[asterisk-users] features.conf disconnect and local channels
Asterisk 16.1
This statement appears in the features.conf doc: "Note that the DTMF
features listed below only work when two channels have answered and are
bridged together. They can not be used while the remote party is ringing
or in progress. If you require this feature you can use chan_local in
combination with Answer to accomplish it."
I need attended transfer and disconnect from features.conf to work.
Below is what I came up with that seems to work fine. Is there a better
way? This seems a bit verbose.
[InternalSets]
exten =>298,1,NoOp()
same =>n,Dial(Local/M${EXTEN}@InternalSets/nj,,H)
exten =>M298,1,NoOp()
same =>n,Answer()
same =>n,GoSub(sub-voicemail,start,1(${MITCHIPHONE},${EXTEN:1}))
exten =>299,1,NoOp()
same =>n,Dial(Local/M${EXTEN}@InternalSets/nj,,H)
exten =>M299,1,NoOp()
same =>n,Answer()
same =>n,GoSub(sub-voicemail,start,1(${MLCX450},${EXTEN:1}))
[sub-voicemail]
do some checks and then Dial or send to voicemail.
--
Mitch
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[asterisk-users] Features.conf and variable length DTMF sequences
Hello,
I have an Asterisk 13-enabled system.
1. Using features.conf application map (or something else), is it possible
to define a single map matching several DTMF sequences, such as in the
imaginary example bellow ?
features.conf:
foobar => _*123.,peer,Gosub,"foobar,s,1"
_*123. would match DTMF sequences *1234 or *12345 or anything starting with
*123
2. I used the mapping bellow:
foobar => *123,peer,Gosub,"foobar,s,1"
In my diaplan, I edited something like:
[foobar]
exten = s,1,Read(FOO,silence/1)
same = n,Noop(FOO is ${FOO})
When I originate a call (with "channel originate Local/1@test/n" or
"channel originate Local/1@test") to an other Asterisk box which sends a
*123ww456 DTMF sequence, I can see that:
- the DTMF is coming in,
- its *123 prefix is recognized and triggers my foobar routine (see above)
- application Read is executed,
- next DTMF digits are coming in but are not read by Read application as if
I was reading on one channel and digits were coming in an other channel.
Any help on this ?
Best regards
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Re: [asterisk-users] features.conf and mixmonitor stop and start
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik i...@pack-net.co.uk:
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename into a variable and give it full
inheritance so you can resume recording to the same file (using the a
option)
On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have a recording started in the dialplan with the MixMonitor
application. I want to be able to stop it during a call and maybe restart
it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is running?
[featuremap]
automixmon = *1
I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:
= {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
}
[applicationmap]
pauseMonitor = #1,self/both,stopMixMonitor
Any advice?
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f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
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37 Ducie Street
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COMPANY REG NO. 04920552
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Re: [asterisk-users] features.conf and mixmonitor stop and start
On 28 August 2014 07:56, Leandro Dardini ldard...@gmail.com wrote:
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik i...@pack-net.co.uk:
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename into a variable and give it full
inheritance so you can resume recording to the same file (using the a
option)
On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have a recording started in the dialplan with the MixMonitor
application. I want to be able to stop it during a call and maybe restart
it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is
running?
[featuremap]
automixmon = *1
I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:
= {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
}
[applicationmap]
pauseMonitor = #1,self/both,stopMixMonitor
Any advice?
extensions.conf:
[macro-pause-recording]
exten = s,1,Verbose(Stopping Recording)
exten = s,n,StopMixMonitor()
[macro-unpause-recording]
exten = s,1,Verbose(Resuming Recording)
exten = s,n,MixMonitor(${REC_FILE_NAME},a)
features.conf
StopMixMonitor = #00,peer/both,Macro(pause-recording)
;
MixMonitor = #01,peer/both,Macro(unpause-recording)
Make sure you set REC_FILE_NAME early on with a double underscore and
remember to add Set(__DYNAMIC_FEATURES=MixMonitor#StopMixMonitor) early on
too
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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[asterisk-users] features.conf and mixmonitor stop and start
Hello,
I have a recording started in the dialplan with the MixMonitor application.
I want to be able to stop it during a call and maybe restart it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is running?
[featuremap]
automixmon = *1
I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:
= {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
}
[applicationmap]
pauseMonitor = #1,self/both,stopMixMonitor
Any advice?
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Re: [asterisk-users] features.conf and mixmonitor stop and start
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename into a variable and give it full
inheritance so you can resume recording to the same file (using the a
option)
On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have a recording started in the dialplan with the MixMonitor
application. I want to be able to stop it during a call and maybe restart
it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is running?
[featuremap]
automixmon = *1
I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:
= {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
}
[applicationmap]
pauseMonitor = #1,self/both,stopMixMonitor
Any advice?
