voip line? I would like to make it so that as soon as the pstn call is recieved it will switch the call to the voip line, thus freeing

Re: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

On Sat, 2005-04-30 at 13:23 -0400, Time Bandit wrote: > > I was wondering if there was a way to have incoming > > calls to my PSTN line be "transferred" to a voip line? > > > > I would like to make it so that as soon as the pstn > > call is recieved it will switch the call to the voip > > line, th

RE: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

> > I was wondering if there was a way to have incoming > calls to my PSTN line be "transferred" to a voip line? > > I would like to make it so that as soon as the pstn > call is recieved it will switch the call to the voip > line, thus freeing up the pstn line to get more calls. > Kind of like roa

Re: [Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

> I was wondering if there was a way to have incoming > calls to my PSTN line be "transferred" to a voip line? > > I would like to make it so that as soon as the pstn > call is recieved it will switch the call to the voip > line, thus freeing up the pstn line to get more calls. > Kind of like roam

[Asterisk-Users] transfer pstn call to voip line, thus freeing up pstn line

I was wondering if there was a way to have incoming calls to my PSTN line be "transferred" to a voip line? I would like to make it so that as soon as the pstn call is recieved it will switch the call to the voip line, thus freeing up the pstn line to get more calls. Kind of like roaming. Tom ___

Re: [Asterisk-Users] Transfer of incoming call from external tointernal number

t; > Sent: Wednesday, April 20, 2005 9:26 AM > > To: Asterisk-Users@lists.digium.com > > Subject: [Asterisk-Users] Transfer of incoming call from external > > tointernal number > > > > When I place a call on my softphone to a external number the call is > > plac

RE: [Asterisk-Users] Transfer of incoming call from external tointernal number

05 9:26 AM > To: Asterisk-Users@lists.digium.com > Subject: [Asterisk-Users] Transfer of incoming call from external > tointernal number > > When I place a call on my softphone to a external number the call is > placed, when I click transfer, dial internal extrention (e.g. 202) >

[Asterisk-Users] Transfer of incoming call from external to internal number

When I place a call on my softphone to a external number the call is placed, when I click transfer, dial internal extrention (e.g. 202) then hit transfer again, the call is transfered to the 202 extention fine. However, when the other way Internal call comes in, extension 201 answers, and transfer

[Asterisk-Users] Transfer of incoming call from external to internal number

When I place a call on my softphone to a external number the call is placed, when I click transfer, dial internal extrention (e.g. 202) then hit transfer again, the call is transfered to the 202 extention fine. However, when the other way Internal call comes in, extension 201 answers, and transfer

[Asterisk-Users] Transfer a call in the IVR

How do I transfer or forward a call that is in the IVR and connect it to a static external phone number? The call would come in on a POTS line into an x100p and could go out using 3-way calling or another POTS line connected to another x100p. Please help!!!   Paul       _

Re: [Asterisk-Users] Transfer a call ? Am I lookingfortheflashcommand ?

> BTW: Whats actually that " SendDTMF" ? thing ? http://www.voip-info.org/wiki-Asterisk+cmd+sendDTMF DTMF definition : http://en.wikipedia.org/wiki/DTMF N.B.: please try to trim your answers, the message is becoming pretty long hth ___ Asterisk-Users m

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