Re: [asterisk-users] Maximum Wait Time queue option
Thanks Warren for your help On Mon, Aug 30, 2010 at 9:21 PM, Warren Selby wrote: > On Mon, Aug 30, 2010 at 10:31 AM, Tino wrote: > >> Hello, >> >> Is there any option to set the maximum number of seconds a caller can wait >> in a queue before being pulled out ? >> >> Thanks >> >> > > In the Queue() command itself there is a timeout parameter. From your > asterisk box, try running: > > asterisk -rx "core show application Queue" > > and pay attention to the timeout parameter. > > -- > 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 > -- _ -- 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] Maximum Wait Time queue option
Hello, Is there any option to set the maximum number of seconds a caller can wait in a queue before being pulled out ? 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
[asterisk-users] queue agent and blind transfer
Hello, When an agent does a blind transfer the call hangups for him but shows as "In use" in queue in my CRM (used for auto dialing). As a result the agent have to wait until the transfered call completes. Is there any way to change this behaviour ? -- _ -- 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] music on hold in blind transfer
Hello, Is it possible to avoid playing music on hold during a blind transfer ? 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
[asterisk-users] sms - your suggestions
Hello, I planning to use a web interface to send sms through Asterisk server. I am planning to use php code which will interact with Asterisk Manager Interface(AMI) and use Sms() application to send sms. I am not sure whether it is the write way to do this. Anybody have any suggestions or tips, please let me know . -- _ -- 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] AMD message
Yes, we need to record the message On Wed, Aug 25, 2010 at 12:35 PM, Matt Riddell wrote: > On 20/08/10 1:52 AM, Tino wrote: > > Hello, > > > > Is there a way to capture the answering machine message when the dialer > > detects the answering machine. > > Record? > > -- > Cheers, > > Matt Riddell > ___ > > http://www.venturevoip.com/news.php (Daily Asterisk News) > http://www.venturevoip.com/exchange.php (Full ITSP Solution) > http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) > > -- > _ > -- 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] setting variable for a DID number
But when i call my DID number following dialplans are being executed. What
i need is to set a variable with one value for one DID number and set the
same variable with another value for another DID number. Also any contexts
should be able to use this variable.
-
NoOp("SIP/5070-5407", "Received incoming SIP connection from unknown
peer to ") in new stack
-- Executing [@from-sip-external:2] Set("SIP/5070-5407",
"DID=") in new stack
-- Executing [@from-sip-external:3]
Goto("SIP/5070-5407", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [...@from-sip-external:1] GotoIf("SIP/5070-5407",
"1?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,2)
-- Executing [...@from-sip-external:2] GotoIf("SIP/5070-5407",
"0?setlanguage:from-trunk||1") in new stack
-- Goto (from-trunk,,1)
-- Executing [@from-trunk:1] Set("SIP/5070-5407",
"__FROM_DID=") in new stack
-- Executing [@from-trunk:2] Gosub("SIP/5070-5407",
"app-blacklist-check|s|1") in new stack
-- Executing [...@app-blacklist-check:1]
LookupBlacklist("SIP/5070-5407", "") in new stack
-- Executing [...@app-blacklist-check:2] GotoIf("SIP/5070-5407",
"0?blacklisted") in new stack
-- Executing [...@app-blacklist-check:3] Set("SIP/5070-5407",
"CALLED_BLACKLIST=1") in new stack
-- Executing [...@app-blacklist-check:4] Return("SIP/5070-5407", "")
in new stack
-- Executing [@from-trunk:3] ExecIf("SIP/5070-5407", "0
|Set|CALLERID(name)=Anonymous") in new stack
-- Executing [@from-trunk:4] Set("SIP/5070-5407",
"__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [@from-trunk:5]
SetCallerPres("SIP/5070-5407", "allowed_not_screened")
P.S : used in place of actual DID number
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[asterisk-users] setting variable for a DID number
Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -- _ -- 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] AMD message
Hello, Is there a way to capture the answering machine message when the dialer detects the answering machine. 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] sending sms from Asterisk server