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--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] Features.conf - Blind Transfer
Hi guys, I'm trying to get blind transfer to work and automatically transfer call to another number on key sequence press. Extensions.conf_snippet [from-pstn] exten = _0399377744,1,Set(__DYNAMIC_FEATURES=blindxfer) exten = _0399377744,n,Set(__GOTO_ON_BLINDXFR=to-pstn ^0388924326^1) exten = _0399377744,n,dial(SIP/0399377704@c5400-02,T) [to-pstn] Exten = _XXX.,1,dial(sip/0388924326@ c5400-01) Features.conf_snippet [featuremap] blindxfer = #1 on #1 all I get is silence, and debug shows call going to 'i' ast ver 1.4.24 Appreciate your help Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf - parkedcalls - transfer
2010/6/21 Aksel Celasun ak...@abacus-it.no Hello dear list. I am having issues on parkedcalls. I am using a Cisco SPA525G as a test phone, and I have the transfer button there when I am in a call, But when I want to transfer the current call I am in, I push the transfer button, and onscreen I se “Enter Number”, and if I enter ex sip 200, I have to wait Don't you have an OK button somewhere available when dialing ? To my knowledge, most SIP phone can both use a timer and a dedicated key to send a dialed number. Cheers Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I can’t see any button on the Cisco phone which will function like a “direct transfer now”, do I have to wait…? And, secondly, is there a another way to do transfer/send to another sip phone? Ex. Push *200 and the SIP phone will directly call SIP/200. Or push *401 and the Sip phone will directly call SIP401? Default features.conf context. Thank you. Med vennlig hilsen / Best regards Abacus IT AS - din Visma Software Partner - your Visma Software Partner *Tor Aksel **Celasun* Mobilnummer/cell phone: (+47) 900 15 103 Sentralbord/Support 4000 1850 ak...@abacus-it.no -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf - parkedcalls - transfer
Hello dear list. I am having issues on parkedcalls. I am using a Cisco SPA525G as a test phone, and I have the transfer button there when I am in a call, But when I want to transfer the current call I am in, I push the transfer button, and onscreen I se Enter Number, and if I enter ex sip 200, I have to wait Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I can't see any button on the Cisco phone which will function like a direct transfer now, do I have to wait...? And, secondly, is there a another way to do transfer/send to another sip phone? Ex. Push *200 and the SIP phone will directly call SIP/200. Or push *401 and the Sip phone will directly call SIP401? Default features.conf context. Thank you. Med vennlig hilsen / Best regards Abacus IT AS - din Visma Software Partner - your Visma Software Partner Tor Aksel Celasun Mobilnummer/cell phone: (+47) 900 15 103 Sentralbord/Support 4000 1850 ak...@abacus-it.nomailto:ak...@abacus-it.no -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf - parkedcalls - transfer
On Mon, Jun 21, 2010 at 2:27 AM, Aksel Celasun ak...@abacus-it.no wrote: I am using a Cisco SPA525G as a test phone, and I have the transfer button there when I am in a call, But when I want to transfer the current call I am in, I push the transfer button, and onscreen I se “Enter Number”, and if I enter ex sip 200, I have to wait Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I can’t see any button on the Cisco phone which will function like a “direct transfer now”, do I have to wait…? On the Cisco 79xx series phones, you would modify a config file called dialplan.xml. I think on the SPA5xx series you can configure this parameter either in the main config file or from the web interface. You need to look for something like dialplan or dial plans, etc. This controls the timeout when entering digits. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf - parkedcalls - transfer
At 12:27 AM 6/21/2010, you wrote: Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I cant see any button on the Cisco phone which will function like a direct transfer now, do I have to wait ? On my Aastra phones, I press Transfer 101 Transfer. So you might just try hanging up or pressing transfer again. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf - parkedcalls - transfer
And I can't see any button on the Cisco phone which will function like a direct transfer now, do I have to wait...? Thank you for your reply. In my Dialplan menu on the SPA525g, I have a field where the input are, and I must say, I don't know if this is the right one, but the field contains this: Dial plan: (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) What should I edit, if this is the right code On the Cisco 79xx series phones, you would modify a config file called dialplan.xml. I think on the SPA5xx series you can configure this parameter either in the main config file or from the web interface. You need to look for something like dialplan or dial plans, etc. This controls the timeout when entering digits. -- Thanks, --Warren Selby http://www.selbytech.com Best regards Aksel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf - parkedcalls - transfer
Hello, and thank you for your response. When I push transfer, the buttons with the function transfer disappears, and then I enter the sip number, Wait 10 seconds and then it transfers with the MOH in the background, when the connection/channel is made, Then transfer button is revealed again suddenly, and then I can push transfer again, and it transfers... =). I'm gonna try with callback function when no-answer, and the hangup option which you mentioned. Best regards Aksel -Opprinnelig melding- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Ira Sendt: 21. juni 2010 19:16 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] features.conf - parkedcalls - transfer At 12:27 AM 6/21/2010, you wrote: Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I can't see any button on the Cisco phone which will function like a direct transfer now, do I have to wait.? On my Aastra phones, I press Transfer 101 Transfer. So you might just try hanging up or pressing transfer again. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf : feature map == getting feature to work
just a hint. you might have # assigned the moh in feature.conf and #3 to starting the recording. check your feature.conf and makesure that # isn't assigned to anything. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 7 sep 2009 om 20:40 heeft jonas kellens jonas.kell...@telenet.be het volgende geschreven:\ Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/ jonaskl/recording,m In my dialplan : [from-HostAst] exten = s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten = s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able to record the conversation but when I press these keys on my Grandstream phone, the following is displayed on the CLI : [Sep 7 20:33:49] WARNING[10870]: res_musiconhold.c:665 get_mohbyname: Music on Hold class '/var/samba/profiles/jonaskl/ recording' not found Don't know where this comes from... I have tried the same with *3. Same output on the CLI. Yes, I have restarted Asterisk after changes in features.conf. It's not my Grandstream or the DTMF-input because *8 for picking up a ringing phone works well... When I set : opnemencallee = #*3,self/callee,Monitor,wav,/var/samba/profiles/ jonaskl/recording,m and I press #*3, nothing happens... No output on the CLI. There's not much info. I followed the instructions on voip-info.org (which are the same as in features.conf). The module res_features is loaded. Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf : feature map == getting feature to work
