Re: [asterisk-users] Callback through extensions.conf?
On Mon, 14 Feb 2011 14:21:50 +0100, Gilles
wrote:
>Could it be that while we're in the dialplan after getting a call from
>the FXO, the FXO is just not available until after we exit the
>dialplan?
Made some progress: Asterisk can dial my cellphone if the callback
goes through an SIP trunk instead of reusing the FXO:
[from_fxo]
;Wait for RINGs to stop. Poor man's call progress
exten => s,1,Wait(15)
exten => s,n,Hangup
;to be totally positive FXO is available
exten => h,1,Wait(30)
exten => h,n,Dial(Local/start@callback)
[callback]
;BAD exten => start,n,Dial(Zap/1/${IPPI})
exten => start,n,Dial(SIP/vosp/${GSM})
exten => start,n,Hangup
--
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Re: [asterisk-users] Callback through extensions.conf?
On Mon, 14 Feb 2011 10:35:52 +0100, Gilles
wrote:
>If someone's already built a callback like the above using an FXO
>module, I would appreciate any feedback to try and solve this issue.
I learned more about Local channels, but this doesn't work either:
===
[from_fxo]
;Wait for RINGs to stop. Poor man's call progress
exten => s,1,Wait(15)
exten => s,n,Hangup
;to be totally positive FXO is available
exten => h,1,Wait(30)
exten => h,n,Dial(Local/start@callback)
[callback]
exten => start,1,Dial(Zap/1/${CELLPHONE})
exten => start,n,Hangup
===
...
-- Executing [start@callback:2] Dial("Local/start@callback-6c11,2",
"Zap/1/123456") in new stack
[Feb 14 13:10:26] WARNING[233]: app_dial.c:1183 dial_exec_full: Unable
to create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [start@callback:3] Hangup("Local/start@callback-6c11,2",
"") in new stack
== Spawn extension (callback, start, 3) exited non-zero on
'Local/start@callback-6c11,2'
===
Could it be that while we're in the dialplan after getting a call from
the FXO, the FXO is just not available until after we exit the
dialplan? In this case, I guess the only way to make a callback is to
record the call in a file/database, and create a CRON job so that the
call is made through an external application and AMI.
Thank you.
--
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Re: [asterisk-users] Callback through extensions.conf?
On Sat, 05 Feb 2011 12:07:28 +0100, Gilles
wrote:
>I've seen articles about Call files. Is this the easiest way to solve
>this problem?
I'm reading the 3rd edition of the "Asterisk: The Definitive Guide",
but since it's pretty big and there's no guarantee that the answer to
this issue is even in the book, I'd rather ask for feedback.
At this point, Asterisk is finally able to...
1. Detect a RING from my cellphone
2. Wait long enough to the line to stop ringing
3. Copy/modify/move a call file in the asterisk/outgoing directory
4. Make a callback to my cellphone
The problem is that Asterisk triggers an error when trying to make the
callback:
===
-- Attempting call on Zap/1/123456 for s@callback:1 (Retry 1)
[Jan 1 01:54:34] NOTICE[317]: channel.c:2863 __ast_request_and_dial:
Unable to request channel Zap/1/123456
[Jan 1 01:54:34] NOTICE[317]: pbx_spool.c:341 attempt_thread: Call
failed to go through, reason (0) Call Failure (not BUSY, and not
NO_ANSWER, maybe Circuit busy or down?)
===
Here's the callfile template:
===
/var/tmp> cat ZAP.call.orig
Context: callback
Extension: s
===
... and here's extensions.conf:
===
[from_fxo]
;how to reliably detect that line is now quiet?
exten => s,1,Wait(15)
exten => s,n,Hangup
exten => h,1,system(cp /var/tmp/ZAP.call.orig /var/tmp/ZAP.call)
exten => h,n,System(echo 'Channel: Zap/1/${CELLPHONE}' >>
/var/tmp/ZAP.call)
exten => h,n,system(mv /var/tmp/ZAP.call /var/tmp/asterisk/outgoing)
[callback]
exten => s,1,Verbose(In callback)
exten => s,n,Wait(10)
exten => s,n,Hangup
===
FWIW, I wondered if using a GoTo in "h" to jump to another context
would help, but it made no difference.
If someone's already built a callback like the above using an FXO
module, I would appreciate any feedback to try and solve this issue.
Thank you.
--
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Re: [asterisk-users] Callback through extensions.conf?
