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
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
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
* 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)
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]
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)
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 =
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 =
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
* 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,
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 =
* 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 ${DIALSTATUS})
Hello list,
i need your help about the IVR please
i have asterisk 1.4 installed and i configure an IVR like below
exten = 529,1,Ringing()
exten = 529,n,Wait(4)
exten = 529,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten =
On Thursday 25 July 2013, Salaheddine Elharit wrote:
i have asterisk 1.4 installed and i configure an IVR like below
. stuff deleted .
when i call the number 529 i can get the home and when i press 1 i get the
call when there is no response from my sip/228 i can store the date and
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${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
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
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)
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
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
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
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
--
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).
: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
_
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
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
--
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
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
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
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
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
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
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.
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
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