Erik, I have placed everything in features.conf in comment ( ; ). Still when I run show features, I get this : clarkconnect*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * Park Call clarkconnect*CLI Dynamic Feature Default Current --- --- --- opnemencaller no def #* opnemencallee no def #* clarkconnect*CLI Call parking *CLI Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-750 So there might be indeed a mix-up. It seems to me that the default features, like *8 for pickup, cannot be disabled. Even when in comment they still work ! I have restarted Asterisk after changes in features.conf. Jonas. On Tue, 2009-09-08 at 09:17 +0200, Erik de Wild wrote: just a hint. you might have # assigned the moh in feature.conf and #3 to starting the recording. check your feature.conf and makesure that # isn't assigned to anything. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf : feature map == getting feature to work
8 sep 2009 kl. 10.17 skrev jonas kellens: Erik, I have placed everything in features.conf in comment ( ; ). Still when I run show features, I get this : clarkconnect*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * Park Call clarkconnect*CLI Dynamic Feature Default Current --- --- --- opnemencaller no def #* opnemencallee no def #* clarkconnect*CLI Call parking *CLI Parking extension :700 Parking context :parkedcalls Parked call extensions: 701-750 So there might be indeed a mix-up. It seems to me that the default features, like *8 for pickup, cannot be disabled. Even when in comment they still work ! I have restarted Asterisk after changes in features.conf. The *8 is indeed a hard-coded feature in many channels, but only works if you've enabled callgroups and pickupgroups in the channel configurations, and it's a dialstring, not a DTMF code. The blind transfer functionality is enabled by the 'tT' options to dial(), like the disconnect call feature. The dynamic features is something you've enabled in features.conf. They require dialplan intervention to work. Call parking are again extensions, not DTMF codes. To work, requires you to include the context in the dialplan for a device/line. So apart from the dynamic features, what you see here is the Asterisk defaults that can be changed in features.conf, but will always be there. They only work if you enable them in other files. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf : feature map == getting feature to work
When I enable the automon-feature (*1) the callee can start recording the conversation. No problem there. But I can't get my user-defined features to work. I have setup the following test-feature in features.conf : [applicationmap] testfeat = *3,self/callee,Playback,tt-weasels I have the following in my dialplan : [from-HostAst] exten = s,1,Set(__DYNAMIC_FEATURES=testfeat) exten = s,n,NoOp(...) exten = s,n,NoOp(...) exten = s,n,Dial(SIP/grandstream,30) When pressing *3 on the Grandstream (the callee), the CLI shows nothing : [Sep 8 11:22:14] -- Executing [...@from-hostast:1] Set(IAX2/hostedasterisk-12746, __DYNAMIC_FEATURES=testfeat) in new stack [Sep 8 11:22:14] -- Executing [...@from-hostast:2] NoOp(IAX2/hostedasterisk-12746, ...) in new stack [Sep 8 11:22:14] -- Executing [...@from-hostast:3] NoOp(IAX2/hostedasterisk-12746, ...) in new stack [Sep 8 11:22:14] -- Executing [...@from-hostast:4] Dial(IAX2/hostedasterisk-12746, SIP/grandstream|30) in new stack [Sep 8 11:22:14] -- Called grandstream [Sep 8 11:22:14] -- SIP/grandstream-083d5c10 is ringing [Sep 8 11:22:22] -- SIP/grandstream-083d5c10 answered IAX2/hostedasterisk-12746 ... nothing happens when pressing *3... [Sep 8 11:22:52] == Spawn extension (from-HostAst, s, 4) exited non-zero on 'IAX2/hostedasterisk-12746' [Sep 8 11:22:52] -- Hungup 'IAX2/hostedasterisk-12746' With the automon-feature, it works well : [Sep 8 11:18:35] -- User hit '*1' to record call. filename: wav| auto-1252401515-s-IAX2-hostedasterisk-9817|m [Sep 8 11:18:46] -- User hit '*1' to stop recording call. What am I doing wrong so that my user-defined features don't work ? Even this simple Playback(tt-weasels) won't work. Jonas. On Mon, 2009-09-07 at 16:03 -0500, Anthony Messina wrote: On Monday 07 September 2009 13:40:16 jonas kellens wrote: [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m FeatureName = DTMF_sequence,ActivateOn[/ActivatedBy],Application[,AppArguments[,MOH_Class]] it looks like /var/samba/profiles/jonaskl/recording is in the spot for [,MOH_Class] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf : feature map == getting feature to work
Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten = s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten = s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able to record the conversation but when I press these keys on my Grandstream phone, the following is displayed on the CLI : [Sep 7 20:33:49] WARNING[10870]: res_musiconhold.c:665 get_mohbyname: Music on Hold class '/var/samba/profiles/jonaskl/recording' not found Don't know where this comes from... I have tried the same with *3. Same output on the CLI. Yes, I have restarted Asterisk after changes in features.conf. It's not my Grandstream or the DTMF-input because *8 for picking up a ringing phone works well... When I set : opnemencallee = #*3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m and I press #*3, nothing happens... No output on the CLI. There's not much info. I followed the instructions on voip-info.org (which are the same as in features.conf). The module res_features is loaded. Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf : feature map == getting feature to work
On Monday 07 September 2009 13:40:16 jonas kellens wrote: [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m FeatureName = DTMF_sequence,ActivateOn[/ActivatedBy],Application[,AppArguments[,MOH_Class]] it looks like /var/samba/profiles/jonaskl/recording is in the spot for [,MOH_Class] -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf not working
Also try putting Asterisk in the audiopath by setting canreinvite=no in sip.conf Regards Ian On Sat, Jun 7, 2008 at 4:07 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 08:36, Sat 07 Jun 08, Russell Bryant wrote: On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote: i have this on my features.conf: [applicationmap] testfeature = *9,callee,Playback,tt-monkeys extensions.conf: [globals] DYNAMIC_FEATURES=testfeature trunk_1 = Zap/g1 trunk_2 = Zap/g2 what else i have to add in order to make this works? im using 2 xlite, Just a hunch ... if you're using xltite, it's likely that you're not pressing the digits fast enough to satisfy the default timeout. The default featuredigittimeout is 500 ms. Change this option in features.conf and increase it to 2000 ms and try again. another tip: Make sure you have the dtmfmode for the xlite sip stanza set correctly. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf not working