On Wed, 9 Feb 2011 12:33:00 -0600, Tilghman Lesher wrote: >Inotify for spoolfiles is supported starting in Asterisk 1.8. Thanks for the tip, but I'm stuck with a 1.4 because it must be patched to run on uClinux :-/ A possible explanation for this issue could be that Asterisk uses fork() to handle call files, which isn't available on uClinux. I don't have the courage to go through Asterisk's source code to understand how pool_pbx.c is used to manage call files. It looks like the simplest work-around is to use the "h" extension in the dial lplan to insert a job in Cron that will call "asterisk -rx "originate Zap/1 extension 123@context" to make a callback. I'll have to figure out how to save and retrieve the phone number from which the user called and expects to be called back. Thank you. -- _ -- 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] Callback through extensions.conf?
On Wednesday 09 February 2011 06:28:43 Gilles wrote: > On Wed, 09 Feb 2011 11:47:09 +0100, Gilles wrote: > >Unfortunately, I checked how the uClinux kernel was configured for > >compiling, and the inotify is indeed selected by default :-/ > > Greping the Asterisk source code for "inotify" only returned a couple > of hits, in binaries (./main/logger.o and ./main/asterisk). So I guess > Asterisk doesn't use the inotify Linux feature. Inotify for spoolfiles is supported starting in Asterisk 1.8. -- Tilghman -- _ -- 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] Callback through extensions.conf?
On Wed, 09 Feb 2011 11:47:09 +0100, Gilles wrote: >Unfortunately, I checked how the uClinux kernel was configured for >compiling, and the inotify is indeed selected by default :-/ Greping the Asterisk source code for "inotify" only returned a couple of hits, in binaries (./main/logger.o and ./main/asterisk). So I guess Asterisk doesn't use the inotify Linux feature. Could it be that a Yaffs filesystem on a NAND flash memory handles files in such a way that an application might not notice that a file has been created? As a work-around, is there an alternative way to schedule a call from within Asterisk? The appliance only offers Asterisk 1.4.x, and "originate" was apparently introduced in 1.6.2 www.voip-info.org/wiki/view/Asterisk+cmd+Originate Thank you. -- _ -- 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] Callback through extensions.conf?
On Wed, 9 Feb 2011 00:01:49 -0600, Sherwood McGowan wrote: >Nice! That was some good reading! Unfortunately, I checked how the uClinux kernel was configured for compiling, and the inotify is indeed selected by default :-/ Linux Kernel Configuration File systems [*] Inotify file change notification support [*] Inotify support for userspace [*] Dnotify support Miscellaneous filesystems <*> YAFFS2 file system support -- _ -- 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] Callback through extensions.conf?
Gilles, Nice! That was some good reading! On Tue, Feb 8, 2011 at 6:01 PM, Gilles wrote: > Interesting... > > http://en.wikipedia.org/wiki/Inotify > > http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/?action=ForumBrowse&forum_id=39&_forum_action=ForumMessageBrowse&thread_id=33403 > > > -- > _ > -- 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] Callback through extensions.conf?
Interesting... http://en.wikipedia.org/wiki/Inotify http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/?action=ForumBrowse&forum_id=39&_forum_action=ForumMessageBrowse&thread_id=33403 -- _ -- 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] Callback through extensions.conf?
On Tue, 08 Feb 2011 14:23:12 +0100, Gilles wrote: >However, by chance, I happened on a pattern: The callfile is handled >only if I... >1. Stop Asterisk through its init.d script >2. Mv the callfile >3. Start Asterisk through its init.d script It also works if I launch Asterisk manually with eg. "asterisk -ddc". -- _ -- 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] Callback through extensions.conf?
Thanks much everyone for the great help. I did go through the last suggestions about the callfile (no CRLF issue, permissions are 644 and file owned by root, starting asterisk through strace, etc.), but none helped. However, by chance, I happened on a pattern: The callfile is handled only if I... 1. Stop Asterisk through its init.d script 2. Mv the callfile 3. Start Asterisk through its init.d script Here are the commands I run, the little script I use to move the callfile, and what it contains: === /var/tmp> /etc/init.d/asterisk start /var/tmp> ./mvSIP.bash /var/tmp> /etc/init.d/asterisk stop /var/tmp> /etc/init.d/asterisk start === /var/tmp> cat mvSIP.bash #!/bin/sh cp callfileSIP.call.backup callfileSIP.call mv callfileSIP.call /var/spool/asterisk/outgoing === Channel: SIP/xlite Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Archive: yes === Once the callfile has been handled, it is moved from /var/spool/asterisk/outgoing to ./outgoing_done and has a couple of lines appended: === ... StartRetry: 2306 1 (1297171283) Status: Completed === I don't know if it means anything, but here's the output of "mount" on this appliance (the root filesystem uses yaffs for persistence): === /var/tmp> mount rootfs on / type rootfs (rw) /dev/root on / type yaffs (rw) proc on /proc type proc (rw) ramfs on /var/tmp type ramfs (rw) sysfs on /sys type sysfs (rw) devpts on /dev/pts type devpts (rw) usbfs on /proc/bus/usb type usbfs (rw) securityfs on /sys/kernel/security type securityfs (rw) === Thank you. -- _ -- 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] Callback through extensions.conf?
