Re: [asterisk-users] asterisk and IVR
i have Create a h extension and all works without issue .thank you so
much for your help and support i really appreciate it.
2013/7/31 A J Stiles asterisk_l...@earthshod.co.uk
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
So it looks as though it's breaking out of the extension logic altogether,
if
the call gets answered. In that case, you'll have to do it the
old-fashioned
way: Create a h extension (which fires when a call is hung up) *in the
same context as your 534 extension* (you can have a h extension in each
context, if needs be), and do all your fancy end-of-call stuff there.
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,Goto(home,s,1)
exten = h,1,NoOp(Hangup received. Dial status is ${DIALSTATUS})
Note that if there are other extensions in the context, h will be called
when
they get hung up -- you might need some logic in there to deal with this
(or
cheat by just having one extension besides h in this context, and use a
fully-
specified Goto() to jump into it.)
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
On Thursday 01 August 2013, Salaheddine Elharit wrote: i have Create a h extension and all works without issue .thank you so much for your help and support i really appreciate it. Good -- glad you got it working. But in future, please remember to type your reply *after* the thing you are replying to, not before it. That way, the logical flow of the conversation can be maintained in the correct order, with answers appearing after questions. -- AJS Answers come *after* questions. -- _ -- 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 IVR
hello,
the CLI for whe the call is answered :
Accepting call from '0661xx' to '534' on channel 0/26, span 1
-- Executing [534@default:1] Dial(Zap/26-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e71378 is ringing
-- SIP/228-09e71378 answered Zap/26-1
== Spawn extension (default, 534, 1) exited non-zero on 'Zap/26-1'
-- Hungup 'Zap/26-1'
srvradio*CLI
the CLI for whe the call is no answer :
Accepting call from '0661xx' to '534' on channel 0/23, span 1
-- Executing [534@default:1] Dial(Zap/23-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e8b4b0 is ringing
-- Nobody picked up in 1 ms
-- Executing [534@default:2] NoOp(Zap/23-1, Dial status is
NOANSWER) in new stack
-- Executing [534@default:3] GotoIf(Zap/23-1, 0?answered) in new
stack
-- Executing [534@default:4] Goto(Zap/23-1, home|s|1) in new stack
-- Goto (home,s,1)
-- Executing [s@home:1] SetGlobalVar(Zap/23-1,
sounds_path=/var/lib/asterisk/sounds/) in new stack
== Setting global variable 'sounds_path' to '/var/lib/asterisk/sounds/'
-- Executing [s@home:2] BackGround(Zap/23-1,
/var/lib/asterisk/sounds/welcome) in new stack
-- Zap/23-1 Playing '/var/lib/asterisk/sounds/welcome' (language 'en')
-- Channel 0/23, span 1 got hangup request, cause 16
== Spawn extension (home, s, 2) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
2013/7/26 A J Stiles asterisk_l...@earthshod.co.uk
* THIS IS NOT WHERE YOUR RESPONSE GOES *
On Friday 26 July 2013, Salaheddine Elharit wrote:
thanks for your response
but i get the same result i can't execut the next (go to home,s,1) with
the
code below
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
any help please
Do you get the dial status displayed? Then the NoOp() immediately before
the
GotoIf is executing. It's just possible I messed up the syntax of the
GotoIf() since I can't actually test that right now -- I do have an
Asterisk
box with a dialplan stuffed with GotoIf() statements; but right at the
moment,
I can't get to that machine.
Please paste your CLI output below, for the cases where (1) the call is
answered and (2) the Dial() command times out.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
* PLEASE NOTE: YOUR RESPONSE BELONGS AT THE END, NOT HERE *
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hello,
the CLI for whe the call is answered :
Accepting call from '0661xx' to '534' on channel 0/26, span 1
-- Executing [534@default:1] Dial(Zap/26-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e71378 is ringing
-- SIP/228-09e71378 answered Zap/26-1
== Spawn extension (default, 534, 1) exited non-zero on 'Zap/26-1'
-- Hungup 'Zap/26-1'
srvradio*CLI
the CLI for whe the call is no answer :
Accepting call from '0661xx' to '534' on channel 0/23, span 1
-- Executing [534@default:1] Dial(Zap/23-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e8b4b0 is ringing
-- Nobody picked up in 1 ms
-- Executing [534@default:2] NoOp(Zap/23-1, Dial status is
NOANSWER) in new stack
-- Executing [534@default:3] GotoIf(Zap/23-1, 0?answered) in new
stack
-- Executing [534@default:4] Goto(Zap/23-1, home|s|1) in new stack
-- Goto (home,s,1)
-- Executing [s@home:1] SetGlobalVar(Zap/23-1,
sounds_path=/var/lib/asterisk/sounds/) in new stack
== Setting global variable 'sounds_path' to '/var/lib/asterisk/sounds/'
-- Executing [s@home:2] BackGround(Zap/23-1,
/var/lib/asterisk/sounds/welcome) in new stack
-- Zap/23-1 Playing '/var/lib/asterisk/sounds/welcome' (language
'en')
-- Channel 0/23, span 1 got hangup request, cause 16
== Spawn extension (home, s, 2) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
Right. I can see what is going on now. After executing step 1, Dial(), if
the call was answered and hung up, the dialplan is not going on to step 2.