On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote: i have this on my features.conf: [applicationmap] testfeature = *9,callee,Playback,tt-monkeys extensions.conf: [globals] DYNAMIC_FEATURES=testfeature trunk_1 = Zap/g1 trunk_2 = Zap/g2 what else i have to add in order to make this works? im using 2 xlite, Just a hunch ... if you're using xltite, it's likely that you're not pressing the digits fast enough to satisfy the default timeout. The default featuredigittimeout is 500 ms. Change this option in features.conf and increase it to 2000 ms and try again. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf not working
On 08:36, Sat 07 Jun 08, Russell Bryant wrote: On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote: i have this on my features.conf: [applicationmap] testfeature = *9,callee,Playback,tt-monkeys extensions.conf: [globals] DYNAMIC_FEATURES=testfeature trunk_1 = Zap/g1 trunk_2 = Zap/g2 what else i have to add in order to make this works? im using 2 xlite, Just a hunch ... if you're using xltite, it's likely that you're not pressing the digits fast enough to satisfy the default timeout. The default featuredigittimeout is 500 ms. Change this option in features.conf and increase it to 2000 ms and try again. another tip: Make sure you have the dtmfmode for the xlite sip stanza set correctly. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf not working
Hi, im a new user to asterisk. i have configured one box using asterisknow. now i want to enable *9 (or some code) to play for example tt-monkeys. i read a lot in voip-info but cant do it: i have this on my features.conf: [applicationmap] testfeature = *9,callee,Playback,tt-monkeys extensions.conf: [globals] DYNAMIC_FEATURES=testfeature trunk_1 = Zap/g1 trunk_2 = Zap/g2 what else i have to add in order to make this works? im using 2 xlite, please help me ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf Problem with DTMF_sequence
Tilghman Lesher wrote: On Tuesday 22 April 2008 05:22, Sergey Shumeyko wrote: I have following problem with my Asterisk installation (version 1.6.0. beta 7.1). I want to assign start record conversation to #7 and stop record conversation to #8, but it isn't working (on previous Asterisk 1.2.17 it was working fine). When I assign those functions to 7/8 (without #) correspondingly it also works fine, but it works only from caller side. I would appreciate very much if somebody can take a look at my configuration below and give me comments what I am doing wrong. http://bugs.digium.com/view.php?id=12299 No, I think this issue have other root. The issue I've reported have deal only with pickup number. Sergey, it is interesting for me, I'll try to look at you configuration till weekend. -- Best regards, Igor A. Goncharovsky ___ ICQ: 648337 blog: http://igorg.ru mailto: [EMAIL PROTECTED] ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf Problem with DTMF_sequence
Hello,
I have following problem with my Asterisk installation (version 1.6.0. beta
7.1). I want to assign start record conversation to #7 and stop record
conversation to #8, but it isn't working (on previous Asterisk 1.2.17 it was
working fine). When I assign those functions to 7/8 (without #)
correspondingly it also works fine, but it works only from caller side. I
would appreciate very much if somebody can take a look at my configuration
below and give me comments what I am doing wrong.
My configuration:
features.conf:
[featuremap]
blindxfer = 111222333 ; Blind transfer (default is #)
disconnect = 444555666 ; Disconnect (default is *)
;automon = *1 ; One Touch Record a.k.a. Touch Monitor
;atxfer = *2 ; Attended transfer
;parkcall = #72; Park call (one step parking)
;automixmon = *3; One Touch Record a.k.a. Touch MixMonitor
[applicationmap]
testfeature = #9,peer/both,Playback,beep
record_start = #7,self/both,Macro,RECORD_START ; doesn't work, peer or
self doesn't make difference
record_stop = #8,self/both,Macro,RECORD_STOP ; doesn't work, peer or
self doesn't make difference
;record_start = 7,self/both,Macro,RECORD_START; works fine only from
caller side
;record_stop = 8,self/both,Macro,RECORD_STOP ; works fine only from
caller side
extension.conf:
[general]
autofallthrough=yes
[macro-RECORD_START]
exten = s,1,Playback(beep)
exten = s,2,AGI(${AGI_SERVER}${RECORD_AGI}?MODE=start)
[macro-RECORD_STOP]
exten = s,1,AGI(${AGI_SERVER}${RECORD_AGI}?MODE=stop)
--
Regards, Shuma
___
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Re: [asterisk-users] features.conf Problem with DTMF_sequence
On Tuesday 22 April 2008 05:22, Sergey Shumeyko wrote: I have following problem with my Asterisk installation (version 1.6.0. beta 7.1). I want to assign start record conversation to #7 and stop record conversation to #8, but it isn't working (on previous Asterisk 1.2.17 it was working fine). When I assign those functions to 7/8 (without #) correspondingly it also works fine, but it works only from caller side. I would appreciate very much if somebody can take a look at my configuration below and give me comments what I am doing wrong. http://bugs.digium.com/view.php?id=12299 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf / DTMF / automon hell
I've ran into an issue (1.4.17) where anything in features.conf is being totally ignored after the * or # for a particular feature. Everything works just fine as long as I restrict the digit to only a * or #, but apps that require #1 or *1 simply never get recognized. No clue what's causing this, but you could try just setting one of the features to only a * or # and seeing what happens. On a related note, anyone know what could cause this behavior? On Jul 1, 2007 6:26 PM, Russell Bryant [EMAIL PROTECTED] wrote: Russell Bryant wrote: Here are some examples of setting up automon access when calling SIP/1234. Examples 2 and 3 use variable inheritance. I think I made these examples *way* more complicated than they needed to be. After going back and refreshing myself on configuring dynamic features in features.conf (in 1.4), I see that you can control who has access to the feature by the definition of the feature itself. You'll have to provide the line that you are using from features.conf, as well as the full extension where it isn't working. Then, I can probably tell you the problem. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Features.conf and passing DTMF to the other end
Hi folks, We have a problem here where users are calling a remote PBX and need to use # and * to navigate it. We were using the Tt options in Dial() so that we could later perhaps take advantage of this feature. Features.conf's sections are fully commented out, so I wasn't expecting the options on Dial() (the T and t) to have any effects. Anyway... I plan to turn off the dialing options, but I was wondering how other people handle this? Is there a way to say yeah pass # through if it doesn't match locally or better yet, when dialing a dtmf digit, to prefix it with something to force asterisk to ignore it and pass it along? Am I stuck with absolutely no features that depend on # or * if I want users to use those digits on a remote PBX? Thanks :) Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf / DTMF / automon hell