> > > > In my (1.4.X) experience, the file just stays in > /var/spool/asterisk/outgoing and gets “little tags” added until you get the > problem resolved or delete the file. > > > That is absolutely true if the file is not processed. I guess he can again do a "ls -la" in that folder to check permissions for the file not processed. -Bruce -- _ -- 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] Callback through extensions.conf?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Monday, February 07, 2011 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Callback through extensions.conf? *** *** If you are sure that permissions are not the problem and you have archive set to yes then you can browse the /var/spoo/asterisk/outgoing_done folder to see if the call file is transferred there or not. The file should contain some info to help you and it's existence also means that somehow you are not seeing the call through your CLI as it's processed. However I doubt this is happening. -Bruce The archived file, if I recall correctly, will be appended with "failed" or something similar...I'd check the wiki In my (1.4.X) experience, the file just stays in /var/spool/asterisk/outgoing and gets "little tags" added until you get the problem resolved or delete the file. -- _ -- 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] Callback through extensions.conf?
*** *** > If you are sure that permissions are not the problem and you have archive > set to yes then you can browse the */var/spoo/asterisk/outgoing_done*folder > to see if the call file is transferred there or not. The file should > contain some info to help you and it's existence also means that somehow you > are not seeing the call through your CLI as it's processed. However I doubt > this is happening. > > -Bruce > The archived file, if I recall correctly, will be appended with "failed" or something similar...I'd check the wiki -- _ -- 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] Callback through extensions.conf?
Asterisk runs as root but what about the bash script or the php file that creates the file? Maybe comment the "mv" command and check the file permissions by *"ls -la call-filename.call"* to be sure. *chown root.root call-filename* (if root is really the user running Asterisk) and then the "mv" command line should do the trick. If you are sure that permissions are not the problem and you have archive set to yes then you can browse the */var/spoo/asterisk/outgoing_done* folder to see if the call file is transferred there or not. The file should contain some info to help you and it's existence also means that somehow you are not seeing the call through your CLI as it's processed. However I doubt this is happening. -Bruce On Mon, Feb 7, 2011 at 11:46 AM, Gilles wrote: > On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards > wrote: > >> sudo /usr/sbin/asterisk -d -d -d -n -v -v -v > > > >Oops. A '-c' should be in there :) > > Thanks Steve for the help. > > I launched * with "asterisk -d -d -d -n -v -v -v -c", and ran "module > show" to check that pbx_spool.so is loaded: > = > *CLI> module show like pbx_spool.so > Module Description Use Count > pbx_spool.so Outgoing Spool Support 0 > 1 modules loaded > = > > Next, I moved the following callfile to /var/spool/asterisk/outgoing: > = > #callfileSIP.call > Channel: SIP/xlite > Context: callback-dialtone-auth > Extension: s > Priority: 1 > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > = > > Asterisk runs as root, and owns this file as well. > > Unfortunately, nothing shows up in the console, the xlite extension > isn't called, even after waiting for a few minutes. > > Could it be that pbx_spool.so isn't really loaded, or is Asterisk > somehow configured to ignore callfiles? > > Thank you. > > -- _ -- 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] Callback through extensions.conf?
On 02/07/2011 11:46 AM, Gilles wrote: Asterisk runs as root, and owns this file as well. Have you tried setting the permissions of this file to world readable, to ensure that any user can read it and eliminate potential permissions problems? Worth a shot. While you're at it, output from the ps command that shows the output, command line, and header for asterisk will help, too. Tom -- _ -- 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] Callback through extensions.conf?
Real quick, please respond to my question about where the callfile ends up after a few minutes, as well as the modification time and the permissions on the file ;-) These are good bits to know On Mon, Feb 7, 2011 at 10:46 AM, Gilles wrote: > On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards > wrote: > >> sudo /usr/sbin/asterisk -d -d -d -n -v -v -v > > > >Oops. A '-c' should be in there :) > > Thanks Steve for the help. > > I launched * with "asterisk -d -d -d -n -v -v -v -c", and ran "module > show" to check that pbx_spool.so is loaded: > = > *CLI> module show like pbx_spool.so > Module Description Use Count > pbx_spool.so Outgoing Spool Support 0 > 1 modules loaded > = > > Next, I moved the following callfile to /var/spool/asterisk/outgoing: > = > #callfileSIP.call > Channel: SIP/xlite > Context: callback-dialtone-auth > Extension: s > Priority: 1 > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > = > > Asterisk runs as root, and owns this file as well. > > Unfortunately, nothing shows up in the console, the xlite extension > isn't called, even after waiting for a few minutes. > > Could it be that pbx_spool.so isn't really loaded, or is Asterisk > somehow configured to ignore callfiles? > > Thank you. > > > -- > _ > -- 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] Callback through extensions.conf?