So, try adding a step 102:
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
See what happens now if the call is answered. If you get We reached step
102, then this is where your answered stuff needs to be.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
A J Stiles wrote: * PLEASE NOTE: YOUR RESPONSE BELONGS AT THE END, NOT HERE * On Wednesday 31 July 2013, Salaheddine Elharit wrote: hello, the CLI for whe the call is answered : Accepting call from '0661xx' to '534' on channel 0/26, span 1 -- Executing [534@default:1] Dial(Zap/26-1, SIP/228| 10) in new stack -- Called 228 -- SIP/228-09e71378 is ringing -- SIP/228-09e71378 answered Zap/26-1 == Spawn extension (default, 534, 1) exited non-zero on 'Zap/26-1' -- Hungup 'Zap/26-1' srvradio*CLI As dialplan execution stops if the outgoing call is answered and then bridged the approach of using a Goto afterwards for ANSWER as well will not work. You *must* use the h extension that was previously mentioned to cover this case. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 IVR
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
2013/7/31 Joshua Colp jc...@digium.com
A J Stiles wrote:
* PLEASE NOTE: YOUR RESPONSE BELONGS AT THE END, NOT HERE *
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hello,
the CLI for whe the call is answered :
Accepting call from '0661xx' to '534' on channel 0/26, span 1
-- Executing [534@default:1] Dial(Zap/26-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-09e71378 is ringing
-- SIP/228-09e71378 answered Zap/26-1
== Spawn extension (default, 534, 1) exited non-zero on 'Zap/26-1'
-- Hungup 'Zap/26-1'
srvradio*CLI
As dialplan execution stops if the outgoing call is answered and then
bridged the approach of using a Goto afterwards for ANSWER as well will not
work. You *must* use the h extension that was previously mentioned to cover
this case.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
--
__**__**_
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Re: [asterisk-users] asterisk and IVR
Salaheddine Elharit wrote:
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
As I mentioned this won't work for ANSWER.
exten = h,1,NoOp(Dialing attempt got status ${DIALSTATUS})
will
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] asterisk and IVR
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
So it looks as though it's breaking out of the extension logic altogether, if
the call gets answered. In that case, you'll have to do it the old-fashioned
way: Create a h extension (which fires when a call is hung up) *in the
same context as your 534 extension* (you can have a h extension in each
context, if needs be), and do all your fancy end-of-call stuff there.
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,Goto(home,s,1)
exten = h,1,NoOp(Hangup received. Dial status is ${DIALSTATUS})
Note that if there are other extensions in the context, h will be called when
they get hung up -- you might need some logic in there to deal with this (or
cheat by just having one extension besides h in this context, and use a fully-
specified Goto() to jump into it.)
--
AJS
Answers come *after* questions.
--
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Re: [asterisk-users] asterisk and IVR
hi
in the CLI i have :
1) for CONGESTION i get the status is 'CONGESTION'
Accepting call from '06' to '534' on channel 0/12, span 1
-- Executing [534@default:1] Dial(Zap/12-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08361358 is ringing
-- Got SIP response 480 Temporarily Unavailable back from
192.168.5.131
-- SIP/228-08361358 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'Zap/12-1' status is 'CONGESTION'
2) for no answer i get status is 'NOANSWER'
Accepting call from '06' to '534' on channel 0/4, span 1
-- Executing [534@default:1] Dial(Zap/4-1, SIP/228| 10) in new stack
-- Called 228
-- SIP/228-08362880 is ringing
-- Nobody picked up in 1 ms
== Auto fallthrough, channel 'Zap/4-1' status is 'NOANSWER'
3) for answered i don't get the status is 'answered'
Accepting call from '06' to '534' on channel 0/15, span 1
-- Executing [534@default:1] Dial(Zap/15-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08363bb8 is ringing
-- SIP/228-08363bb8 answered Zap/15-1
when i have the result is 'CONGESTION' or 'NOANSWER'i can go to the next
(home,s,1)
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = CONGESTION])
exten = 534,n,Goto(home,s,1)
how to do in order to go to the next if the result is answered
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
thanks and regards
2013/7/25 Salaheddine Elharit salah.elharit...@gmail.com
ok thank you i will verify and i will update you
thanks for your help
2013/7/25 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 25 July 2013, Salaheddine Elharit wrote:
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = NOANSWER,1,Goto(call,s,1)
with no answer i can coto [call] without issue but with answer like
below i
can't get [call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = ANSWER,1,Goto(call,s,1)
Immediately after the Dial() statement, add a line like
exten = s,nNoOp(Dial status is ${DIALSTATUS})
That will show you the actual contents of ${DIALSTATUS} in the CLI (in
case
it is not what you are expecting). Call your extension a few times, and
see
exactly what you get when the line is answered, unanswered, engaged and
maybe
if the phone is unplugged.