Hello Johann, Thanks for your advice in this matter. But i am not sure how to pass the numbers to be sent sms in the dialplan. On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn wrote: > On 08/17/2010 09:00 AM, Tino wrote: > >> Hello, >> >> I would like to send sms to some external phone numbers from my asterisk >> server. Is it possible to send sms via softphones like X-Lite ? . Any tips >> regarding this will be helpful >> >> thanks >> >> >> This is easy to do by using email to SMS gateways. A list of them is on > wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the > Asterisk side, you have an extension that sends the email. I personally use > an AGI script for this part, but you could use a System() call as well. > > > --johann > -- _ -- 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] sending sms from Asterisk server
Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful 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
[asterisk-users] Asterisk Hardwares
Hello, Can antbody recommend devices that can be used along with my Asterisk server Paging Amplifier SIP enabled Paging Gateway VOIP SIP loudspeaker Also , please recommend video phone sets that suppot paging, intercom (autoanswer) 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
[asterisk-users] channel variables in AGI
Hello, How to take the values of channel variables like 'agi_uniqueid' and 'agi_callerid' in agi script. For example #!/bin/bash -x T="$agi_uniqueid" I want to save value of 'agi_uniqueid' channel variable into a variable called 'T' in my script -- _ -- 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] asterisk on Vmware
Thanks Gareth for your quick reply. It is the lateset version and i think i need access to Dahdi interface. Is there any disadvantages other than this. On Wed, Aug 11, 2010 at 2:11 PM, Gareth Blades wrote: > Tino wrote: > > Hello, > > > > Is it possible to install Asterisk on Vmware(centos) from source. Is > > there any difference or disadvantage for this compared to asterisk > > running on physical machine. > > > > What version of vmware? > > Generally it works but it could be a problem if you require access to > dahdi interface cards. > > -- > _ > -- 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] asterisk on Vmware
Hello, Is it possible to install Asterisk on Vmware(centos) from source. Is there any difference or disadvantage for this compared to asterisk running on physical machine. -- _ -- 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] 'System' application in asterisk
Hello Julian,
I am using Asterisk 1.4.33.1(AsteriskNOW iso) and curl function is not
available in this version.
On Tue, Aug 10, 2010 at 1:21 PM, Julian Lyndon-Smith wrote:
> You could always use the CURL function directly in the dialplan
>
> Julian
>
> On 10 August 2010 08:36, Tino wrote:
> >
> > Hi Steve, thanks for your interest in this matter.
> > I will explain my requirement here.
> >
> > In my asterisk server before an agent doing manual dial is allowed a
> call,
> > asterisk will make an http request (to my crm, do not worry about this
> part
> > ) and get back an OK or something else. … if it receives OK, it allows
> the
> > call, otherwise we just play an "unauthorized call" recording to the
> agent.
> > We make the http request using a "wget | perl " command and we want to
> > capture the output of the wget | perl command.
> >
> >
> > On Tue, Aug 10, 2010 at 12:42 PM, Steve Edwards sedwards.com>
> > wrote:
> >>
> >> Un-top-posting...
> >>
> >> On Tue, 10 Aug 2010, Tino wrote:
> >>
> >>> >Is there any way to capture the output of the 'System' application in
> >>> > >asterisk dialplan and evaluate it.
> >>
> >>> On Mon, Aug 9, 2010 at 11:51 PM, Danny Nicholas
> >>> wrote:
> >>
> >>> I think this answer is no. system only returns ${SYSTEMSTATUS} as
> >>> SUCCESS or FAILURE to tell you that the command finished or died. You
> could
> >>> however do a bash AGI that would set a variable with the result of what
> you
> >>> would have sent to system
> >>
> >> On Tue, 10 Aug 2010, Tino wrote:
> >>
> >>> Sorry Dany, I am new to agi scripting. If you do not mind can you
> please
> >>> give me a sample script for this. That would be really helpful to me.
> >>
> >> Unless the output from your system command is trivial, you should parse
> it
> >> in the AGI and set channel variables as needed.
> >>
> >> If you provide a bit more detail, you may get a more specific answer.
