Re: [asterisk-users] Transfer calls "on demand"

You are searching for „Call Pickup“. It is implemented in Asterisk by default. https://wiki.asterisk.org/wiki/display/AST/Call+Pickup Take a look under section „Configuration Options“. Daniel > Am 29.12.2015 um 07:53 schrieb Luca

Re: [asterisk-users] Transfer calls "on demand"

Daniel Heckl schrieb: > You are searching for „Call Pickup“. It is implemented in Asterisk by > default. > > https://wiki.asterisk.org/wiki/display/AST/Call+Pickup > Take a look under > section „Configuration

Re: [asterisk-users] Transfer calls "on demand"

On top of the page: "Call pickup support added in Asterisk 11“ I think that is the problem. I do not know a solution for 1.8, but maybe someone other. > Am 29.12.2015 um 10:20 schrieb Luca Bertoncello : > > Daniel Heckl schrieb: > >> You are

Re: [asterisk-users] Transfer calls "on demand"

Luca Bertoncello wrote: So, my sip.conf: callgroup=1,3-4 ; We are in caller groups 1,3,4 pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 Keep it simple for testing. My sip.conf on a working Asterisk system below: [4220](stemplet)

Re: [asterisk-users] Transfer calls "on demand"

Doug Lytle schrieb: > Keep it simple for testing. My sip.conf on a working Asterisk system below: IT WORKS!!! Thanks a lot! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation

[asterisk-users] Transfer calls "on demand"

Hi list! Right now I configured my Asterisk to forward the calls for the number X to both phones (mine and the phone of my wife). It works, of course, but I'm not enthusiast... I see what we have at office: if one phone rings, other phones in the same group can "catch the call", so that if a

[asterisk-users] Transfer

I am running Asterisk 13.5.0. I have the Transfer working when using the chan_sip support. However, when I try to perform a Transfer using pjsip, it is failing. The one difference I am seeing in the SIP trace is chan_sip automatically sends the Referred-By. PJSIP does not. The switch provider

[asterisk-users] Transfer call question

Hello guys, I have trunk “1, Internal num “99 and MeetMe “1010 now I calling 99 - 89264959635 via 1 /pbx/agi.php: [agi_channel] = PJSIP/99-0012 /pbx/agi.php: [agi_callerid] = 99 /pbx/agi.php: [agi_calleridname] = 99 /pbx/agi.php:

[asterisk-users] Transfer call placed from console (with chan_alsa)

Hi everyone. Having experimented a but with a prototype of a system I described in an earlier thread (Reading DTMF sent by callee during a SIP call), I decided to implement my requirement by transferring the call to another extension. This way, the callee can open the door by pressing #1, and the

[asterisk-users] transfer capabilities

The company that looks after my clients internal phone system has a problem with logging in to the PABX using their data modem. Connection looks like this ISDN PRA from telco - Asterisk - SIP Trunk to my clients Asterisk - my clients Asterisk - E1 port to his old PABX I am planning to use

[asterisk-users] Transfer rights for attended transfers

Recently I asked a question about possibly unwanted calls due to extended transfer rights after attended transfers using DTMF sequences (http://lists.digium.com/pipermail/asterisk-users/2013-September/280536.html). Obviously, transferring with SIP INVITEs (hold + transfer keys) is not

Re: [asterisk-users] Transfer Fraud

I should have mentioned that I am looking at attended transfers. When a Local channel gets created for the announcement it is in a way on behalf of the caller, but has the permissions of an internal channel. Depending on the origin of the call the transfer permissions should then be set

Re: [asterisk-users] Transfer Fraud

On 09/13/2013 04:12 PM, jg wrote: Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context

[asterisk-users] Transfer Fraud

Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context contains Tt. Then an outside call

Re: [asterisk-users] Transfer Fraud

create a separate context for outbound calls. Wouldn't that be more or less identical to my way? I would have to dispatch the channel to see whether it is allowed to enter the outbound context. Maybe I misunderstood something. jg --

Re: [asterisk-users] Transfer Fraud

Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Fraud create a separate context for outbound calls. Wouldn't that be more or less identical to my way? I would have to dispatch the channel to see whether it is allowed to enter the outbound context. Maybe I

[asterisk-users] Transfer cmd via AsyncAGI

Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-0002

Re: [asterisk-users] Transfer only, no outbound calling

Nathan, Yes, SIP.. :-) I ended up deciding to just not allow attended transfer at all since it seemed so hard to deal with. If someone really wants attended transfer they can put the call on hold, dial using the other line then transfer the call on the other line if they want the call on the