Danny Brown wrote: I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=#9 in my features.conf, I have Dial(,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is in fact understanding the dtmf info packets from the sip phone (the read app works). I have verified that the Monitor() app is present and works. I just can not get * to do anything from my features.conf file. I have also done include= featuremap. I have been all over the web, posted multiple times on the irc channels. Could someone please help here. This could be anything. You haven't told us which version of Asterisk you're running. It would help to see your dialplan and features.conf file. Are all your modules loading? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf / DTMF / automon hell
Danny Brown wrote: I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=#9 in my features.conf, I have Dial(,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is in fact understanding the dtmf info packets from the sip phone (the read app works). I have verified that the Monitor() app is present and works. You may not be setting the DYNAMIC_FEATURES variable on the channel that you think you are. From what I can gather, you are only setting it on the calling party's channel. That means that the called party will not have the ability to use this feature. The core issue is to understand which channel has the DYNAMIC_FEATURES variable set. Here are some examples of setting up automon access when calling SIP/1234. Examples 2 and 3 use variable inheritance. 1) Calling party only [default] exten = 1234,1,Set(DYNAMIC_FEATURES=automon) exten = 1234,n,Dial(SIP/1234) 2) Called party only [default] exten = 1234,1,Dial(Local/[EMAIL PROTECTED]) exten = 1234-with-automon,1,Set(__DYNAMIC_FEATURES=automon) exten = 1234-with-automon,n,Dial(SIP/1234) 3) Calling and Called party [default] exten = 1234,1,Set(__DYNAMIC_FEATURES=automon) exten = 1234,n,Dial(SIP/1234) -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf / DTMF / automon hell
Russell Bryant wrote: Here are some examples of setting up automon access when calling SIP/1234. Examples 2 and 3 use variable inheritance. I think I made these examples *way* more complicated than they needed to be. After going back and refreshing myself on configuring dynamic features in features.conf (in 1.4), I see that you can control who has access to the feature by the definition of the feature itself. You'll have to provide the line that you are using from features.conf, as well as the full extension where it isn't working. Then, I can probably tell you the problem. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf / DTMF / automon hell
I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=#9 in my features.conf, I have Dial(,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is in fact understanding the dtmf info packets from the sip phone (the read app works). I have verified that the Monitor() app is present and works. I just can not get * to do anything from my features.conf file. I have also done include= featuremap. I have been all over the web, posted multiple times on the irc channels. Could someone please help here. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf and blind xfer
I was wanting to automate entirely a blind transfer. We are not yet using a powerdialler, so when we hit an answermachine we have to manually leave a message. In order to make this a little quicker, I want to leave a standard message on the answermachine. attempt #1. Use the blind transfer feature. set blind transfer to be **. extension 22 is exten 22 = Goto (leavemessage,answermachine,1) during the call, agent presses **. Asterisk says transfer. Agent enters 22#. Call is transferred to extension 22. A Standard Message is left. cool. Agents say We always have to xfer to extension 22. Can't you automate that? attempt #2. Use the features application map doxfer = **,caller,goto,leavemessage|answermachine|1 during the call, agent presses **. CLI says Feature Found: doxfer exten: answermachine -- Goto (leavemessage,answermachine,1) but nothing happens. I've tried it with **,callee as well. Anyone been able to automate this type of thing ? Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf problems
Hi all, I am having a couple of problems with features.conf I was hoping to get some help with. #1. If an outside caller is parked, when retrieved, that caller will now have the ability to transfer. This only happens when they are put in call parking and then retrieved. #2. I cannot get any other keys to register for features. For instance, I tried assigned blindxfer = *1, but both my grandstream and soft phones only react to # (pound) key. Also, it's just the pound key. As soon as it is pressed, the transferring message. == features.conf = transferdigittimeout = 2 courtesytone = beep xfersound = beep xferfailsound =beeperr asdipark = yes pickupexten = *8 parkingtime = 30 parkingpos = 701-720 context = parkedcalls parkext = *70 [featuremap] blindxfer =*1 atxfer =*2 Any help would be appreciated. Thank you, -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording2006/5/10, Dave Morrow [EMAIL PROTECTED] : I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success.Is there a trick to this?May be a problem with the way you are sending the dialtones. Try sending as data.--Alejandro Vargas___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf *1 Call Recording
Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording2006/5/10, Dave Morrow [EMAIL PROTECTED] : I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success.Is there a trick to this?May be a problem with the way you are sending the dialtones. Try sending as data.--Alejandro Vargas___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
RE: [Asterisk-Users] features.conf *1 Call Recording
All I see when I press *1 is -- Attempting native bridge of SIP/8001-252e and SIP/3020-5171 I still cannot make this work. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users
RE: [Asterisk-Users] features.conf *1 Call Recording
It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording2006
Re: [Asterisk-Users] features.conf *1 Call Recording
hi Dave i get the following log on *CLI -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4 -- Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1147452537-200-204|m -- Playing 'beep' (language 'en') -- User hit '*1' to stop recording call. -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4what are you using as SIP client ( imean softphone/ analog phone + ATA / IPphone ) ? if you are using a softphone and that doesnot have a dtmf signaling then asterisk will not be able to recognize that you are pressing.--Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential
RE: [Asterisk-Users] features.conf *1 Call Recording