> >> System() may not be the "best" approach.
> >>
> >> --
> >> Thanks in advance,
> >>
> -
> >> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
> PST
> >> Newline Fax:
> +1-760-731-3000
> >> --
> >> _
> >> -- 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
> >
>
> --
> _
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> http://www.asterisk.org/hello
>
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>
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Re: [asterisk-users] 'System' application in asterisk
Hi Steve, thanks for your interest in this matter.
I will explain my requirement here.
In my asterisk server before an agent doing manual dial is allowed a call,
asterisk will make an http request (to my crm, do not worry about this part
) and get back an OK or something else. … if it receives OK, it allows the
call, otherwise we just play an "unauthorized call" recording to the agent.
We make the http request using a "wget | perl " command and we want to
capture the output of the wget | perl command.
On Tue, Aug 10, 2010 at 12:42 PM, Steve Edwards
wrote:
> Un-top-posting...
>
>
> On Tue, 10 Aug 2010, Tino wrote:
>
> >Is there any way to capture the output of the 'System' application in
>> >asterisk dialplan and evaluate it.
>>
>
> On Mon, Aug 9, 2010 at 11:51 PM, Danny Nicholas
>> wrote:
>>
>
> I think this answer is no. system only returns ${SYSTEMSTATUS} as SUCCESS
>> or FAILURE to tell you that the command finished or died. You could however
>> do a bash AGI that would set a variable with the result of what you would
>> have sent to system
>>
>
> On Tue, 10 Aug 2010, Tino wrote:
>
> Sorry Dany, I am new to agi scripting. If you do not mind can you please
>> give me a sample script for this. That would be really helpful to me.
>>
>
> Unless the output from your system command is trivial, you should parse it
> in the AGI and set channel variables as needed.
>
> If you provide a bit more detail, you may get a more specific answer.
> System() may not be the "best" approach.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
> --
> _
> -- 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
>
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Re: [asterisk-users] 'System' application in asterisk
Sorry Dany, I am new to agi scripting. If you do not mind can you please
give me a sample script for this. That would be really helpful to me.
On Mon, Aug 9, 2010 at 11:51 PM, Danny Nicholas wrote:
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tino
> *Subject:* [asterisk-users] 'System' application in asterisk
>
> >Hello,
> >Is there any way to capture the output of the 'System' application in
> asterisk dialplan and evaluate it.
>
> >For example, i would like to get the output of following System
> application and use its value in next line
> >for decision making
>
> >exten => 5000,n,System(command)
>
> I think this answer is no. system only returns ${SYSTEMSTATUS} as
> SUCCESS or FAILURE to tell you that the command finished or died. You could
> however do a bash AGI that would set a variable with the result of what you
> would have sent to system
>
> Replace
>
> Exten => 5000,n,System(‘/bin/ls’)
>
> With
>
> Exten => 5000,n,AGI(bashsys.sh,”/bin/ls’)
>
> Exten => 5000,n.Gotoif(${RESULT}…
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] 'System' application in asterisk
Hello, Is there any way to capture the output of the 'System' application in asterisk dialplan and evaluate it. For example, i would like to get the output of following System application and use its value in next line for decision making exten => 5000,n,System(command) -- _ -- 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] AMD setup in Astersik
In my Asterisk server following things have been done to detect answering
machines before the answered call connects to the agents in queue.
In extension_additional.conf
==
[ext-queues]
include => ext-queues-custom
exten => 5000,20,Macro(user-callerid,); changed the priority to 20
...
==
In extension_custom.conf added following amd dialplan
===
[ext-queues-custom]
exten => 5000,1,Answer()
exten => 5000,n,AMD(2500|1500|300|5000|120|50|4|384)
exten => 5000,n,GotoIf($["${AMDSTATUS}" = "MACHINE"]?machine:human)
exten => 5000,n(machine),Verbose(3, We found an answring machine)
exten => 5000,n,Set(AMP=${CALLERID(num)})
exten => 5000,n,Set(date=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten => 5000,n,System(not showing the actual command)
exten => 5000,n,Goto(ext-queues,5000,20)
exten => 5000,n(human),Verbose(3, We've got a human on the line!)
exten => 5000,n,Goto(ext-queues,5000,20)
===
This setup is working fine but the problem is that when i reload freepbx,
extension_additional.conf will go to its original form
and the changes made will be lost. Is there any way to make the changes in
extension_additional.conf conf permanent . Or is there any alternative
method for this ?