[asterisk-users] Transfer only, no outbound calling

OK, it's been a while since I drank from the pool of wisdom hear on the list. After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink. So, on to my question: I have some agents/operators setup in sip.conf which

Re: [asterisk-users] Transfer only, no outbound calling

On Tuesday, April 16, 2013 6:25 PM, Todd Routhier wrote: New Problem, now operators can pick up the previous inbound only line and dial out to anything that matches the patterns I have defined in the context for their extension in sip.conf. What I really need to make work here is

[asterisk-users] Transfer by transfer button, distinguish in dialplan

Hi list! We're using Asterisk in a setup where you can transfer a call via the Asterisk feature, or via the Transfer button on a SIP phone. Both work. However, in my dialplan I cannot distinguish normal calls from calls made by pressing the Transfer button on a phone. To clarify, I would

Re: [asterisk-users] Transfer call issue

Is anybody else experiencing this problem ? -- Thanks, Phil - Original Message - Hello, a client attempted to transfer a call today which failed and returned the channel back to her. When this happened on the console we saw: Got OK on REFER Notify message the version that we

[asterisk-users] Transfer call issue

Hello, a client attempted to transfer a call today which failed and returned the channel back to her. When this happened on the console we saw: Got OK on REFER Notify message the version that we are running is 1.8.9.2. Are you aware of any none issues please with this version as I could

Re: [asterisk-users] Transfer CDRs

Please share if anyone has encountered this cdr issue with call transfer. On Fri, May 18, 2012 at 5:32 PM, [Digital^Dude] ® millennium@gmail.comwrote: Hello, I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine. Each CDR entry of calls that are transferred is

[asterisk-users] Transfer CDRs

Hello, I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine. Each CDR entry of calls that are transferred is repeated once. Every field including uniqueid, calldate, billsec, duration, src, dst, channel, dstchannel is exactly the same. Besides adding a constraint in the

Re: [asterisk-users] Transfer to fax

On Tuesday, March 13, 2012, Kevin P. Fleming kpflem...@digium.com wrote: On 03/13/2012 05:45 PM, Eric Wieling wrote: The faxdetect option is documented in the 1.8 sip.conf.sample. Right, I forgot about that. Now I've really confused things. /me heads back to his hole It was actually added

Re: [asterisk-users] Transfer to fax

So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the detection working. I've added this to my

Re: [asterisk-users] Transfer to fax

To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Transfer to fax So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't

Re: [asterisk-users] Transfer to fax

-users] Transfer to fax So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the detection working. I've

Re: [asterisk-users] Transfer to fax

On 03/13/2012 04:18 PM, Mike Diehl wrote: So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the

Re: [asterisk-users] Transfer to fax

On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi). It's a feature that can be

Re: [asterisk-users] Transfer to fax

On 14/03/2012 5:18 AM, Mike Diehl wrote: So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the

Re: [asterisk-users] Transfer to fax

On 03/13/2012 04:56 PM, Mike Diehl wrote: On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. 'faxdetect' is not a chan_sip configuration option (unlike

Re: [asterisk-users] Transfer to fax

On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote: On 03/13/2012 04:56 PM, Mike Diehl wrote: On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls.

Re: [asterisk-users] Transfer to fax

: [asterisk-users] Transfer to fax On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote: On 03/13/2012 04:56 PM, Mike Diehl wrote: On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I

Re: [asterisk-users] Transfer to fax

On 03/13/2012 05:45 PM, Eric Wieling wrote: The faxdetect option is documented in the 1.8 sip.conf.sample. Right, I forgot about that. Now I've really confused things. /me heads back to his hole -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com

[asterisk-users] Transfer to fax

Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax call comes in, he wants to do some incantation on the keypad and have the call go to the fax machine. As I see it, he has 3 options. 1. (blind?) Transfer it to the fax

Re: [asterisk-users] Transfer to fax

: [asterisk-users] Transfer to fax Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax call comes in, he wants to do some incantation on the keypad and have the call go to the fax machine. As I see it, he has 3 options. 1. (blind

Re: [asterisk-users] Transfer to fax

On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax call comes in, he wants to do some incantation on the keypad and have the call go to the fax machine. As I see it, he has 3 options. 1.