Instead of having a separate extension named after every possible value of
${DIALSTATUS} it might be easier to use a GotoIf() statement to jump away
in
one case (most sensibly, if the call was answered), and fall through to
the
default otherwise (engaged and phone not connected are similar
enough to
no answer for that probably to be what you want, barring special values
--
feel free to use more GotoIf() statements if required).
Something like:
exten = s,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = s,n,NoOp(execution continues here if no answer)
...
exten = s,n,Hangup()
exten = s,n(answered),NoOp(we jump here if call was answered)
...
exten = s,n,Hangup()
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Re: [asterisk-users] asterisk and IVR
* THIS IS NOT WHERE YOUR RESPONSE GOES *
On Friday 26 July 2013, Salaheddine Elharit wrote:
in the CLI i have :
1) for CONGESTION i get the status is 'CONGESTION'
Accepting call from '06' to '534' on channel 0/12, span 1
-- Executing [534@default:1] Dial(Zap/12-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08361358 is ringing
-- Got SIP response 480 Temporarily Unavailable back from
192.168.5.131
-- SIP/228-08361358 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'Zap/12-1' status is 'CONGESTION'
2) for no answer i get status is 'NOANSWER'
Accepting call from '06' to '534' on channel 0/4, span 1
-- Executing [534@default:1] Dial(Zap/4-1, SIP/228| 10) in new
stack -- Called 228
-- SIP/228-08362880 is ringing
-- Nobody picked up in 1 ms
== Auto fallthrough, channel 'Zap/4-1' status is 'NOANSWER'
3) for answered i don't get the status is 'answered'
Accepting call from '06' to '534' on channel 0/15, span 1
-- Executing [534@default:1] Dial(Zap/15-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08363bb8 is ringing
-- SIP/228-08363bb8 answered Zap/15-1
when i have the result is 'CONGESTION' or 'NOANSWER'i can go to the next
(home,s,1)
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = CONGESTION])
exten = 534,n,Goto(home,s,1)
how to do in order to go to the next if the result is answered
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
You're nearly there; you need to have a label answered in your dialplan.
This is done by inserting the name, in round brackets, after the priority and
before the following comma. After a Goto() would be an excellent place to put
it. Try this:
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
...
Note that if you answer the phone, as far as Asterisk is concerned, the Dial()
statement is still being executed; so it won't fall through to the next
priority until the phone is hung up.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
thanks for your response
but i get the same result i can't execut the next (go to home,s,1) with the
code below
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
any help please
2013/7/26 A J Stiles asterisk_l...@earthshod.co.uk
* THIS IS NOT WHERE YOUR RESPONSE GOES *
On Friday 26 July 2013, Salaheddine Elharit wrote:
in the CLI i have :
1) for CONGESTION i get the status is 'CONGESTION'
Accepting call from '06' to '534' on channel 0/12, span 1
-- Executing [534@default:1] Dial(Zap/12-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08361358 is ringing
-- Got SIP response 480 Temporarily Unavailable back from
192.168.5.131
-- SIP/228-08361358 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'Zap/12-1' status is 'CONGESTION'
2) for no answer i get status is 'NOANSWER'
Accepting call from '06' to '534' on channel 0/4, span 1
-- Executing [534@default:1] Dial(Zap/4-1, SIP/228| 10) in new
stack -- Called 228
-- SIP/228-08362880 is ringing
-- Nobody picked up in 1 ms
== Auto fallthrough, channel 'Zap/4-1' status is 'NOANSWER'
3) for answered i don't get the status is 'answered'
Accepting call from '06' to '534' on channel 0/15, span 1
-- Executing [534@default:1] Dial(Zap/15-1, SIP/228| 10) in new
stack
-- Called 228
-- SIP/228-08363bb8 is ringing
-- SIP/228-08363bb8 answered Zap/15-1
when i have the result is 'CONGESTION' or 'NOANSWER'i can go to the next
(home,s,1)
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = CONGESTION])
exten = 534,n,Goto(home,s,1)
how to do in order to go to the next if the result is answered
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
You're nearly there; you need to have a label answered in your dialplan.
This is done by inserting the name, in round brackets, after the priority
and
before the following comma. After a Goto() would be an excellent place to
put
it. Try this:
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
...
Note that if you answer the phone, as far as Asterisk is concerned, the
Dial()
statement is still being executed; so it won't fall through to the next
priority until the phone is hung up.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
* THIS IS NOT WHERE YOUR RESPONSE GOES *
On Friday 26 July 2013, Salaheddine Elharit wrote:
thanks for your response
but i get the same result i can't execut the next (go to home,s,1) with the
code below
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
any help please
Do you get the dial status displayed? Then the NoOp() immediately before the
GotoIf is executing. It's just possible I messed up the syntax of the
GotoIf() since I can't actually test that right now -- I do have an Asterisk
box with a dialplan stuffed with GotoIf() statements; but right at the moment,
I can't get to that machine.