I have one Sipura SPA-841 which is configured to use dtmfmode=info and one Cisco 7905 which is using the default signalling (I believe this is rfc2833) I have also set relaxdtmf=yes in sip.conf I've tried pressing *1 on both phones (they are both on my desk) and both behave the same. ;; Sample Parking configuration; [general]parkext = 700 ; What ext. to dial to parkparkpos = 701-720 ; What extensions to park calls oncontext = parkedcalls ; Which context parked calls are in;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds);transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call;courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call;xfersound = beep ; to indicate an attended transfer is complete;xferfailsound = beeperr ; to indicate a failed transfer;adsipark = yes ; if you want ADSI parking announcements;findslot = next ; Continue to the 'next' parking space. Defaults to 'first' available;pickupexten = *8 ; Configure the pickup extension. Default is *8featuredigittimeout = 2000 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap]blindxfer = #1 ; Blind transferdisconnect = *0 ; Disconnectautomon = *1 ; One Touch Recordatxfer = *2 ; Attended transfer [applicationmap];testfeature = #9,callee,Playback,tt-monkeys ;Play tt-monkes to ;callee if #9 was pressed ~~~ David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 12:55 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording hi Dave i get the following log on *CLI -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4 -- Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1147452537-200-204|m -- Playing 'beep' (language 'en') -- User hit '*1' to stop recording call. -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4what are you using as SIP client ( imean softphone/ analog phone + ATA / IPphone ) ? if you are using a softphone and that doesnot have a dtmf signaling then asterisk will not be able to recognize that you are pressing.--Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non
RE: [Asterisk-Users] features.conf *1 Call Recording
I found the issue. It was my Dial command! In my dialplan I had Dial(SIP/100|20|Ttr,,wW) as this was something I gleaned from a sample config for call forwarding. I removed the |20|Ttr andnow the call recording works! Anyone know what the |20|Ttr did anyhow? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: Dave Morrow Sent: Friday, May 12, 2006 10:41 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] features.conf *1 Call Recording It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo
Re[2]: [Asterisk-Users] features.conf *1 Call Recording
CLI* show application dial -Original Message- From: Dave Morrow [EMAIL PROTECTED] To: Dave Morrow [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Fri, 12 May 2006 14:12:21 -0400 Delivered: Fri, 12 May 2006 15:16:15 Subject:[Asterisk-Users] features.conf *1 Call Recording I found the issue. It was my Dial command! In my dialplan I had Dial(SIP/100|20|Ttr,,wW) as this was something I gleaned from a sample config for call forwarding. I removed the |20|Ttr and now the call recording works! Anyone know what the |20|Ttr did anyhow? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: Dave Morrow Sent: Friday, May 12, 2006 10:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] features.conf *1 Call Recording It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Friday, May 12, 2006 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy Bandi Sent: Friday, May 12, 2006 3:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap] automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response. How would I change the DTMF transfer mode? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fabio Sent: Thursday, May 11, 2006 6:55 PM To: Asterisk Users Mailing List - Non-Commercial
Re[2]: [Asterisk-Users] features.conf *1 Call Recording
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Re[2]: [Asterisk-Users] features.conf *1 Call Recording
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Re[2]: [Asterisk-Users] features.conf *1 Call Recording
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Re[2]: [Asterisk-Users] features.conf *1 Call Recording
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Re[2]: [Asterisk-Users] features.conf *1 Call Recording
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Re: [Asterisk-Users] features.conf *1 Call Recording
Can't this guy read? [EMAIL PROTECTED] wrote: Please change the email address of [EMAIL PROTECTED] to [EMAIL PROTECTED] Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf *1 Call Recording
John Novack wrote: Can't this guy read? I'll bet he runs Micro$oftware, and has fallen prey to one of its thousands of exploits-du-jour. Set a filter and move on. . . B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf *1 Call Recording
if you ar using SIP clients, try changing DTMF transfer mode. For test use sip debug on your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are using inband transfer mode (DTMF codes are transferred like sounds) you don't see the codes. Also, try adjusting featuredigittimeout in features.conf: [general] featuredigittimeout = 2000 ; 2 seconds because the default 500ms is a very short time. Fabio Olaechea 3Tech SRL Calle 48 Nro 632, Of. 67. La Plata, CP B1900AMZ Buenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301 Fax. +54 221 445 0245 www.trestech.com.ar -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Dave Morrow Enviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] features.conf *1 Call Recording OK. You lost me. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] features.conf *1 Call Recording 2006/5/10, Dave Morrow [EMAIL PROTECTED]: I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? May be a problem with the way you are sending the dialtones. Try sending as data. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf *1 Call Recording
Thanks for the response. How would I change the DTMF transfer mode? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fabio Sent: Thursday, May 11, 2006 6:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recording if you ar using SIP clients, try changing DTMF transfer mode. For test use sip debug on your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are using inband transfer mode (DTMF codes are transferred like sounds) you don't see the codes. Also, try adjusting featuredigittimeout in features.conf: [general] featuredigittimeout = 2000 ; 2 seconds because the default 500ms is a very short time. Fabio Olaechea 3Tech SRL Calle 48 Nro 632, Of. 67. La Plata, CP B1900AMZ Buenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301 Fax. +54 221 445 0245 www.trestech.com.ar -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Dave Morrow Enviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] features.conf *1 Call Recording OK. You lost me. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] features.conf *1 Call Recording 2006/5/10, Dave Morrow [EMAIL PROTECTED]: I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? May be a problem with the way you are sending the dialtones. Try sending as data. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] features.conf *1 Call Recording
Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=automon [default] exten=123,2,Dial(SIP/3000,,wW);wWallowone-touchrecording During the call, I press *1 but it records nothing. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf *1 Call Recording