Re: [asterisk-users] Transfer to fax

On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax call comes in, he wants to do some incantation on the keypad

Re: [asterisk-users] Transfer to fax

On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax

Re: [asterisk-users] Transfer to fax

On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both

Re: [asterisk-users] Transfer to fax

On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote: On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a

Re: [asterisk-users] Transfer to fax

On 02/24/2012 05:20 PM, Mike Diehl wrote: On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote: On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike

Re: [asterisk-users] Transfer to fax

On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote: On 02/24/2012 05:20 PM, Mike Diehl wrote: On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote: On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February

[asterisk-users] Transfer to VoiceMail Asterisk 1.6

Hello, I'm using Asterisk 1.6 with Polycom SoundPoint 650, everything is running fine except that I can't program a button on Polycom to transfer inbound call to Voicemail directly. I have the following in my extension.conf exten = _547xx,1,Voicemail(${EXTEN:1}@default,u) Reception can

[asterisk-users] Transfer beep w/ Polycom phone

Hi all. When a user transfers a call by pressing the transfer soft button on their phone, I'd like it to beep at them when the transfer is complete. I've got it turned on in features.conf: xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr

Re: [asterisk-users] Transfer beep w/ Polycom phone

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, April 25, 2011 4:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Transfer beep w/ Polycom phone Hi all

[asterisk-users] Transfer feature dialing out after one digit

Because some users have requested transfer beep confirmations I've switched our phones over to using the asterisk transfer feature instead of the built in transfer functions of the phones. While testing it was working fine, but I changed something in features.conf and suddenly any time I hit

Re: [asterisk-users] Transfer feature dialing out after one digit

What you say...Hose (hose+aster...@bluemaggottowel.com): Because some users have requested transfer beep confirmations I've switched our phones over to using the asterisk transfer feature instead of the built in transfer functions of the phones. While testing it was working fine, but I

[asterisk-users] Transfer Device Data

Hello! I am trying to find out the device name and/or other identifying data to be used in a context when a device transfers the call to new a phone number. From running tests, it looks like the account code variable (${CDR(accountcode)}) is set to the account code of the device that placed the

Re: [asterisk-users] Transfer Device Data

${BLINDTRANSFER} should hold the device name of the one doing the blind transfer. On Sat, Feb 12, 2011 at 6:06 PM, Elliot Murdock murdo...@gmail.com wrote: Hello! I am trying to find out the device name and/or other identifying data to be used in a context when a device transfers the call to

Re: [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target

John Reynolds wrote: Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John, Without seeing any of your dial plan or any of the output from your console during the failed transfer, nobody is going to be able to help. Why don't you

Re: [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target

On Sat, Dec 18, 2010 at 6:51 AM, Doug Lytle supp...@drdos.info wrote: John Reynolds wrote: Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John, Without seeing any of your dial plan or any of the output from your console during the

[asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target

The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux 0.7.2 While I do have FXO capabilities, no POTS lines are connected. When a call comes in

[asterisk-users] Transfer (sip - dahdi) results in moh for dahdi

I have had this problem for a while, so I can't be sure when it started or what was changed. The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux

[asterisk-users] Transfer + speed dial button problem?

Hi everyone, I'm having a bit of an issue after upgrading from asterisk ~1.2.24 to 1.6.2.11, with the old version when the user would go to transfer a call, they would press Transfer, then the speed dial button for the extension, optionally introduce the call, and then press Transfer again to

[asterisk-users] Transfer to non registered extension creates call hangup

Hi list, I have a small problem happening due to call transfer. Whenever the call gets transfered to a remote extension ( which is not registered to asterisk ) it results in hangup(). When asterisk tries to dial the other extension it results in failure making the call cut down :( Is there

[asterisk-users] Transfer fails

Hello list, this is the dialplan : snip exten = s,n,Dial(SIP/test1SIP/test2,,t) snip exten = 10,1,Dial(SIP/test1) exten = 20,1,Dial(SIP/test2) So there is an incoming call that rings SIPaccounts test1 and test2. Account test1 answers and wants to transfer the call to test2. Transfer is : #20

Re: [asterisk-users] Transfer fails

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Friday, July 02, 2010 4:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Transfer fails Hello list

Re: [asterisk-users] Transfer fails

Danny, thank you for you feedback. I have the following setting in sip.conf : limitonpeer = yes and for every sip peer definition I have : asterisk*CLI sip show peer test1 * Name : test1 Realtime peer: Yes, cached Secret : Set MD5Secret: Not set Context :

Re: [asterisk-users] Transfer calls using ##

web interface? Thanks! From: Noah Miller noahisaacmil...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wed, May 5, 2010 1:35:38 PM Subject: Re: [asterisk-users] Transfer calls using ## I have

Re: [asterisk-users] Transfer calls using ##

I have a question about the blind transfer using ##. This works great on our cordless phone, but there have been occasions that we can't transfer using ##. I was able to reproduce the issue by doing the following: 1) Call in from the outside line, 2) Ask the operator to transfer me to an

