Re: [asterisk-users] Page() bumps user out of a call
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.
The problem comes when a user is on the line, and someone else uses the
intercom function to page all extensions, the call in progress gets
disconnected. I'm wondering if there is a way to either:
1. dynamically figure out the subset of extensions that are not in a call,
or
2. use some other function that will not bump a call?
Has anyone else run into this?
Thanks
Cassius
Here is my intercom context:
[intercom]
exten = s,1,Answer
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
exten = t,1,NoOp(timeout)
exten = t,n,Playback(sorry-youre-having-problemsgoodbye)
exten = t,n,Hangup()
exten = *,1,SIPAddHeader(Call-Info:sip:${SERVER_IP}\;answer-after=0)
exten = *,n,Page(${ALL-PAGE-EXTS}) ; add all your devices here
exten = _,1,SIPAddHeader(Call-Info:
sip:${SERVER_IP}\;answer-after=0) ; 4 digit extensions
exten = _,n,Dial(SIP/${EXTEN})
Hey Cassius!
Nice to hear from you, what crazy country are you deploying Asterisk
in now? You might want to checkout the DEVICE_STATE() function. Should
be able to build your ALL-PAGE-EXTS while leaving out the busy
extensions. Probably not the best solution, but the first one I thought of.
--
Russ Meyerriecks
Digium | Linux Kernel Developer
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Re: [asterisk-users] Page() bumps user out of a call
On 6/14/11 4:25 PM, Russ Meyerriecks wrote:
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.
The problem comes when a user is on the line, and someone else uses the
intercom function to page all extensions, the call in progress gets
disconnected. I'm wondering if there is a way to either:
1. dynamically figure out the subset of extensions that are not in a
call,
or
2. use some other function that will not bump a call?
Has anyone else run into this?
Thanks
Cassius
Here is my intercom context:
[intercom]
exten = s,1,Answer
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
exten = t,1,NoOp(timeout)
exten = t,n,Playback(sorry-youre-having-problemsgoodbye)
exten = t,n,Hangup()
exten = *,1,SIPAddHeader(Call-Info:sip:${SERVER_IP}\;answer-after=0)
exten = *,n,Page(${ALL-PAGE-EXTS}) ; add all your devices here
exten = _,1,SIPAddHeader(Call-Info:
sip:${SERVER_IP}\;answer-after=0) ; 4 digit extensions
exten = _,n,Dial(SIP/${EXTEN})
Hey Cassius!
Nice to hear from you, what crazy country are you deploying Asterisk in
now? You might want to checkout the DEVICE_STATE() function. Should be
able to build your ALL-PAGE-EXTS while leaving out the busy extensions.
Probably not the best solution, but the first one I thought of.
This may be a better solution, actually. Checkout example 1. It sets up
a macro to handle the check for each extension.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page
--
Russ Meyerriecks
Digium | Linux Kernel Developer
--
_
-- 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] Page() bumps user out of a call
On 6/14/11 4:37 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote:
On 6/14/11 4:25 PM, Russ Meyerriecks wrote:
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for
under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys
SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.
The problem comes when a user is on the line, and someone else uses the
intercom function to page all extensions, the call in progress gets
disconnected. I'm wondering if there is a way to either:
1. dynamically figure out the subset of extensions that are not in a
call,
or
2. use some other function that will not bump a call?
Has anyone else run into this?
Thanks
Cassius
Here is my intercom context:
[intercom]
exten = s,1,Answer
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
exten = t,1,NoOp(timeout)
exten = t,n,Playback(sorry-youre-having-problemsgoodbye)
exten = t,n,Hangup()
exten = *,1,SIPAddHeader(Call-Info:sip:${SERVER_IP}\;answer-after=0)
exten = *,n,Page(${ALL-PAGE-EXTS}) ; add all your devices here
exten = _,1,SIPAddHeader(Call-Info:
sip:${SERVER_IP}\;answer-after=0) ; 4 digit extensions
exten = _,n,Dial(SIP/${EXTEN})
Hey Cassius!
Nice to hear from you, what crazy country are you deploying Asterisk in
now? You might want to checkout the DEVICE_STATE() function. Should be
able to build your ALL-PAGE-EXTS while leaving out the busy extensions.
Probably not the best solution, but the first one I thought of.
This may be a better solution, actually. Checkout example 1. It sets up
a macro to handle the check for each extension.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page
Hi Russ,
Thanks for this. I was thinking of the DEVICE_STATE() also, just hoping
someone
Had a snippet that might make it easier. I've implemented something very
much like
The example 1 code on the referenced page. (The above code was actually
from example 2!).
I will have the crew in Vienna check it out when they get into the office.
Cassius
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