Please paste your CLI output below, for the cases where (1) the call is
answered and (2) the Dial() command times out.
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[asterisk-users] asterisk and IVR
Hello list,
i need your help about the IVR please
i have asterisk 1.4 installed and i configure an IVR like below
exten = 529,1,Ringing()
exten = 529,n,Wait(4)
exten = 529,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}welcome)
exten = s,n,WaitExten(5)
exten = s,n,goto(home,s,1)
exten = i,1,Playback(${sounds_path}erreur-saisie)
exten = i,2,goto(home,s,1)
exten = t,1,Goto(home,s,1)
exten = 1,1,Goto(call,s,1)
[call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 30)
exten = s,n,NoOp(User chose support option)
exten = s,n,MYSQL(Connect connid localhost database login password)
exten = s,n,MYSQL(Query resultid ${connid} INSERT\ INTO\ menu\ SET\
callerid='${CALLERID(num)}'\, calldate=now()\, ext=no response\)
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,hangup
when i call the number 529 i can get the home and when i press 1 i get the
call when there is no response from my sip/228 i can store the date and
time in my database
but when i handel the call from my sip i can't store the data in my table
calldate callerid ext
2013-07-25 14:09:20 0661xx No response
my question how can i do in order to store the data in my database with the
ext = response or no response
thanks and regards
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Re: [asterisk-users] asterisk and IVR
On Thursday 25 July 2013, Salaheddine Elharit wrote:
i have asterisk 1.4 installed and i configure an IVR like below
. stuff deleted .
when i call the number 529 i can get the home and when i press 1 i get the
call when there is no response from my sip/228 i can store the date and
time in my database
but when i handel the call from my sip i can't store the data in my table
calldate callerid ext
2013-07-25 14:09:20 0661xx No response
my question how can i do in order to store the data in my database with the
ext = response or no response
You need to do this from an extension called h (which gets run when a call
is hung up), in the same context where the call was placed. You can look at
the variables ${DIALSTATUS} and ${HANGUPCAUSE} to see how the call went.
--
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Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = NOANSWER,1,Goto(call,s,1)
with no answer i can coto [call] without issue but with answer like below i
can't get [call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = ANSWER,1,Goto(call,s,1)
any help please
2013/7/25 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 25 July 2013, Salaheddine Elharit wrote:
i have asterisk 1.4 installed and i configure an IVR like below
. stuff deleted .
when i call the number 529 i can get the home and when i press 1 i get
the
call when there is no response from my sip/228 i can store the date and
time in my database
but when i handel the call from my sip i can't store the data in my table
calldate callerid ext
2013-07-25 14:09:20 0661xx No response
my question how can i do in order to store the data in my database with
the
ext = response or no response
You need to do this from an extension called h (which gets run when a
call
is hung up), in the same context where the call was placed. You can look
at
the variables ${DIALSTATUS} and ${HANGUPCAUSE} to see how the call went.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] asterisk and IVR
On Thursday 25 July 2013, Salaheddine Elharit wrote:
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = NOANSWER,1,Goto(call,s,1)
with no answer i can coto [call] without issue but with answer like below i
can't get [call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = ANSWER,1,Goto(call,s,1)
Immediately after the Dial() statement, add a line like
exten = s,nNoOp(Dial status is ${DIALSTATUS})
That will show you the actual contents of ${DIALSTATUS} in the CLI (in case
it is not what you are expecting). Call your extension a few times, and see
exactly what you get when the line is answered, unanswered, engaged and maybe
if the phone is unplugged.
Instead of having a separate extension named after every possible value of
${DIALSTATUS} it might be easier to use a GotoIf() statement to jump away in
one case (most sensibly, if the call was answered), and fall through to the
default otherwise (engaged and phone not connected are similar enough to
no answer for that probably to be what you want, barring special values --
feel free to use more GotoIf() statements if required).
Something like:
exten = s,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = s,n,NoOp(execution continues here if no answer)
...
exten = s,n,Hangup()
exten = s,n(answered),NoOp(we jump here if call was answered)
...
exten = s,n,Hangup()
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Re: [asterisk-users] asterisk and IVR
ok thank you i will verify and i will update you
thanks for your help
2013/7/25 A J Stiles asterisk_l...@earthshod.co.uk
On Thursday 25 July 2013, Salaheddine Elharit wrote:
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = NOANSWER,1,Goto(call,s,1)
with no answer i can coto [call] without issue but with answer like
below i
can't get [call]
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${DIALSTATUS},1)
exten = ANSWER,1,Goto(call,s,1)
Immediately after the Dial() statement, add a line like
exten = s,nNoOp(Dial status is ${DIALSTATUS})
That will show you the actual contents of ${DIALSTATUS} in the CLI (in
case
it is not what you are expecting). Call your extension a few times, and
see
exactly what you get when the line is answered, unanswered, engaged and
maybe
if the phone is unplugged.
