[asterisk-users] Delaying retry since we're currently running
Hi, I am making 200 call concurrently via call files. But i get these messages in asterisk logs: *Delaying retry since we're currently running* * * * * Also, in call files i have the following lines: *DelayedRetry: 28662 0 (1356701828)* *DelayedRetry: 28662 0 (1356702128) * *DelayedRetry: 28662 0 (1356702428) * * * * * I set MaxRetries: 0. I did not understand the problem, any idea? -- Necati DEMİR -- _ -- 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] Disappearing Call Files / Two threads dealing with my call files
Hello,
I noticed that when i move a call file to outgoing directory, two asterisk
threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1]
System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3]
System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or
extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/FAX_44731.call
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in
/var/spool/asterisk/outgoing/FAX_44731.call, deleting
As you see there are two thread dealing with my call file. Now let's
inspect the thread 18852.
]# grep \[18852\] /var/log/asterisk/full.2
[Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on
DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME: NUM: 90312xxx
[Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1]
System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:2]
SendFAX(DAHDI/i1/0312xxx-b08, /tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in
new stack
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel
'DAHDI/i1/0312xxx-b08' sending FAX:
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
/tmp/Qg90Ox5YGF5kYkJu.tif
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3]
System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel
'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
[Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup
'DAHDI/i1/0312xxx-b08'
[Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to
DAHDI/g0/0312xxx
It seems that the thread 18852 executes it normally but the thread 26842
deletes my call file. And when I inspected the asterisk log file, i saw
that the thread 26842 is deleting all my call files.
Here is my custom_extensions.conf file:
[asteriskgw_fax]
exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
exten = s,2,SendFAX(${FAXFILE},zdfs)
exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
And here is a sample of call file:
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
Set: ID=44884
Callerid: 90312xxx
Archive: Yes
--
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Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files
Should I use priority in call files? How the lack of priority causes this
problem?
On 29 November 2012 12:48, Matt Riddell (lists) li...@venturevoip.comwrote:
There's no priority in your call file.
Sent from my iPhone
On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:
Hello,
I noticed that when i move a call file to outgoing directory, two
asterisk threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:1] System(DAHDI/i1/0312xxx-b08, echo Set:
UNIQUEID=1354000990.39861 /var/spool/asterisk/outgoing/FAX_44731.call)
in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:3] System(DAHDI/i1/0312xxx-b08, echo Set:
FAXSTATUS=SUCCESS /var/spool/asterisk/outgoing/FAX_44731.call) in new
stack
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or
extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/FAX_44731.call
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in
/var/spool/asterisk/outgoing/FAX_44731.call, deleting
As you see there are two thread dealing with my call file. Now let's
inspect the thread 18852.
]# grep \[18852\] /var/log/asterisk/full.2
[Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on
DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME: NUM: 90312xxx
[Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:1] System(DAHDI/i1/0312xxx-b08, echo Set:
UNIQUEID=1354000990.39861 /var/spool/asterisk/outgoing/FAX_44731.call)
in new stack
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:2] SendFAX(DAHDI/i1/0312xxx-b08,
/tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in new stack
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel
'DAHDI/i1/0312xxx-b08' sending FAX:
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
/tmp/Qg90Ox5YGF5kYkJu.tif
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:3] System(DAHDI/i1/0312xxx-b08, echo Set:
FAXSTATUS=SUCCESS /var/spool/asterisk/outgoing/FAX_44731.call) in new
stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel
'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
[Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup
'DAHDI/i1/0312xxx-b08'
[Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to
DAHDI/g0/0312xxx
It seems that the thread 18852 executes it normally but the thread 26842
deletes my call file. And when I inspected the asterisk log file, i saw
that the thread 26842 is deleting all my call files.
Here is my custom_extensions.conf file:
[asteriskgw_fax]
exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
exten = s,2,SendFAX(${FAXFILE},zdfs)
exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
And here is a sample of call file:
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
Set: ID=44884
Callerid: 90312xxx
Archive: Yes
--
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Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files
Thanks, i will add priority and see the results.
