Re: [asterisk-users] limitation on number of contexts in extensions.conf
Thank you Carlos, you were right, there was one empty file among all included files which were causing this problem. Couple of more queries: Will system performance be affected if there are 20k dialplan entries(including all external files and contexts) in extensions.conf? Can we define variable in external file, and include that external file in extensions.conf and then use that variable in dialplan? Thanks, Kamlesh Date: Thu, 25 Jul 2013 08:50:39 -0700 From: car...@televolve.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] limitation on number of contexts in extensions.conf On Wed, Jul 24, 2013 at 11:49 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: 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 AlvarezTelEvolve602-889-3003 -- _ -- 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 -- _ -- 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] Random dead calls
Gopalakrishnan N писал 26.07.2013 08:55: I have a confusion, or how to find out are these numbers are from any auto dialer or from real customers. At least you need to check debug log, it will show you in which context or which app was the call originated and how -- With Best Regards Mikhail Lischuk -- _ -- 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
On Friday 26 July 2013, Kamlesh Kumar wrote: Thank you Carlos, you were right, there was one empty file among all included files which were causing this problem. Couple of more queries: Will system performance be affected if there are 20k dialplan entries(including all external files and contexts) in extensions.conf? Not by as much as you think, because the dialplan is compiled into an intermediate form when Asterisk starts (and again when you execute `dialplan reload`) -- it doesn't have to parse the whole text file for every call. Can we define variable in external file, and include that external file in extensions.conf and then use that variable in dialplan? Yes (and that's a sensible way of doing it anyway). Just remember, a variable won't have a value until the include statement which includes the file with the line that defines it is parsed. -- 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
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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Answers come *after* questions.
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[asterisk-users] Dial plan flow control
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4
FreePBX = 2.11.0.4
I am trying to understand flow control in Asterisk dial plans and not
having very much luck. I have read the Asterisk book from O'Rielly,
or at least those parts I believe might apply, but that has not helped
me much on this particular issue.
What I wish is to set three distinct ring tones on our Snom phones for
external, internal and transferred calls. The first is accomplished
simply enough by setting the ALERT_INFO setting on the inbound route
for all phones. Done.
The second was much more complicated but I found a recipe which
demonstrated how to do that using extensions_override_freepbx.conf and
eventually got it working. Done.
However, in my ignorance I believed that I could use the same
technique, indeed the same code, to check whether the call was
internal or a transfer. In this belief I appear sadly mistaken.
So, I am left with trying to understand the nature of flow control in
Asterisk dial plans and specifically those distributed with FreePBX.
In FreePBX I see this in extensions.conf:
;-
; from-internal:
;
; Internal dialplan that most internal phones have access to
;
[from-internal]
include = from-internal-noxfer
include = from-internal-xfer
include = bad-number ; auto-generated
;-
;-
; from-internal-noxfer:
;
; Place to put internal dialplan that should not be accessible during
; a blind transfer, this context will not be visible during such.
;
[from-internal-noxfer]
include = from-internal-noxfer-custom
include = from-internal-noxfer-additional ; auto-generated
;-
;-
; from-internal-xfer:
;
; Place to put most internal dialplan, will be visible during
: normal calls and blind transfers.
;
[from-internal-xfer]
include = from-internal-custom
include = from-internal-additional ; auto-generated
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
;-
What I would like to do is to add checks for whether or not a call is
internal or transferred between extensions in the
[from-internal-custom] context, which is presumably best placed in the
file named /etc/asterisk/extensions_custom.conf. To begin testing I
did this
[from-internal-custom]
include = set-snom-ringtone-variables
[set-snom-ringtone-variables]
exten = _X.,1,Noop(CALLERID_ALL=${CALLERID(all)})
exten = _X.,n,Set(CallerIDNum=${CALLERID(num)})
Which simply does not work at all. The effect is that the extensions
stop working. So, clearly I misunderstand something very basic about
flow control and thus my question. How do I return from my
from-internal-custom context back to the from-internal-xfer context at
the point following the include = from-internal-custom statement?
Thank you.