Instead of having a separate extension named after every possible value of
${DIALSTATUS} it might be easier to use a GotoIf() statement to jump away
in
one case (most sensibly, if the call was answered), and fall through to
the
default otherwise (engaged and phone not connected are similar enough
to
no answer for that probably to be what you want, barring special values
--
feel free to use more GotoIf() statements if required).
Something like:
exten = s,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = s,n,NoOp(execution continues here if no answer)
...
exten = s,n,Hangup()
exten = s,n(answered),NoOp(we jump here if call was answered)
...
exten = s,n,Hangup()
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Re: [asterisk-users] asterisk as IVR using 3g usb modem
On Thu, 2012-10-18 at 15:45 +, Mahendra Dobariya wrote: hi, I want to use asterisk as IVR system , but to make and receive GSM call, i want to use 3g usb modem.(voice enabled) http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php and i want to install this system on two different machine 1 on mac os x - 2 raspberry pi- (debian wheezy)--http://www.raspberrypi.org/ thanx in advance.. Are you very sure about the last one (i.e. the r-pi)? These have a very few resourses (cpu, mem) If looking for something small, how about latest pandaboard, a bit more expensive, but less limited: http://www.hardware-modules.com/index.php?page=Browseproduct_type=SBCdesigner=Texas%20Instrumentmodule=Pandaboard%20ES%20(Texas%20Instrument%20-%20OMAP4460)lang=en And still cheaper than most intel-based sff-boards. hw -- _ -- 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 as IVR using 3g usb modem
hi, I want to use asterisk as IVR system ,but to make and receive GSM call, i want to use 3g usb modem.(voice enabled)http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php and i want to install this system on two different machine1 on mac os x -2 raspberry pi- (debian wheezy)--http://www.raspberrypi.org/ thanx in advance.. -- _ -- 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 as IVR using 3g usb modem
Short answer is, its not possible Long answer, why it is not !! U would have to write a dahdi module for this 3G modem to help asterisk understand it as standard gsm channel. Hope that help, Mitul On Oct 18, 2012 9:16 PM, Mahendra Dobariya mahendra_mahen...@hotmail.com wrote: hi, I want to use asterisk as IVR system , but to make and receive GSM call, i want to use 3g usb modem.(voice enabled) http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php and i want to install this system on two different machine 1 on mac os x - 2 raspberry pi- (debian wheezy)--http://www.raspberrypi.org/ thanx in advance.. -- _ -- 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] asterisk as IVR using 3g usb modem
On 18 Oct 2012, at 16:50, Mitul Limbani wrote: U would have to write a dahdi module for this 3G modem to help asterisk understand it as standard gsm channel. Look up chan_datacard (i think that's what it's called from memory). Steve -- _ -- 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 as IVR using 3g usb modem
On Thu, 2012-10-18 at 17:18 +0100, Steven Howes wrote: On 18 Oct 2012, at 16:50, Mitul Limbani wrote: U would have to write a dahdi module for this 3G modem to help asterisk understand it as standard gsm channel. Look up chan_datacard (i think that's what it's called from memory). Steve It got renamed, and is now: http://code.google.com/p/asterisk-chan-dongle/ It's just a tgz, no docu, no wiki.. chan_dongle-1.1.r14.tgz chan_dongle version 1.1 revision 14 sources Featured 6 days ago 6 days ago 184 KB 159 Looks a bit fresh... -- _ -- 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 OUtbound IVR Recording
nO , How to make an out bout call and have a dialplan and record the same . I got it from the VOIP Wiki . Thanks Mahesh From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext Jayson Baker Sent: Sunday, October 10, 2010 3:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk OUtbound IVR Recording cmd record ? On Sat, Oct 9, 2010 at 1:28 AM, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: HI, I have a scenario like the following . A user clicks on the web page . This triggers an outbound call to users phone number . Now the user has to leave a message . What is the best way of doing this ? Do we have any example of such a dial plan . Regards Mahesh -- _ -- 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] Asterisk OUtbound IVR Recording
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jayson Baker
Sent: Saturday, October 09, 2010 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk OUtbound IVR Recording
cmd record ?
On Sat, Oct 9, 2010 at 1:28 AM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
HI,
I have a scenario like the following .
A user clicks on the web page . This triggers an outbound call to users
phone number .
Now the user has to leave a message .
What is the best way of doing this ? Do we have any example of such a
dial plan .
Regards
Mahesh
This is a simple context that plays static messages welcome, important and
calllater. It plays a passed message as well. To use as you want, just
replace Background(${Data}) with Record(${Data}.gsm). Lines 3-4 incorporate
a wait if the call isn't a SIP line because DAHDI has a 3-7 second delay on
Answer (worse if calling a cell phone).