On 29 November 2012 17:00, Danny Nicholas da...@debsinc.com wrote:
Priority is a required parameter. In your call file you are telling
Asterisk to
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Go to context asteriskgw_fax, extension s. Priority tells Asterisk where
to start in asteriskgw_fax. Since C would assume 0 and contexts start with
1, priority: 1 tells it to go to line 1. Another use for this would be to
tell Asterisk to start further down to skip a wait or something.
Sample:
[asteriskgw_fax]
Exten = s,1,answer()
Exten = s,n,wait(5)
Exten = s,n,playback(sending-fax)
** **
You could use priority 1 for DAHDI to compensate for PSTN delays and
priority 3 for SIP calls.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Necati Demir
*Sent:* Thursday, November 29, 2012 8:50 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Disappearing Call Files / Two threads
dealing with my call files
** **
** **
Should I use priority in call files? How the lack of priority causes this
problem?
** **
On 29 November 2012 12:48, Matt Riddell (lists) li...@venturevoip.com
wrote:
There's no priority in your call file.
Sent from my iPhone
On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:
Hello,
I noticed that when i move a call file to outgoing directory, two
asterisk threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:1] System(DAHDI/i1/0312xxx-b08, echo Set:
UNIQUEID=1354000990.39861 /var/spool/asterisk/outgoing/FAX_44731.call)
in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:3] System(DAHDI/i1/0312xxx-b08, echo Set:
FAXSTATUS=SUCCESS /var/spool/asterisk/outgoing/FAX_44731.call) in new
stack
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or
extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/FAX_44731.call
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in
/var/spool/asterisk/outgoing/FAX_44731.call, deleting
As you see there are two thread dealing with my call file. Now let's
inspect the thread 18852.
]# grep \[18852\] /var/log/asterisk/full.2
[Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on
DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME: NUM: 90312xxx
[Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:1] System(DAHDI/i1/0312xxx-b08, echo Set:
UNIQUEID=1354000990.39861 /var/spool/asterisk/outgoing/FAX_44731.call)
in new stack
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:2] SendFAX(DAHDI/i1/0312xxx-b08,
/tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in new stack
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel
'DAHDI/i1/0312xxx-b08' sending FAX:
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
/tmp/Qg90Ox5YGF5kYkJu.tif
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:3] System(DAHDI/i1/0312xxx-b08, echo Set:
FAXSTATUS=SUCCESS /var/spool/asterisk/outgoing/FAX_44731.call) in new
stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel
'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
[Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup
'DAHDI/i1/0312xxx-b08'
[Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to
DAHDI/g0/0312xxx
It seems that the thread 18852 executes it normally but the thread 26842
deletes my call file. And when I inspected the asterisk log file, i saw
that the thread 26842 is deleting all my call files.
Here is my custom_extensions.conf file:
[asteriskgw_fax]
exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
exten = s,2,SendFAX(${FAXFILE},zdfs)
exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
And here is a sample of call file:
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
Set: ID=44884
Callerid: 90312xxx
Archive: Yes
--
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[asterisk-users] DTMF Payload Settings
Hello, The service provider wants me to setup dtmfmode to rfc2833 and dtmf payload to 101. I can configure SIP trunk as dtmfmode=rfc2833 but how to configure payload? -- Necati DEMİR -- _ -- 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] OneAPI / ParlayX
Is there any open source web service implementaion like oneAPI or ParlayX to integrate with Asterisk? -- Necati DEMİR -- _ -- 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] when to use e1/t1 card?
Thanks for your answers. I think i still have questions. Now without a ISDN PRI card, i can connect to SIP server and do what i want. The card that i mentioned has a RJ45 port, so i think i still did not understand the advantage of it. On 21 June 2010 22:04, Necati Demir nde...@demir.web.tr wrote: This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEMİR --- -- Necati DEMİR http://demir.web.tr Pi Bilişim Teknolojileri http://www.pibilisim.com.tr -- -- _ -- 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] when to use e1/t1 card?