On Mon, Feb 7, 2011 at 10:46 AM, Gilles wrote: > = > #callfileSIP.call > Channel: SIP/xlite > Context: callback-dialtone-auth > Extension: s > Priority: 1 > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > = > Just a thought... Did you originally generate this callfile on the linux box itself, or did you create it first on a windows box and then move it over to the linux box? There's something to do with the way Windows text editor does line breaks that linux doesn't like, and asterisk expects line breaks in certain spots for it to work properly. If that isn't the case - maybe look at using a local channel instead of "SIP/xlite" to setup the call? -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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] Callback through extensions.conf?
On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards wrote: >> sudo /usr/sbin/asterisk -d -d -d -n -v -v -v > >Oops. A '-c' should be in there :) Thanks Steve for the help. I launched * with "asterisk -d -d -d -n -v -v -v -c", and ran "module show" to check that pbx_spool.so is loaded: = *CLI> module show like pbx_spool.so Module Description Use Count pbx_spool.so Outgoing Spool Support 0 1 modules loaded = Next, I moved the following callfile to /var/spool/asterisk/outgoing: = #callfileSIP.call Channel: SIP/xlite Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 = Asterisk runs as root, and owns this file as well. Unfortunately, nothing shows up in the console, the xlite extension isn't called, even after waiting for a few minutes. Could it be that pbx_spool.so isn't really loaded, or is Asterisk somehow configured to ignore callfiles? Thank you. -- _ -- 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] Callback through extensions.conf?
On Mon, 7 Feb 2011, Steve Edwards wrote: sudo /usr/sbin/asterisk -d -d -d -n -v -v -v Oops. A '-c' should be in there :) -- 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
Re: [asterisk-users] Callback through extensions.conf?
On Mon, 7 Feb 2011 04:06:52 -0600, Sherwood McGowan wrote: ok, first of all, it can take a little while for those spooled callfiles to be executed in Asterisk... On Mon, 7 Feb 2011, Gilles wrote: Thanks for your help. The same callfile works fine in Ubuntu, but not at that appliance. Since I can dial through the FXO, it doesn't seem to be a Zaptel issue either. I'll investigate further, and find a work-around if the appliance just doesn't support this feature for some reason. Like maybe pbx_spool.so not being loaded? Bump up the logging in logger.conf, verbose and debugging in the CLI and see if you can get any clues. Another useful exercise is to start Asterisk like: script startup-log sudo /usr/sbin/asterisk -d -d -d -n -v -v -v exit and then read every line of startup-log. -- 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
Re: [asterisk-users] Callback through extensions.conf?
On Mon, 7 Feb 2011 04:06:52 -0600, Sherwood McGowan wrote: >ok, first of all, it can take a little while for those spooled callfiles to >be executed in Asterisk... Thanks for your help. The same callfile works fine in Ubuntu, but not at that appliance. Since I can dial through the FXO, it doesn't seem to be a Zaptel issue either. I'll investigate further, and find a work-around if the appliance just doesn't support this feature for some reason. Thank you. -- _ -- 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] Callback through extensions.conf?
> > ... but Asterisk does nothing, altough "show modules" says that > pbx_spool.so is loaded. Weird :-/ > > FWIW, Asterisk runs as root, and root owns callfile.call. > > Maybe it's the uClinux or the Asterisk I'm using that's configured in > such a way that callfiles don't work as planned. > > Apparently, there's no other way than callfiles to have Asterisk dial > out from the dialplan? > > ok, first of all, it can take a little while for those spooled callfiles to be executed in Asterisk... Also, have you READ the callfile documentation? Maybe you've thought to check the modification times for the files? Are the files you've created still in the outgoing directory or are they somewhere else? If they're still in the outgoing dir, then something is causing the call files to not be executed Regarding your second inquiryare you serious? I'm not trying to be mean, but WOW... There are several ways to initate a call from the dialplan (as well as other places) The Dial command is pretty handy for thatjust an example... I'm going to have to stop assisting you at this point, it's apaprent you've not done all your homework...To not know the 'h' extension was where you could put dialplan commands to be executed after the calling channel hangs up...that's in Asterisk 101...as is the Dial command, and a few other ways of initiating calls...Hell, I'd venture to say that if I was taking a class on Asterisk configuration, callfiles would be near the END of the section on initiating calls, because they're quite often NOT used due to the presence of easier methods... Cheers bud -- _ -- 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] Callback through extensions.conf?