[accept]
exten = s,1,Answer
exten = s,n,Set(IVRTRY=0)
exten = s,n,Gotoif($[${EXTEN} SIP]?start)
exten = s,n,Wait(9)
exten = s,n(start),Background(welcome)
exten = s,n,Background(important)
exten = s,n,WaitExten(5,m)
exten = s,n,Set(IVRTRY=$[${IVRTRY} +1])
exten = s,n,Verbose(Try ${IVRTRY})
exten = s,n,Gotoif($[${IVRTRY} 4]?accept|s|start)
exten = s,n,Goto(end-call|s|1)
exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))
exten = 1,n,Background(${Data})
exten = 1,n,Background(repeatmsg)
exten = 1,n,WaitExten(5,m)
exten = 1,n,Goto(end-call|s|1)
exten = 2,1,Background(calllater)
exten = 2,n,ForkCDR(v,s(reject=${Data}))
exten = 2,n,Goto(end-call|s|1)
exten = 3,1,Goto(accept|1|2)
exten = *,1,Goto(accept|s|1)
exten = i,1,Goto(accept|s|1)
exten = t,1,Goto(accept|s|1)
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[asterisk-users] Asterisk OUtbound IVR Recording
HI, I have a scenario like the following . A user clicks on the web page . This triggers an outbound call to users phone number . Now the user has to leave a message . What is the best way of doing this ? Do we have any example of such a dial plan . Regards Mahesh -- _ -- 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 OUtbound IVR Recording
cmd record ? On Sat, Oct 9, 2010 at 1:28 AM, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: HI, I have a scenario like the following . A user clicks on the web page . This triggers an outbound call to users phone number . Now the user has to leave a message . What is the best way of doing this ? Do we have any example of such a dial plan . Regards Mahesh -- _ -- 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 - VoiceGenie IVR
Hi,
I'm currently working on a setup between Asterisk and VoiceGenie (which
is a IVR system).
The way my setup is done, is that I have a PRI line coming in my
Asterisk server, and then VoiceGenie is connected to Asterisk via SIP,
like any other softphone basically. I'm able to receive calls in
Asterisk and then link them with VoiceGenie. But one of my issues is
that when I get an outside call, transfer the call to VoiceGenie, then
for that specific calls VoiceGenie would decide that this call has to be
transfered to an outside party, so then VoiceGenie calls up that number,
it goes through Asterisk and it reached the other person. But the link
doesn't stay up very long, max 15 seconds.
That's one of the errors that I see in Asterisk(for obvious reasons I've
replaced some numbers with *):
-- Hungup 'Zap/8-1'
Feb 15 14:10:19 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 1
(Critical Response)
Feb 15 14:10:28 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 1
(Critical Response)
-- Hungup 'Zap/1-1'
Here's a part of my dialplan for outside calls:
exten = _9XX,1,Set(CALLERID(all)=450-655-)
exten = _9XX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
And here's a Macro that I use for incoming call for VoiceGenie:
[macro-voicegenie]
exten = s,1,Answer
exten = s,2,SIPAddHeader(X-Asterisk-DID: ${ARG1})
exten = s,3,SIPAddHeader(X-Asterisk-CallerName: ${ARG2})
exten = s,4,Dial(SIP/108)
exten = 514380,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514380,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
Here's the config in sip.conf:
[108]
type=friend
context=internal
host=10.1.1.40
callerid=VoiceGenie 108
progressinband=never
disallow=all
allow=ulaw
Also, the support team at Voicegenie they asked me if I stop sending
183 Session Progress before 180 Ringing.
It seems that this could be part of my issue.
Thanks,
--
Eric Rousse
System Administrator
514.380.2992
450.655.1001
1.888.641.5800
Telmatik inc.