2006/5/10, Dave Morrow [EMAIL PROTECTED]: I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? May be a problem with the way you are sending the dialtones. Try sending as data. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf *1 Call Recording
OK. You lost me. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] features.conf *1 Call Recording 2006/5/10, Dave Morrow [EMAIL PROTECTED]: I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? May be a problem with the way you are sending the dialtones. Try sending as data. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] features.conf
[featuremap] blindxfer = #1; Blind transfer ;disconnect = *0; Disconnect automon = *1; One Touch Record atxfer = *2; Attended transfer extensions.conf of all phones I tried have the dial options: tTwWr I try to call from one phone to the other and try to transfer from callee and called phone with #1601# to transfer a call to 601, but I just hear at both phones the keys I press, but nothing happens. Same with *2601# What do I miss here? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
i used to have this problem, with me, it appeared that i had to press the feature keys very quickly. my solution was to set featuredigittimeout higher than the default 500. also, when i use IAX phones, i had to set dtmf to ouband audio for asterisk to recognize the keys pressed.On 4/28/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:[featuremap]blindxfer = #1; Blind transfer ;disconnect = *0; Disconnectautomon = *1; One Touch Recordatxfer = *2; Attended transferextensions.conf of all phones I tried have the dial options: tTwWr I try to call from one phone to the other and try to transfer fromcallee and called phone with #1601# to transfer a call to 601, but Ijust hear at both phones the keys I press, but nothing happens. Same with *2601#What do I miss here?byeRonald Wiplinger___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
In extensions.conf file, in that context, you must have: include = featuremap Thats lets you transfer calls. Regards. -- José Luis Gómez Qualis Information Technology Av. Rivadavia 2553, PB Of. 43 EP +54-342-4565684 int 102 www.qualis.com.ar Soporte 0810-8880022 Santa Fe - Argentina El vie, 28-04-2006 a las 22:51 +0800, Ronald Wiplinger escribió: [featuremap] blindxfer = #1; Blind transfer ;disconnect = *0; Disconnect automon = *1; One Touch Record atxfer = *2; Attended transfer extensions.conf of all phones I tried have the dial options: tTwWr I try to call from one phone to the other and try to transfer from callee and called phone with #1601# to transfer a call to 601, but I just hear at both phones the keys I press, but nothing happens. Same with *2601# What do I miss here? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] features.conf and CVS
This is my features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements [featuremap] blindxfer= ## automon = *1 atxfer = *2 I out in dial the options wWtT and I use the CVS CVS-v1-0-08/16/05-08:47:44 If I press the # on my phone asterisk wants to trasfer the call and if I press *1 it hung up. any idea? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Features.conf Set Language
I use features.conf in order to park call, but I would like use french speaker. how set langage in features.conf? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Features.conf for secretary function
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer = *0
blindxfer = #0
I completly restart asterik, and not just make a RELOAD. But during a
call, when I press # it runs a blind transfer and if I press * I am
disconnected.
I am using the CVS version of * get as explain here
http://www.voip-info.org/tiki-index.php?page=Asterisk+Download
Please do you have any advise ?
Alexis Fécourt.
For info:
My sip.conf:
[general]
context=from-sip
port=5060
bindaddr=0.0.0.0
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
[100]
type=friend
context=from-sip
host=dynamic
username=100
[106]
type=friend
host=dynamic
context=from-sip
username=106
Extension.conf:
[general]
[globals]
include = from-sip
[from-sip]
exten = _10X,1,Answer
exten = _10X,2,Dial(SIP/${EXTEN},20,htT)
exten = _10X,3,Hangup
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Re: [Asterisk-Users] Features.conf for secretary function
On 13:57, Tue 14 Jun 05, Alexis FECOURT wrote:
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer = *0
blindxfer = #0
snip/snip
exten = _10X,2,Dial(SIP/${EXTEN},20,htT)
exten = _10X,3,Hangup
Hi,
The problem is in that h param.
If you do a 'show application dial' on the asterisk CLI you
will read this:
'h' -- allow callee to hang up by hitting *.
Remove it, and you'll be fine
Michiel van Baak
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Re: [Asterisk-Users] Features.conf for secretary function
On 13:57, Tue 14 Jun 05, Alexis FECOURT wrote:
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer = *0
blindxfer = #0
snip/snip
exten = _10X,2,Dial(SIP/${EXTEN},20,htT)
exten = _10X,3,Hangup
Hi,
The problem is in that h param.
If you do a 'show application dial' on the asterisk CLI you
will read this:
'h' -- allow callee to hang up by hitting *.
Remove it, and you'll be fine
Michiel van Baak
Hi,
That is not the problem. I have to put tT to make a transfer, but the pad
composition to dial to make it should be defined in feature.conf
I made some test by putting hH in the Dial command options and
disconnect = *0 in the feature.conf, but with that configuration I just need to press * and not *0 to be disconnected.
My problem comes from the fact that my featuremap is not used. I read in google
groups:
This only works on CVS version of *, not the stable 1.0
--
Michiel van Baak
So I followed the CVS way of getting * from the wiki on voip-info.org...
Nevertheless, no result...
Alexis Fécourt.
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Re: [Asterisk-Users] Features.conf - atxfer
Reading through the code, I don't see a way of exiting the transfer and
regaining the call with the customer, unless the third party hangs up or
maybe doesn't answer and the dialplan doesn't do anything else with the
call (send the call into voicemail).
I suggest you request this feature (http://bugs.digium.com), but as an
interim solution you can create a dialplan for internal extensions that
does not send the call to voicemail if unanswered, and only dials the
third party for a limited amount of time (20 seconds?).
You could preface these special extensions with a sequence, such as 9,
or 777 or whatever. Assuming your extensions are 1xx:
exten = _7771XX,1,Dial(SIP/${EXTEN:3},20)
exten = _7771XX,2,Hangup
-mike
Mark Johnson wrote:
I am trying out the new atxfer feature from CVS-HEAD. I set atxfer
equal to *7 and it seems to work OK. I am having a problem getting it
to work the way a receptionist would want. If an extension calls me, I
hit *7 and I hear the voice say transfer. I dial another extension.
If the newly dialed extension goes to voicemail, I can't figure out how
to get the original call back to tell them the person they are trying to
reach is unavailable. Anything I try bridges the call and the caller go
into like the 2nd half of the voicemail greeting. Is there some trick
to this?
Thanks!