As you know, sending fax over ip is not very stable. So do these cards help to make this situation stable? On 22 June 2010 15:18, Zeeshan Zakaria zisha...@gmail.com wrote: If you are doing pure VoIP and no PRIs or PSTN, you don't need these cards. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-22 7:57 AM, Necati Demir nde...@demir.web.tr wrote: Thanks for your answers. I think i still have questions. Now without a ISDN PRI card, i can connect to SIP server and do what i want. The card that i mentioned has a RJ45 port, so i think i still did not understand the advantage of it. On 21 June 2010 22:04, Necati Demir nde...@demir.web.tr wrote: This is a really rookie quest... -- Necati DEMİR http://demir.web.tr Pi Bilişim Teknolojileri http://www.pibilisim.com.tr -- -- _ -- 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 -- Necati DEMİR http://demir.web.tr Pi Bilişim Teknolojileri http://www.pibilisim.com.tr -- -- _ -- 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] when to use e1/t1 card?
This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEMİR --- -- _ -- 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] how to get call duration
First thing which comes to mind is:
exten = h,1,Noop( Call duration was ${CDR(duration)} seconds)
exten = h,n,Hangup()
There is also a variable ${CDR(billsec)} which shows only the duration the
call was actually connected between two channels, however this may not match
with the duration of your provider.
Is there another way for getting a reliable call duration.
${CDR(duration)} show more duration than actual, and ${CDR(billsec)} shows
always 0.
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On 2010-06-03 9:35 AM, Necati Demir nde...@demir.web.tr wrote:
Hello,
I want to ask how to get call duration.
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ndemir ~ demir.web.tr
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[asterisk-users] how to run deadagi script after status: expired
I am using DeadAGI script and using this context.
exten = 10,1,Dial(SIP/${EXTEN})
exten = 10,n,Wait(1)
exten = 10,n,Playback(${PLAYFILE})
exten = 10,n,Wait(1)
exten = 10,n,Hangup()
exten = h,1,DeadAGI(script.agi)
DeadAGI script executes only if the call is successful. How to run DeadAGI
script in both status, successful and expired.
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[asterisk-users] how to get call duration
Hello, I want to ask how to get call duration. -- Necati DEMİR http://demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- 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] run script after completed
DeadAGI is executed if call is successful. I wanna ask how to execute agi script if the call is not only successful but also reject, busy, etc... 2010/5/5 Danny Nicholas da...@debsinc.com Regular AGI with SIGHUP detection? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mickael Monsieur *Sent:* Wednesday, May 05, 2010 12:36 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] run script after completed DeadAGI is deprecated in Asterisk 1.6.x ! 2010/4/9 Danny Nicholas da...@debsinc.com Do the call in a context and have the context run the script as a DeadAGI. [call_and_do] - exten = s,1,Dial… - exten = h,1,Deadagi(…) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Necati Demir *Sent:* Friday, April 09, 2010 7:34 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] run script after completed Hello, I am creating a call file with parameter Archive: yes. When it is completed it is moved to directory outgoing_done. It works. Now i want to execute a script when it is completed. Is there a parameter/configuration for this? -- Necati DEMİR http://blog.demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- 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 -- Necati DEMİR http://demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- 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] run script after completed
Hello, I am creating a call file with parameter Archive: yes. When it is completed it is moved to directory outgoing_done. It works. Now i want to execute a script when it is completed. Is there a parameter/configuration for this? -- Necati DEMİR http://blog.demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- 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] saving pressed keys
On 6 March 2010 23:21, Steve Edwards asterisk@sedwards.com wrote:
On Sat, 6 Mar 2010, Necati Demir wrote:
I created a dialplan. But now i want to save the keys that users press.
How can i do?
You need to be more specific in what you want to do.
Ok!
When a user selects menu from dialplan, i want to save it in a text file or
a database.
You can use the read() application to save user entry in a variable.
You can assign the ${EXTERN} channel variable to a variable of your
choosing at an appropriate priority in your dialplan.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Necati DEMİR
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http://friendfeed.com/ndemir
ndemir ~ demir.web.tr
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[asterisk-users] saving pressed keys
Hi, I created a dialplan. But now i want to save the keys that users press. How can i do? -- Necati DEMİR http://blog.demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- 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