On Mon, 7 Feb 2011 02:59:09 -0600, Sherwood McGowan
wrote:
>That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your
>DAHDI/Zap setup... Barring something in your configuration that I don't know
>about, there's no reason that the system should just hang up the call during
>the Wait() command...
The /etc/zaptel.conf and /etc/asterisk/zapata.conf are pretty basic:
===
> cat /etc/zaptel.conf
loadzone = fr
defaultzone = fr
fxsks=1
===
> cat /etc/asterisk/zapata.conf
[trunkgroups]
[channels]
context=from_fxo
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=yes
busydetect=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
signalling = fxs_ks
channel => 1
===
>I just had a thought thoughAre you, perhaps, hanging your mobile (or
>whatever) phone up after dialing into the system to trigger that context?
>The reason I ask is that would make this suddenly seem more clear
Yes, that's exactly the idea: From my cellphone, I ring Asterisk just
to notify that I want to be called back. Asterisk waits until the line
stops ringing (or 10s, if there's no better way), then it dials back.
>; Note From Sherwood McGowan
>; By Changing the exten => s to exten => h in the section below, we
>guarantee that Asterisk will execute the code IF THE CALL IS ENDED (like in
>the examples given on the mailing list)
By moving the call file part in the "h" extension, the code is
executed. That's better :-)
Problem is... Asterisk doesn't ring the phone, nothing happens:
===
-- Starting simple switch on 'Zap/1-1'
-- Executing [s@from_fxo:1] Wait("Zap/1-1", "1") in new stack
-- Executing [s@from_fxo:2] Set("Zap/1-1", "SOURCE_CIDNUMBER=5551234")
in new stack
-- Executing [s@from_fxo:3] Set("Zap/1-1", "SOURCE_CIDNAME=") in new
stack
-- Executing [s@from_fxo:4] NoOp("Zap/1-1", "Call from - 5551234") in
new stack
-- Executing [s@from_fxo:5] NoOp("Zap/1-1", "CID OK") in new stack
-- Executing [s@from_fxo:6] Wait("Zap/1-1", "10") in new stack
== Spawn extension (from_fxo, s, 6) exited non-zero on 'Zap/1-1'
-- Executing [h@from_fxo:1] NoOp("Zap/1-1", "Before cp") in new stack
-- Executing [h@from_fxo:2] System("Zap/1-1", "cp
/var/spool/asterisk/skelett.call /var/tmp/skelett.call") in new stack
-- Executing [h@from_fxo:3] NoOp("Zap/1-1", "Before echo") in new
stack
-- Executing [h@from_fxo:4] System("Zap/1-1", "echo "Channel:
ZAP/1/5551234" >> /var/tmp/skelett.call") in new stack
-- Executing [h@from_fxo:5] NoOp("Zap/1-1", "Before mv") in new stack
-- Executing [h@from_fxo:6] System("Zap/1-1", "mv
/var/tmp/skelett.call /var/spool/asterisk/outgoing/") in new stack
-- Hungup 'Zap/1-1'
===
To simplify further, I tried building a callfile.call manually and
moving it to /var/spool/asterisk/outgoing/ ...
==
Channel: ZAP/1/5551234
Context: callback-dialtone-auth
Extension: s
Priority: 1
MaxRetries: 2
RetryTime: 60
WaitTime: 30
==
... but Asterisk does nothing, altough "show modules" says that
pbx_spool.so is loaded. Weird :-/
FWIW, Asterisk runs as root, and root owns callfile.call.
Maybe it's the uClinux or the Asterisk I'm using that's configured in
such a way that callfiles don't work as planned.
Apparently, there's no other way than callfiles to have Asterisk dial
out from the dialplan?
Thank you.
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Re: [asterisk-users] Callback through extensions.conf?
On Mon, Feb 7, 2011 at 2:22 AM, Gilles wrote:
> Lowering it to 5 seconds makes no difference. I also tried adding a
> Hangup before Wait but then the script ends before Wait.
>
>
That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your
DAHDI/Zap setup... Barring something in your configuration that I don't know
about, there's no reason that the system should just hang up the call during
the Wait() command...
> Could it be that it's just not possible to reuse a channel to dial out
> after it's been used to receive a call, even though it was just for a
> ring?