204 Montarville, suite 250
Boucherville, QC, Canada
J4B 6S2
www.telmatik.com
___
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[Asterisk-Users] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite
Hello: Please go through this mail, it won't take much time. Fora Unified Messaging system, I have to develop an IVR/Voicemail system based on asterisk. Following are the details of the voice mail / IVR (Interactive voice Response) system to be developed: Develop using Asterisk 100 simultaneous users Integrate with @mail ( www.atmail.com) our email program ( developed in perl ) Provide Fax integration Provide information about the message store, e.g. you have two voice messages and three text messages (emails). Integrate with backend database. Accept response through telephone keypad or through voice response Forward all voicemail to email as well Users can access voicemail through telephone, cell phone, or through a desktop/webmail client Send SMS or IM alerts on receipt of specified messages Identify caller/device and provide him options accordingly. If the owner of the mailbox calls it should give different options than if somebody else calls possibly to leave a message. In-fact there are three different types of callers the system should recognize: the owner, other subscribers to our system, everybody else. Receive voicemails from outside through telephony devices (dialed in), and also receive voicemail from within the system through email or desktop client. Integration with VOIP service/connections and DTMF connection. Integration with IM Obviously provide remote access Text to speech for reading emails or just their headers Please tell me, What asterisk offers by default for my system? What areas I need to work to achieve the mentioned features? What specific hardware(cards,servers etc.)should I look for this system? I appreciate any comments/suggessions/queries. Thanks for reading, Jami ___ --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] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite
Hello: Please go through this mail, it won't take much time. Fora Unified Messaging system, I have to develop an IVR/Voicemail system based on asterisk. Following are the details of the voice mail / IVR (Interactive voice Response) system to be developed: Develop using Asterisk 100 simultaneous users Integrate with @mail ( www.atmail.com) our email program ( developed in perl ) Provide Fax integration Provide information about the message store, e.g. you have two voice messages and three text messages (emails). Integrate with backend database. Accept response through telephone keypad or through voice response Forward all voicemail to email as well Users can access voicemail through telephone, cell phone, or through a desktop/webmail client Send SMS or IM alerts on receipt of specified messages Identify caller/device and provide him options accordingly. If the owner of the mailbox calls it should give different options than if somebody else calls possibly to leave a message. In-fact there are three different types of callers the system should recognize: the owner, other subscribers to our system, everybody else. Receive voicemails from outside through telephony devices (dialed in), and also receive voicemail from within the system through email or desktop client. Integration with VOIP service/connections and DTMF connection. Integration with IM Obviously provide remote access Text to speech for reading emails or just their headers Please tell me, What asterisk offers by default for my system? What areas I need to work to achieve the mentioned features? What specific hardware(cards,servers etc.)should I look for this system? I appreciate any comments/suggessions/queries. Thanks for reading, Jami ___ --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] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite
I am pretty sure Asterisk can handle point 1. PaulH ammad jami [EMAIL PROTECTED] wrote: Hello: Please go through this mail, it won't take much time. For a Unified Messaging system, I have to develop an IVR/Voicemail system based on asterisk. ** Following are the details of the voice mail / IVR (Interactive voice Response) system to be developed: 1. Develop using Asterisk 2. 100 simultaneous users 3. Integrate with @mail ( www.atmail.com) our email program *( developed in perl )* 4. Provide Fax integration 5. Provide information about the message store, e.g. you have two voice messages and three text messages (emails). Integrate with backend database. 6. Accept response through telephone keypad or through voice response 7. Forward all voicemail to email as well 8. Users can access voicemail through telephone, cell phone, or through a desktop/webmail client 9. Send SMS or IM alerts on receipt of specified messages 10. Identify caller/device and provide him options accordingly. If the owner of the mailbox calls it should give different options than if somebody else calls possibly to leave a message. In-fact there are three different types of callers the system should recognize: the owner, other subscribers to our system, everybody else. 11. Receive voicemails from outside through telephony devices (dialed in), and also receive voicemail from within the system through email or desktop client. 12. Integration with VOIP service/connections and DTMF connection. 13. Integration with IM 14. Obviously provide remote access 15. Text to speech for reading emails or just their headers Please tell me, What asterisk offers by default for my system? What areas I need to work to achieve the mentioned features? What specific hardware(cards,servers etc.) should I look for this system? I appreciate any comments/suggessions/queries. Thanks for reading, Jami ___ --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] Asterisk-to-IVR Problem
