[asterisk-users] limitation on number of contexts in extensions.conf
Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include filename) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -- _ -- 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] limitation on number of contexts in extensions.conf
Enter CLI via /usr/sbin/asterisk -r and execute dialplan reload. Any errors? BTW: you should think about upgrading to 1.8 (for example). Am 25.07.2013 08:49, schrieb Kamlesh Kumar: Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include filename) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -- _ -- 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] Echo Cancellation
25.07.2013 13:51, bilal ghayyad пишет: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? oslec, imho. -- _ -- 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] Echo Cancellation
On 07/25/2013 11:51 AM, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Use the free OSLEC echo canceller software module or Digium's commercial HPEC echo canceller software module. Google is your friend. Regards, Patrick -- _ -- 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] Echo Cancellation
Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Regards Bilal -- _ -- 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] Echo Cancellation
On 7/25/2013 5:57 AM, Patrick Lists wrote: On 07/25/2013 11:51 AM, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Use the free OSLEC echo canceller software module or Digium's commercial HPEC echo canceller software module. Google is your friend. Regards, Patrick +1 for OSLEC JohnM -- _ -- 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 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 ${connid})
exten = s,n,hangup
when i call the number 529 i can get the home and when i press 1 i get the
call when there is no response from my sip/228 i can store the date and
time in my database
but when i handel the call from my sip i can't store the data in my table
calldate callerid ext
2013-07-25 14:09:20 0661xx No response
my question how can i do in order to store the data in my database with the
ext = response or no response
thanks and regards
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Re: [asterisk-users] What is my syntax error here?
Have you tried without the double quotes ?
as :
exten = _417XX,n,GotoIf($[$[${CallerIDNum} 41799] | $[${CallerIDNum}
41700]]?notfromlocal:)
From: James B. Byrne byrn...@harte-lyne.ca
To: asterisk-users@lists.digium.com
Sent: Wednesday, July 24, 2013 10:14 PM
Subject: Re: [asterisk-users] What is my syntax error here?
On Wed, July 24, 2013 10:33, James B. Byrne wrote:
Additional data:
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4
FreePBX = 2.11.0.4
. . .
So my question is simple. What error in syntax have I committed here?
I expect that CallerIDNum == 41712 in the check:
exten = _417XX,n,GotoIf(
$[$[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700]]?notfromlocal:)
But I am getting a message say there is no variable to check. So what
I have done that is wrong?
As suggested I made these additions to the dial plan:
; Line 8
exten = _417XX,n,NoOp($[${CallerIDNum} 41799])
; Line 9
exten = _417XX,n,NoOp($[${CallerIDNum} 41700])
; Line 10
exten = _417XX,n,NoOp($[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700])
; Line 11
exten = _417XX,n,NoOp($[${CallerIDNum} 41799] |
$[${CallerIDNum} 41700])
; Line 12
exten = _417XX,n,NoOp($[$[${CallerIDNum} 41799] ||
$[${CallerIDNum} 41700]])
; Line 13
exten = _417XX,n,NoOp($[$[${CallerIDNum} 41799] ||
$[${CallerIDNum} 41700]])
; Line 14 - original
exten = _417XX,n,GotoIf(
$[$[${CallerIDNum} 41799] ||
$[${CallerIDNum} 41700]]?notfromlocal:)
Which changed nothing but the results did provide a clue. Taking the
earlier suggestion I ensured that my original line did not contain
line breaks, which I cannot reproduce in this email because of its
length. However, putting everything on one line caused the missing
variable error to disappear.
exten = _417XX,n,GotoIf($[$[${CallerIDNum} 41799] ||
$[${CallerIDNum} 41700]]?notfromlocal:)
Thank you both for the help. I much appreciate it.
--
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James B. Byrne mailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
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[asterisk-users] Asterisk dropping calls on transfer
Hi, I'm having a weird problem with asterisk (asterisk18-1.8.12.2_1). Every call on the system, whatever it comes from the PSTN or from local extensions, when we hit the '#' button to transfer the call, asterisk just disconnects it, without any error or log, here is my current features configuration: Builtin Feature Default Current --- --- --- Pickup *8 *8 Blind Transfer # *4 Attended Transfer # One Touch Monitor Disconnect Call * * Park Call One Touch MixMonitor Dynamic Feature Default Current --- --- --- (none) Feature Groups: --- (none) Call parking (Parking lot: default) Parking extension : 70 Parking context : parkedcalls Parked call extensions: 71-91 Parkingtime : 45000 ms MusicOnHold class : default Enabled : Yes Can anybody help me? -- _ -- 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
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 ${HANGUPCAUSE} to see how the call went.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Echo Cancellation
On Thu, Jul 25, 2013 at 02:51:02AM -0700, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Regards Bilal Also, just FYI, those cards do support adding a hardware echocancelation module. But I would recommend trying the software solutions first. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and IVR
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${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 ${HANGUPCAUSE} to see how the call went.
--
AJS
Answers come *after* questions.
--
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Re: [asterisk-users] limitation on number of contexts in extensions.conf
On Wed, Jul 24, 2013 at 11:49 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include filename) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. There probably is a limit, but I don't know what it is. We have many hundreds of contexts and around 80 include files in our main server. My guess is you have an error somewhere. If you show dialplan, does it seem to stop at a certain point as if it loaded only up to a certain file/directory? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and IVR
On Thursday 25 July 2013, Salaheddine Elharit wrote:
thanks for your help when i use
exten = s,1,NoOp(User chose support option)
exten = s,n,Dial(SIP/228, 10)
exten = s,n,Goto(${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] Meetme and maxusers option
Thiago wrote: I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Asterisk version? Any error messages? Is the conference you are attempting to limit stored in a db (Realtime)? Dan -- _ -- 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] DAHDI - Tickless Kernel?
Greetings- I'm running some USB DAHDI hardware on a system with a tickless kernel. The audio quality is quite poor. Could the tickless kernel be to blame? If so, when recompiling a kernel that is *not* tickless, is there a recommended KERNEL_HZ value? IIRC, older kernels used to be 1000, but newer ones are 250. Thoughts? --Tim -- _ -- 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] Random dead calls
Hi, Am getting dead or silence calls at sometimes for my agents, when I checked my CDR the caller-id shows my vendor's name and some shows as real customer name. When I call back again the real customer's number its reaching, the answering machine owned by customer. I have a confusion, or how to find out are these numbers are from any auto dialer or from real customers. 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