>
>
Well, first of all, the channel (in the example dialplan and logs you posted
earlier) wouldn't even be dialing a call, it would just be responsible for
the generation of the callfile that would then cause Asterisk to spawn a
call via whatever Channel you specified
I just had a thought thoughAre you, perhaps, hanging your mobile (or
whatever) phone up after dialing into the system to trigger that context?
The reason I ask is that would make this suddenly seem more clear
Basically, try this modified version of the dialplan code:
[from_fxo]
exten => s,1,Wait(2)
exten => s,n,Set(SOURCE_CIDNUMBER=${
CALLERID(num)})
exten => s,n,Set(SOURCE_CIDNAME=${CALLERID(name)})
exten => s,n,NoOp(Call from ${SOURCE_CIDNAME} - ${SOURCE_CIDNUMBER})
exten => s,n,GotoIf($["${SOURCE_CIDNUMBER}" = ${GSM}]?goodcid:badcid)
exten => s,n(goodcid),NoOp(CID OK)
;how to reliably detect that line is now quiet?
exten => s,n,Wait(10)
; Note From Sherwood McGowan
; By Changing the exten => s to exten => h in the section below, we
guarantee that Asterisk will execute the code IF THE CALL IS ENDED (like in
the examples given on the mailing list)
; Good Luck!
exten => h,1,NoOp(Before cp)
exten => h,n,system(cp /var/spool/asterisk/skelett.call
/var/tmp/skelett.call)
exten => h,n,NoOp(Before echo)
exten => h,n,system(echo "Channel: ZAP/1/${IPPI}" >> /var/tmp/skelett.call)
exten => h,n,NoOp(Before mv)
exten => h,n,system(mv /var/tmp/skelett.call /var/spool/asterisk/outgoing/)
Cheers!
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Re: [asterisk-users] Callback through extensions.conf?
On Sun, 6 Feb 2011 16:27:33 -0600, Sherwood McGowan wrote: >Have you tried playing with the length of the wait? Even if you technically >need 10 seconds, you could try a lower amount to see if the other priorities >in that context execute... Lowering it to 5 seconds makes no difference. I also tried adding a Hangup before Wait but then the script ends before Wait. Could it be that it's just not possible to reuse a channel to dial out after it's been used to receive a call, even though it was just for a ring? Thank you. -- _ -- 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] Callback through extensions.conf?
>From what I can see from your log and the previously supplied snippet of your dialplan, yes it looks like it's hanging up for a reason other than a dialplan issue. It definitely doesn't appear to be an issue in the callfile, since it never gets to the commands that interact with it Have you tried playing with the length of the wait? Even if you technically need 10 seconds, you could try a lower amount to see if the other priorities in that context execute... Cheers, SKM -- _ -- 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] Callback through extensions.conf?
On Sun, 6 Feb 2011 10:10:06 -0600, Sherwood McGowan
wrote:
>Can you give me/us the output of the full log (verbose set to 5 please)?
>Once I hve that, I can probably help you quite quickly, I work with callfile
>generation often without problem
Here's the output with the console launched with "asterisk
-ddr":
===
-- Starting simple switch on 'Zap/1-1'
-- Executing [s@from_fxo:1] Wait("Zap/1-1", "2") in new stack
-- Executing [s@from_fxo:2] Set("Zap/1-1",
"SOURCE_CIDNUMBER=555-1234") in new stack
-- Executing [s@from_fxo:3] Set("Zap/1-1", "SOURCE_CIDNAME=") in new
stack
-- Executing [s@from_fxo:4] NoOp("Zap/1-1", "Call from - 555-1234")
in new stack
-- Executing [s@from_fxo:5] GotoIf("Zap/1-1", "1?goodcid:badcid") in
new stack
-- Goto (from_fxo,s,6)
-- Executing [s@from_fxo:6] NoOp("Zap/1-1", "CID OK") in new stack
-- Executing [s@from_fxo:7] Wait("Zap/1-1", "10") in new stack
== Spawn extension (from_fxo, s, 7) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
===
Could it some time-out issue, not a bug in my call file?
Thank you.
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Re: [asterisk-users] Callback through extensions.conf?
On Sun, 06 Feb 2011 12:05:25 -0500, John Novack wrote: >Later Dahdi code may do what you want IF, and only if, your provider >signals when the call is answered. Thanks for the information. Telling Asterisk to wait long enough after I dialed in should be enough. -- _ -- 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] Callback through extensions.conf?