This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line -X100P-asterisk-TDM10B- phone cord-Dialogic analog port-IVR system. Asterisk should dial the IVR system, which should answer and play back its IVR scenario script to the caller. However, when a call comes in, asterisk answers on Zap1-1 and dials Zap2-1. The IVR system answers the call and begins to play back its scenario. After 5 to 15 seconds, asterisk apparently senses an on-hook condition (exception 17?) and disconnects the connection bridge. The logs on the IVR system shows that it is not initially aware of the hangup, and continues playing its scenario. Going to an analog phone in the TDM10B instead of the IVR system appears to work OK, with the exception that asterisk is still sending dial tones when the analog phone is answered. The phone stays connected to asterisk until it really does hang up. What causes the hangup? What generates the exception? I have looked at the chan_zap.c code and can not see how zt_exception gets into the picture. Is there a TDMF incompatibility problem? Is this a case where DAX should be used? My config files and sample debug output are given below. Thanks for anyone's help. Rollin Weeks # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # fxsks=1 # For the X100P fxoks=2 # For the TDM400P (TDM10B) # loadzone = us # defaultzone=us ; ; Zapata telephony interface ; ; Configuration file [channels] ; language=en context=incoming switchtype=national signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; Whether or not to use caller ID ; usecallerid=asreceived ; Whether or not to hide outgoing caller ID (Override with *67 or *82) ; hidecallerid=no ; callwaitingcallerid=yes ; for default voicemail context, the example below is fine: ; mailbox=1234 echocancel=yes ; If you are having trouble with DTMF detection, you can relax the ; DTMF detection parameters. Relaxing them may make the DTMF detector ; more likely to have talkoff where DTMF is detected when it ; shouldn't be. ; relaxdtmf=yes ; You may also set the default receive and transmit gains (in dB) ; rxgain=0.0 txgain=0.0 immediate=no ; On trunk interfaces (FXS) and EM interfaces (EM, Wink, Feature Group D ; etc, it can be useful to perform busy detection either in an effort to ; detect hangup or for detecting busies ; busydetect=yes ; ; If busydetect is enabled, is also possible to specify how many ; busy tones to wait before hanging up. The default is 4, but ; better results can be achieved if set to 6 or even 8. Mind that ; higher the number, more time is needed to hangup a channel, but ; lower is probability to get random hangups ; busycount=40 ; Select which class of music to use for music on hold. If not specified ; then the default will be used. ; musiconhold=default channel = 1 signalling=fxo_ks context=internal channel = 2 ; Home grown extension file [globals] ;RECEPTIONIST=Zap/1 ; [incoming] exten = s,1,Wait(1) exten = s,2,Answer() exten = s,3,Playback(demo-congrats) ; Plays the demo-congrats file after answering the line exten = s,4,Dial,Zap/2/1000\20 exten = s,5,Hangup [internal] exten = 1000,1,Answer() exten = 1000,2,Wait(2) exten = 1000,3,Hangup --- -- Starting simple switch on 'Zap/1-1' Urgent handler Aug 4 15:43:39 DEBUG[5059]: pbx.c:1274 pbx_extension_helper: Launching 'Wait' -- Executing Wait(Zap/1-1, 1) in new stack Urgent handler Aug 4 15:43:40 DEBUG[5059]: pbx.c:1274 pbx_extension_helper: Launching 'Answer' -- Executing Answer(Zap/1-1, ) in new stack Urgent handler Aug 4 15:43:40 DEBUG[5059]: chan_zap.c:2301 zt_answer: Took Zap/1-1 off hook Aug 4 15:43:40 DEBUG[5059]: chan_zap.c:1231 zt_enable_ec: Enabled echo cancellation on channel 1 Aug 4 15:43:40 DEBUG[5059]: chan_zap.c:1250 zt_train_ec: No echo training requested Aug 4 15:43:40 DEBUG[5059]: pbx.c:1274 pbx_extension_helper: Launching 'Playback' -- Executing Playback(Zap/1-1, demo-congrats) in new stack Urgent handler Aug 4 15:43:40 DEBUG[5059]: channel.c:1719 ast_set_write_format: Set channel Zap/1-1 to write format gsm Aug 4 15:43:40 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'demo-congrats' (language 'en') Urgent handler Aug 4 15:44:08 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Aug 4 15:44:08 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Aug 4 15:44:08 DEBUG[5059]: channel.c:1719 ast_set_write_format: Set channel Zap/1-1 to write format ulaw Aug 4 15:44:08 DEBUG[5059]: pbx.c:1274
[Asterisk-Users] Asterisk scalability IVR/Voicemail only
I have searched a bit on the Wiki and mailing list archives, but didnt see direct information regarding my scenario: 1. Asterisk for IVR/Voicemail ONLY (no PSTN, no MOH) 2. BudgeTone IP phones and HandyTone 286 ATAs 3. SIP only - separate Proxy+Registrar+CallRouter on other servers 4. G.711u codec, dtmfmode=rfc2883 5. No NAT/firewall (private ethernet network) What I'm looking for is scalability factors: 1. concurrent users accessing IVR and retrieving VM 2. concurrent mailboxes receiving VM (greeting playback, recording a msg) 3. impact of using configuration files vs. postgres vs. mysql Considering these specs for the Asterisk server, will adjust in accordance with scalability forecast: RHEL3 on single Xeon 3.2GHz, 4GB RAM Does anyone have general statistics/findings? Much appreciated! My apologies if this info is already out there somewhere in the archives. ___ 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
[Asterisk-Users] Asterisk dimensioning (IVR, mass calling)
Hi, I presently have 6 PRIs of IVR traffic that I am planning to migrate from Dialogic on SCO-Unix to Digium-Asterisk on Debian. Here is the general description of the traffic in question : - IVR system, 138 PRI channels (6 PRIs, multiple D-channel) - Some traffic from TV ads, so all traffic typically arrives within a few seconds - Up to 12-15 simul. outbound (external) conferencing calls for customer service - No internal phones - Several HTTPS transactions per minute - MySQL queries (to separate server) - Want do drop minimum of calls, mostly 900 services I am assuming the following things, performance-wise : - Since I will be recording to and playing back from ?-law, which is the format used by PRI in North America, I will require no codec translation and that should be the easiest on the CPU (right?). - I will need no echo cancellation since I am exclusively on PRI (right?). If I rely on previous posts, mainly from Scott Stingel and Azher Amin, I won't be able to put all this in a single server. I would instead have 2-3 servers (ex. Xeon bi-proc.) with 2-3 PRIs each, and NFS for dynamic content (recordings from callers). Does this sound realistic? Is it too risky to go with Asterisk with such a set-up? Thanks a lot for your help, Yves Chouinard Vox-Tel ___ 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