Mark
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Re: [Asterisk-Users] Features.conf - atxfer
You can use super-valet-parking
On Tue, 2005-06-07 at 06:18 -0500, Mike Holloway wrote:
Reading through the code, I don't see a way of exiting the transfer and
regaining the call with the customer, unless the third party hangs up or
maybe doesn't answer and the dialplan doesn't do anything else with the
call (send the call into voicemail).
I suggest you request this feature (http://bugs.digium.com), but as an
interim solution you can create a dialplan for internal extensions that
does not send the call to voicemail if unanswered, and only dials the
third party for a limited amount of time (20 seconds?).
You could preface these special extensions with a sequence, such as 9,
or 777 or whatever. Assuming your extensions are 1xx:
exten = _7771XX,1,Dial(SIP/${EXTEN:3},20)
exten = _7771XX,2,Hangup
-mike
Mark Johnson wrote:
I am trying out the new atxfer feature from CVS-HEAD. I set atxfer
equal to *7 and it seems to work OK. I am having a problem getting it
to work the way a receptionist would want. If an extension calls me, I
hit *7 and I hear the voice say transfer. I dial another extension.
If the newly dialed extension goes to voicemail, I can't figure out how
to get the original call back to tell them the person they are trying to
reach is unavailable. Anything I try bridges the call and the caller go
into like the 2nd half of the voicemail greeting. Is there some trick
to this?
Thanks!
Mark
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RE: [Asterisk-Users] Features.conf - atxfer
On superviced you cancel on a zap channel you can cancel the transfer by
a hook flash this will send you back to the original caller.
On sip phones you hit the cancel button or if you have line buttons you just
pick the original callers line.
--
From: Mike Holloway[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, June 07, 2005 7:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] Features.conf - atxfer
Reading through the code, I don't see a way of exiting the transfer and
regaining the call with the customer, unless the third party hangs up or
maybe doesn't answer and the dialplan doesn't do anything else with the
call (send the call into voicemail).
I suggest you request this feature (http://bugs.digium.com), but as an
interim solution you can create a dialplan for internal extensions that
does not send the call to voicemail if unanswered, and only dials the
third party for a limited amount of time (20 seconds?).
You could preface these special extensions with a sequence, such as 9,
or 777 or whatever. Assuming your extensions are 1xx:
exten = _7771XX,1,Dial(SIP/${EXTEN:3},20)
exten = _7771XX,2,Hangup
-mike
Mark Johnson wrote:
I am trying out the new atxfer feature from CVS-HEAD. I set atxfer
equal to *7 and it seems to work OK. I am having a problem getting it
to work the way a receptionist would want. If an extension calls me, I
hit *7 and I hear the voice say transfer. I dial another extension.
If the newly dialed extension goes to voicemail, I can't figure out how
to get the original call back to tell them the person they are trying to
reach is unavailable. Anything I try bridges the call and the caller go
into like the 2nd half of the voicemail greeting. Is there some trick
to this?
Thanks!
Mark
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[Asterisk-Users] Features.conf - atxfer
I am trying out the new atxfer feature from CVS-HEAD. I set atxfer equal to *7 and it seems to work OK. I am having a problem getting it to work the way a receptionist would want. If an extension calls me, I hit *7 and I hear the voice say transfer. I dial another extension. If the newly dialed extension goes to voicemail, I can't figure out how to get the original call back to tell them the person they are trying to reach is unavailable. Anything I try bridges the call and the caller go into like the 2nd half of the voicemail greeting. Is there some trick to this? Thanks! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] features.conf
Hello list, i configured correctly the codes in features.conf, loaded successfully res_features, but while in a call (any type of call including zaptel to zaptel, zaptel to sip, sip to sip) both sides hear DTMFs and nothing happens... i'm i missing something? Thanks, Calin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] features.conf
Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf
-Original Message- From: AJ Grinnell [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 10:28 AM To: Asterisk Subject: [Asterisk-Users] features.conf Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it? I think that it should be in configs/features.conf.sample unless you have run make samples in which case this file is copied to /etc/asterisk. Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf
Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Jackson Sent: Tuesday, August 03, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] features.conf -Original Message- From: AJ Grinnell [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 10:28 AM To: Asterisk Subject: [Asterisk-Users] features.conf Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it? I think that it should be in configs/features.conf.sample unless you have run make samples in which case this file is copied to /etc/asterisk. Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned by Arialink for dangerous content and is believed to be clean. For more information please email [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
- Original Message - From: AJ Grinnell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 8:02 AM Subject: RE: [Asterisk-Users] features.conf Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's been in there for over a week now, I just checked out a new copy and it's in there... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf
Chris Shaw [EMAIL PROTECTED] wrote: Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's been in there for over a week now, I just checked out a new copy and it's in there... Or simply rename musiconhold.conf as features.com and restart Asterisk. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
Kevin Walsh wrote: Chris Shaw [EMAIL PROTECTED] wrote: Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's been in there for over a week now, I just checked out a new copy and it's in there... Or simply rename musiconhold.conf as features.com and restart Asterisk. no.. WRONG. rename parking.conf, as parking.conf is what features.conf is derived from. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf
Josh Roberson [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Or simply rename musiconhold.conf as features.com and restart Asterisk. no.. WRONG. rename parking.conf, as parking.conf is what features.conf Oops. I knew it was one of them. At least I didn't say sip.conf :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
Kevin Walsh wrote: Josh Roberson [EMAIL PROTECTED] wrote: no.. WRONG. rename parking.conf, as parking.conf is what features.conf Oops. I knew it was one of them. At least I didn't say sip.conf :-) True that. This is another reminder that everyone needs to make sure that when they update, they check all of the files in the configs/ path in the src tree to see what's changed. Also, if you're confused about why something that's supposed to be in cvs isn't, a good method would be to make clean; make update; make install. If that still doesn't cure it, blow away the source tree and start with a new checkout. Just a friendly reminder to the list. twisted ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