Gilles wrote: On Sat, 05 Feb 2011 16:27:35 -0500, Paul Belanger wrote: Easy enough. I would suggest using Disa() for added security. Thanks for the tip, but then, I would be charged for the call from my cellphone to Asterisk. I guess it's not a big risk to assume a call with my CID number is indeed from my cellphone, so I can just ring Asterisk a couple of times before hanging up and being called back for free. A couple of articles on dialing out through an FXO port say that Asterisk considers the call answered when it gets a dialtone, not when the remote number goes off-hook. Not quite. It considers the call answered when dialing is complete Has this been solved in Asterisk 1.4.20 + Zaptel 1.4.3-1? Are there work-arounds? Not really solvable, unless you move to a digital circuit. Few providers provide answer supervision , at least in the US. Later Dahdi code may do what you want IF, and only if, your provider signals when the call is answered. John Novack Thank you. -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Callback through extensions.conf?
Can you give me/us the output of the full log (verbose set to 5 please)?
Once I hve that, I can probably help you quite quickly, I work with callfile
generation often without problem
First, you're trying to copy
On Sun, Feb 6, 2011 at 9:36 AM, Gilles wrote:
> On Sat, 05 Feb 2011 12:07:28 +0100, Gilles
> wrote:
> >I've seen articles about Call files. Is this the easiest way to solve
> >this problem?
>
> For some reason, Asterisk executes Wait(10), but then hangs up without
> running the rest of the commands (cp, echo, mv):
>
> ==
> [from_fxo]
> exten => s,1,Wait(2)
>
> exten => s,n,Set(SOURCE_CIDNUMBER=${CALLERID(num)})
> exten => s,n,Set(SOURCE_CIDNAME=${CALLERID(name)})
> exten => s,n,NoOp(Call from ${SOURCE_CIDNAME} - ${SOURCE_CIDNUMBER})
>
> exten => s,n,GotoIf($["${SOURCE_CIDNUMBER}" = ${GSM}]?goodcid:badcid)
>
> exten => s,n(goodcid),NoOp(CID OK)
> ;how to reliably detect that line is now quiet?
> exten => s,n,Wait(10)
>
> exten => s,n,NoOp(Before cp)
> exten => s,n,system(cp /var/spool/asterisk/skelett.call
> /var/tmp/skelett.call)
> exten => s,n,NoOp(Before echo)
> exten => s,n,system(echo 'Channel: ZAP/1/${IPPI}' >>
> /var/tmp/skelett.call)
> exten => s,n,NoOp(Before mv)
> exten => s,n,system(mv /var/tmp/skelett.call
> /var/spool/asterisk/outgoing)
>
> ;exten => s,n,Hangup
>
> ;Context set in call file
> [callback-dialtone-auth]
> exten => s,1,NoOp(Call from ${CALLERIS(num)})
> ==
>
> FWIW, "show modules" says that pbx_spool.so is loaded.
>
> Has someone experienced this?
>
> Thank you.
>
>
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Re: [asterisk-users] Callback through extensions.conf?
On Sat, 05 Feb 2011 12:07:28 +0100, Gilles
wrote:
>I've seen articles about Call files. Is this the easiest way to solve
>this problem?
For some reason, Asterisk executes Wait(10), but then hangs up without
running the rest of the commands (cp, echo, mv):
==
[from_fxo]
exten => s,1,Wait(2)
exten => s,n,Set(SOURCE_CIDNUMBER=${CALLERID(num)})
exten => s,n,Set(SOURCE_CIDNAME=${CALLERID(name)})
exten => s,n,NoOp(Call from ${SOURCE_CIDNAME} - ${SOURCE_CIDNUMBER})
exten => s,n,GotoIf($["${SOURCE_CIDNUMBER}" = ${GSM}]?goodcid:badcid)
exten => s,n(goodcid),NoOp(CID OK)
;how to reliably detect that line is now quiet?
exten => s,n,Wait(10)
exten => s,n,NoOp(Before cp)
exten => s,n,system(cp /var/spool/asterisk/skelett.call
/var/tmp/skelett.call)
exten => s,n,NoOp(Before echo)
exten => s,n,system(echo 'Channel: ZAP/1/${IPPI}' >>
/var/tmp/skelett.call)
exten => s,n,NoOp(Before mv)
exten => s,n,system(mv /var/tmp/skelett.call
/var/spool/asterisk/outgoing)
;exten => s,n,Hangup
;Context set in call file
[callback-dialtone-auth]
exten => s,1,NoOp(Call from ${CALLERIS(num)})
==
FWIW, "show modules" says that pbx_spool.so is loaded.
Has someone experienced this?
Thank you.
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Re: [asterisk-users] Callback through extensions.conf?