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Re: [asterisk-users] Tweaking AMD in Asterisk
Thanks Danny, What should be the length of audio file ? On Wed, Aug 4, 2010 at 9:21 PM, Danny Nicholas wrote: > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tino > *Subject:* Re: [asterisk-users] Tweaking AMD in Asterisk > > > > >Hi Aurimas, > > >Thanks for your thoughts on this. Can you please let me know how playing > a silent audio file before AMD will help to tweak the parameter values. > > Just a WAG – playing the file gives AMD a few more seconds to properly do > it’s thing. > > -- > _ > -- 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] Tweaking AMD in Asterisk
Hi Aurimas, Thanks for your thoughts on this. Can you please let me know how playing a silent audio file before AMD will help to tweak the parameter values. On Wed, Aug 4, 2010 at 8:30 PM, Aurimas Skirgaila wrote: > Hi, > > the basic settings are pretty good ones. What I did to do improve the > performance and prevent the false positives, I started to recorded every > call, and analyzed every incorrect detection :) Fairly soon I came with > optimal set for my environment: > > initial_silence= 2500 > greeting = 1500 > after_greeting_silence = 300 > total_analysis_time= 5000 > min_word_length= 120 > between_words_silence = 50 > maximum_number_of_words= 4 ; it's usuall to pickup saying "Jon > Anderssen, hello" in here > silence_threshold = 384 > > by the way, for outgoing SIP calls you might want to do this Background > trick as it helped me a lot regarding AMD on SIP. > > exten => _X.,n,Background(blank_audio) > exten => _X.,n,AMD > > > On Wed, Aug 4, 2010 at 5:08 PM, Tino wrote: > >> Hello , >> >> I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My >> current values are >> >> AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx >> 25-30 % of all answering machines. >> >> Anybody have any suggestion to improve the accuracy of AMD. >> >> 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 >> > > > > -- > Mvh, > Aurimas Skirgaila > > -- > _ > -- 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] Tweaking AMD in Asterisk
Hello , I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My current values are AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx 25-30 % of all answering machines. Anybody have any suggestion to improve the accuracy of AMD. 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
[asterisk-users] can't write to queues_additional.conf
Hello, In my Asterisk server when i try to set the value for the queue option "Skip Busy Agents" in Freepbx GUI it is not being written into the backend file queues_additional.conf. As a result sometimes agents in queue gets calls when they are already busy with another call. So i set "ringinuse=no" option manually from backend. Is it bug ? Is there any fix for this?. I am providing the details of version of asterisk and freepbx. Asterisk : Asterisk 1.4.33.1 FreePBX version : 2.7.0.5 queue Module version : 2.5.4.8 -- _ -- 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] Asterisk and TV media server
Sorry, I am a newbie to this concept. Can you please briefly explain how it is possible to watch TV channels using a video phone by just dialing a number. Is there any website links that you can share with me on this subject ? . Thanks for your interest in this matter. On Mon, Aug 2, 2010 at 8:54 PM, Kyle Kienapfel wrote: > On Mon, Aug 2, 2010 at 5:37 AM, Tino wrote: > > Hello, > > > > I would like to know whether there is a way to associate a TV media > server with Asterisk. Is it possible to access TV Chanels in the Telephone > Sets. Anybody have any tips or documents related to this please let me know. > > > > 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 > > > > That idea could go two ways, dial a number and get audio, or dial a > number with a video phone and watch the channel. The video phone idea > sounds like it'd be neat to use. > > -- > _ > -- 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] Asterisk and TV media server
Hello, I would like to know whether there is a way to associate a TV media server with Asterisk. Is it possible to access TV Chanels in the Telephone Sets. Anybody have any tips or documents related to this please let me know. 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