--
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[asterisk-users] RTP from pcap file
Howdy all, Does anyone know of a niffty CLI tool for Linux that can take a PCAP file that was created on a SIP PBX for example, and then dump the payload of the various RTP streams in there into seperate files so I can listen to them? I can go this graphically with Wireshark, but I'd like to script it for automation. Cheers, James. -- _ -- 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 ip authentication
Hi all, I've tried to sen calls to asterisk from different soft switch. I want to define ip authentication(not register) to an extension for make call through asterisk. Is there any way to make call from asterisk without register. Only ip authentication. I tried too many different configurations but it hasn't worked. This is my sip.conf --sip.conf [] host=x.x.x.x qualify=yes type=peer insecure=port,invite context=from-internal disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm But gives SIP/2.0 401 Unauthorized error. Kind Regards. -- _ -- 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 ip authentication
You should take a look at this options: type=friend context=my_context host=ip_address Am 26.07.2013 16:52, schrieb jin jan: Hi all, I've tried to sen calls to asterisk from different soft switch. I want to define ip authentication(not register) to an extension for make call through asterisk. Is there any way to make call from asterisk without register. Only ip authentication. I tried too many different configurations but it hasn't worked. This is my sip.conf --sip.conf [] host=x.x.x.x qualify=yes type=peer insecure=port,invite context=from-internal disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm But gives SIP/2.0 401 Unauthorized error. Kind Regards. -- _ -- 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 ip authentication
Additionally you shoudl take a look at sip set debug on (in cli) and then place a call. Am 26.07.2013 17:14, schrieb Thorsten Göllner: You should take a look at this options: type=friend context=my_context host=ip_address Am 26.07.2013 16:52, schrieb jin jan: Hi all, I've tried to sen calls to asterisk from different soft switch. I want to define ip authentication(not register) to an extension for make call through asterisk. Is there any way to make call from asterisk without register. Only ip authentication. I tried too many different configurations but it hasn't worked. This is my sip.conf --sip.conf [] host=x.x.x.x qualify=yes type=peer insecure=port,invite context=from-internal disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm But gives SIP/2.0 401 Unauthorized error. Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] RTP from pcap file
On Fri, 26 Jul 2013, James Bensley wrote: Does anyone know of a niffty CLI tool for Linux that can take a PCAP file that was created on a SIP PBX for example, and then dump the payload of the various RTP streams in there into seperate files so I can listen to them? http://wiki.freeswitch.org/wiki/Packet_Capture I think pcapsipdump is what you are looking for. -- 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
[asterisk-users] Sending 603 Declined message
In my dialplan I'd like to send a 603 Declined message to the user placing the call. I see the commands for the Busy and Congestion, but not the one for the Declined. Any help? Leandro -- _ -- 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] DAHDI - Tickless Kernel?
On Thu, Jul 25, 2013 at 03:52:49PM -0500, Tim Nelson wrote: 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. I doubt it's the tickless kernel that is causing your issue. Normally if you have DAHDI hardware, then it should be generating interrupts which drive the mixing / passing of audio. There shouldn't be any reliance on the system timer tick. Also, for most Asterisk installations that are using the system timer for mixing I've not seen any problems with 250 HZ. Most VOIP systems mix audio at in at least 20ms size packets, so even at 250 HZ there are 5 timer expirations for each audio packet. Keeping the kernel tickless should also work fine here when using a software timer to mix your audio since the kernel will set the timer to fire at the next timer expiration (whatever that was set to). So, in summary...I've not seen any problems related to the tickless kernel. -- 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] Sending 603 Declined message
On 26/07/13 16:32, Leandro Dardini wrote: In my dialplan I'd like to send a 603 Declined message to the user placing the call. I see the commands for the Busy and Congestion, but not the one for the Declined. Any help? Leandro I dont think you can. Normally you would use the Hangup() command with the hangupcause value that you wish to use. However there are no values which will be translated to a SIP/603. The closest would be Hangup(21) which would be 403 Forbidden -- _ -- 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] RTP from pcap file
Hello James, Il giorno 26/lug/2013 15:50, James Bensley jwbens...@gmail.com ha scritto: Howdy all, Does anyone know of a niffty CLI tool for Linux that can take a PCAP file that was created on a SIP PBX for example, and then dump the payload of the various RTP streams in there into seperate files so I can listen to them? I can go this graphically with Wireshark, but I'd like to script it for automation. Cheers, James. I personally use rtpbreak http://dallachiesa.com/code/rtpbreak/doc/rtpbreak_en.html For similar tasks Gianluca -- _ -- 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