On Sat, 05 Feb 2011 16:27:35 -0500, Paul Belanger wrote: >Easy enough. I would suggest using Disa() for added security. Thanks for the tip, but then, I would be charged for the call from my cellphone to Asterisk. I guess it's not a big risk to assume a call with my CID number is indeed from my cellphone, so I can just ring Asterisk a couple of times before hanging up and being called back for free. A couple of articles on dialing out through an FXO port say that Asterisk considers the call answered when it gets a dialtone, not when the remote number goes off-hook. Has this been solved in Asterisk 1.4.20 + Zaptel 1.4.3-1? Are there work-arounds? Thank you. -- _ -- 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] Callback through extensions.conf?
On Sun, 6 Feb 2011 10:06:42 +0330, Pezhman Lali wrote: >a2billing also provided call_back daemon, try it Thanks for the tip, but A2billing requires a LAMP server, which won't fit on an appliance. -- _ -- 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] Callback through extensions.conf?
On Sat, Feb 5, 2011 at 3:27 PM, Paul Belanger wrote: > On 11-02-05 06:07 AM, Gilles wrote: > > 2. Asterisk waits until I hang up, calls me back, and prompts me for > > the number I wish to call > > > use exten => h to start a local channel, wait x seconds, dial your cell > phone. > Or you could generate a callfile to initiate the call and use the Context, Extension, and Priority configuration items to have the call connected to your menu (the part where you ask for the number to call, etc) once you pick up the callback :) Just thought I'd throw my $0.2 in there Cheers guys -- _ -- 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] Callback through extensions.conf?
Dear a2billing also provided call_back daemon, try it best On Sun, Feb 6, 2011 at 12:57 AM, Paul Belanger wrote: > On 11-02-05 06:07 AM, Gilles wrote: > > I'd like to configure Asterisk so that... > > 1. I ring it from my cellphone with CID number displayed, just to > > notify Asterisk that I wish to make a call > > > Easy enough. I would suggest using Disa() for added security. > > > 2. Asterisk waits until I hang up, calls me back, and prompts me for > > the number I wish to call > > > use exten => h to start a local channel, wait x seconds, dial your cell > phone. > > > 3. Asterisk puts me on hold through Flash(), which is apparently the > > equivalent of hitting the R key on European handsets > > > Straight forward. > > > 4. Asterisk calls the number, and once the remote party has answered, > > bridges the two channels > > > Same, straight forward > > > Ideally, I'd like to do this entirely through extensions.conf, and > > avoid callling an AGI script or having to add Konference: This is an > > appliance, so RAM isn't plentiful, it runs uClinux instead of > > run-of-the-mill Linux, and I would like to avoid having to patch > > Asterisk. > > > Should be no more then 10-15 lines within your extensions.conf > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://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 > -- _ -- 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] Callback through extensions.conf?
On 11-02-05 06:07 AM, Gilles wrote: > I'd like to configure Asterisk so that... > 1. I ring it from my cellphone with CID number displayed, just to > notify Asterisk that I wish to make a call > Easy enough. I would suggest using Disa() for added security. > 2. Asterisk waits until I hang up, calls me back, and prompts me for > the number I wish to call > use exten => h to start a local channel, wait x seconds, dial your cell phone. > 3. Asterisk puts me on hold through Flash(), which is apparently the > equivalent of hitting the R key on European handsets > Straight forward. > 4. Asterisk calls the number, and once the remote party has answered, > bridges the two channels > Same, straight forward > Ideally, I'd like to do this entirely through extensions.conf, and > avoid callling an AGI script or having to add Konference: This is an > appliance, so RAM isn't plentiful, it runs uClinux instead of > run-of-the-mill Linux, and I would like to avoid having to patch > Asterisk. > Should be no more then 10-15 lines within your extensions.conf -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://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
[asterisk-users] Callback through extensions.conf?
Hello I'd like to configure Asterisk so that... 1. I ring it from my cellphone with CID number displayed, just to notify Asterisk that I wish to make a call 2. Asterisk waits until I hang up, calls me back, and prompts me for the number I wish to call 3. Asterisk puts me on hold through Flash(), which is apparently the equivalent of hitting the R key on European handsets 4. Asterisk calls the number, and once the remote party has answered, bridges the two channels Ideally, I'd like to do this entirely through extensions.conf, and avoid callling an AGI script or having to add Konference: This is an appliance, so RAM isn't plentiful, it runs uClinux instead of run-of-the-mill Linux, and I would like to avoid having to patch Asterisk. I've seen articles about Call files. Is this the easiest way to solve this problem? www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Thank you for any help. -- _ -- 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