[asterisk-users] Transfer calls using ##

I have a question about the blind transfer using ##. This works great on our cordless phone, but there have been occasions that we can't transfer using ##. I was able to reproduce the issue by doing the following: 1) Call in from the outside line, 2) Ask the operator to transfer me to an

[asterisk-users] transfer two gsm mobile calls

Hello: when using any fct or sip to gsm gateway, is possible to transfer an incomming call to another number automatically from asterisk say incoming call (gsm gateway) answer; mobile user dial 0XXX then retention 4 SEND KEY then dial XXX then x answer then recover first call 4 send

[asterisk-users] Transfer after pickup

I am probably just being stupid again, but... I have some non-SIP phones which are set up for doing transfers by DTMF, by simply adding T or t to the appropriate Dial options. This works quite well in general. They can also do non-directed call pickup with *8. However, after a call pickup they

[asterisk-users] Transfer Issue with IAX Trunk

I have an IAX trunk configured between 2 Asterisk servers. Everything is working great except if the caller presses # during the call. If they press # the local PBX comes on and says transferring and tries to transfer to a blank extension. Does anyone know how to turn this off? There is no

Re: [asterisk-users] Transfer Issue with IAX Trunk

Lutgring, Sam wrote: I have an IAX trunk configured between 2 Asterisk servers. Everything is working great except if the caller presses # during the call. If they press # the local PBX comes on and says transferring and tries to transfer to a blank extension. Does anyone know how to

Re: [asterisk-users] Transfer Issue with IAX Trunk

Doug Lytle a écrit : Lutgring, Sam wrote: I have an IAX trunk configured between 2 Asterisk servers. Everything is working great except if the caller presses # during the call. If they press # the local PBX comes on and says transferring and tries to transfer to a blank extension.

Re: [asterisk-users] transfer option and pressing #

Alex Samad wrote: Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. My problem is that when you call something like internet banking they want #, but when # is pressed asterisk gets it instead. is there a way around this ? I haven't

Re: [asterisk-users] transfer option and pressing #

On Mon, Jul 13, 2009 at 11:50:00AM -0500, Brent Davidson wrote: Alex Samad wrote: Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. My problem is that when you call something like internet banking they want #, but when # is

[asterisk-users] transfer option and pressing #

Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. My problem is that when you call something like internet banking they want #, but when # is pressed asterisk gets it instead. is there a way around this ? I haven't been able to get

Re: [asterisk-users] transfer option and pressing #

Alex Samad wrote: Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. No, that's simply the order of evaluation. If the caller is inside an Asterisk application that listens for #, it is going to be intercepted and preempted instead

[asterisk-users] Transfer dropping calls

When doing transfers the call drops as follows: 1. I receive a call (internal or not) 2. I dial *2, wait for transfer sound plus dialtone 3. I dial for destinantion person, who pickups the phone 4. We talk to each other 5. I hangup my phone and the call drops if I dial * when talking with

Re: [asterisk-users] Transfer call from analog telephone

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Daniel Bareiro wrote: As I've commented in a previous message, after dial *60 (of *600 to Echo test), I obtain like a tone cut in three parts followed of a continuous tone, causing that I'm incapable to dial the extension completely. The

Re: [asterisk-users] Transfer call from analog telephone

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Tilghman and Grygoriy. Tilghman Lesher escribió: I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer = #1 atxfer = *2 verifying previously that the extension uses the arguments tT with

Re: [asterisk-users] Transfer call from analog telephone

Remember that the time between the two digits is VERY short. You must press those two digits in quick succession or else the requested feature code will not activate. - Or set featuredigittimeout longer. ___ -- Bandwidth and Colocation Provided

[asterisk-users] Transfer call from analog telephone

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm trying to doing a transfer from an analog extension to a SIP extension but until the moment I was not successful. I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer = #1 atxfer = *2

Re: [asterisk-users] Transfer call from analog telephone

On Monday 01 June 2009 04:52:14 Daniel Bareiro wrote: I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer = #1 atxfer = *2 verifying previously that the extension uses the arguments tT with the Dial application and to include the context

[asterisk-users] TRANSFER EVENT ON QUEUE_LOG

Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

Sebastian wrote: Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? If you do, be sure to headline it in all caps. -- Alex Balashov

Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

Forget about this. Is still working. From: Sebastian [mailto:s...@adinet.com.uy] Sent: viernes, 13 de marzo de 2009 10:05 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: TRANSFER EVENT ON QUEUE_LOG Hi, Anyone knows if TRANSFER event on queue_log is

Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

Sent: viernes, 13 de marzo de 2009 10:14 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG Sebastian wrote: Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer

Re: [asterisk-users] Transfer Asterisk 1.6 Telephone IP

Hi, My IP phone has an option to send the flash via DTMF. Enable the sending and the DEBUG I get the following: [ TYPE: Control (4) SUBCLASS: Flash (9) ] [SIP/34314730-b7714f38] [Feb 10 15:12:51] WARNING[29203]: chan_sip.c:5350 sip_indicate: Don't know how to indicate condition 9 [Feb 10

[asterisk-users] Transfer Asterisk 1.6 Telephone IP

Hi List. I have a small problem in using the transfer key transfer of IP Phone in Asterisk 1.6, I think I spend some detail in the configuration but can not find. What happens is, when I do a transfer using the Transfer button, the phone, does not play the music on hold, which is waiting on

[asterisk-users] Transfer in Asterisk 1.6

Hi, All How to enable the transfer in Asterisk 1.6? Not found the module res_features.so. blindxfer = # atxfer = *2 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] Transfer/Park Question.

Daniel Hazelbaker wrote: On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote: Short answer: currently no. Medium answer: I just rolled out 60+ Snom phones (300s and 320s) and we do call parking with DTMF. People were used to just hitting PARK and their phone displaying the park

Re: [asterisk-users] Transfer/Park Question.

Brent Davidson wrote: Also be aware that in 1.2.x and 1.4.x, if you park a call and then pick it up, you can't park it again. At least not with the DTMF I wasn't aware of the inability to re-park calls in 1.4 That could have been a nasty surprise. I would be very interested in

Re: [asterisk-users] Transfer/Park Question.

Doug Lytle wrote: Brent Davidson wrote: Also be aware that in 1.2.x and 1.4.x, if you park a call and then pick it up, you can't park it again. At least not with the DTMF I wasn't aware of the inability to re-park calls in 1.4 That could have been a nasty surprise. I would be

Re: [asterisk-users] Transfer/Park Question.

Brent Davidson wrote: Ok, the patch is working great. Any idea what would make the one step parking not work? I've tried several DTMF combinations in features.conf Check your featuredigittimeout, it defaults to 1/2 second. You may need to increase it. I have mine set to ## to

Re: [asterisk-users] Transfer/Park Question.

On Oct 10, 2008, at 1:00 PM, Brent Davidson wrote: Doug Lytle wrote: I don't remember where I got it (Might have been the bug tracker) that works fine under the current 1.4.x. I had to do a minor change to get it to apply. Copy into Asterisk source directory patch -p0 *.patch rm

Re: [asterisk-users] Transfer/Park Question.

Doug Lytle wrote: Brent Davidson wrote: Ok, the patch is working great. Any idea what would make the one step parking not work? I've tried several DTMF combinations in features.conf Check your featuredigittimeout, it defaults to 1/2 second. You may need to increase it. I

Re: [asterisk-users] Transfer/Park Question.

Daniel Hazelbaker wrote: You won't. The patch I sent you off-list is incomplete, this one is better. I forgot I fixed the parked has timed out option in another patch before I fixed this part. Anyway, make sure when you dial you put k in the dial options (K too if you want both sides

[asterisk-users] Transfer/Park Question.

I've got a situation where I need to use a transfer to the parking lot as hold, but am not going to use BLF indicators on the phone to pick up the parked calls so I need to hear the 3-digit extension after the transfer. I'm using Snom 300 phones and have tried setting a programmable button to

Re: [asterisk-users] Transfer/Park Question.

On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote: I've got a situation where I need to use a transfer to the parking lot as hold, but am not going to use BLF indicators on the phone to pick up the parked calls so I need to hear the 3-digit extension after the transfer. I'm using Snom 300

[asterisk-users] Transfer a call without announce : no sound

When we receive a call from outside (via a sangoma 104d card) and we do a blind transfer, that is without anouncing to the called party , we have no sound either way. Exemple : I take my cell phone to call my * box, it rings on my aastra 9113i phone, I answer. Then I hit the xfer buton, make my

[asterisk-users] Transfer via AMI

I have a call between two people. I know their channel identifier. I want to trasfer a call away from one person and pass it to another person. To start, let's talk about a blind transfer. My system places both outgoing calls to people and bridges them together (cheaper, works via AGI).

Re: [asterisk-users] Transfer

Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Adrian Marsh wrote: Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send

Re: [asterisk-users] Transfer

Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Adrian Marsh wrote: Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send

[asterisk-users] Transfer

Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send the call back to the network, which in turn then routed the call appropriately. It added a transfer-number

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