Re: [asterisk-users] Call File - CPU spikes

[Lenz Emilitri]

If you are on 13 it would likely be easier to use ARI directly?
l.


2016-05-11 22:52 GMT+02:00 Bryant Zimmerman :
> I am working on a project that we are seeing a 100% CPU spike when we move
> 50 calls files to the folder.
>
> We are running pjsip and asterisk 13..It holds the spike for several minutes
> Are there any tunable that may help with this?
>
>
> Thanks
> Bryant
>
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Re: [asterisk-users] Call File - CPU spikes

[Bryant Zimmerman]

I am working on a project that we are seeing a 100% CPU spike when we move 
50 calls files to the folder.
  
 We are running pjsip and asterisk 13..It holds the spike for several 
minutes Are there any tunable that may help with this?
  

Thanks
Bryant

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[asterisk-users] Call file problem, DelayedRetry/retrying spite MaxRetries: 0

[Mikael Fredin]

I am using Realtime extensions as well, in case that would matter.

Following problem arises from time to time, a call will successfully
terminate:

[May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing
[t@project_init:1] Hangup(SIP/peer-2-2f7e, )
[May 14 14:31:41] VERBOSE[3274] pbx.c:   == Spawn extension (project_init,
t, 1) exited non-zero on 'SIP/peer-2-2f7e'

bye message,  Really destroying SIP dialog, etc

This is the call file:

Channel: SIP/peer-2/00numberhere
CallerID:  +calleridhere
Extension: 123
SetVar: someid=123
Context: setup
WaitTime: 30
MaxRetries: 0
RetryTime: 300
Account: 123
Priority: 1


Some time after the call has hung up, the call file is still there and this
is appended to the file:
StartRetry: 20354 1 (1400070906) (My note: Wed May 14 14:35:06 CEST 2014)

DelayedRetry: 20354 0 (1400070906) same time...

DelayedRetry: 20354 0 (1400071206) five minutes...

DelayedRetry: 20354 0 (1400071506) and so on...

DelayedRetry: 20354 0 (1400071806) never deleting this file

DelayedRetry: 20354 0 (1400072106) are we?

DelayedRetry: 20354 0 (1400072406) nope

DelayedRetry: 20354 0 (1400072706) waiting for someone

DelayedRetry: 20354 0 (1400073006) to do manual work




Asterisk log:
[May 14 14:35:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
currently running '/var/spool/asterisk/outgoing/callfile'
[May 14 14:40:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
currently running '/var/spool/asterisk/outgoing/callfile'
[May 14 14:45:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
currently running '/var/spool/asterisk/outgoing/callfile'
[May 14 14:50:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
currently running '/var/spool/asterisk/outgoing/callfile'
[May 14 14:55:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
currently running '/var/spool/asterisk/outgoing/callfile'
[May 14 15:00:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
currently running '/var/spool/asterisk/outgoing/callfile'
[May 14 15:05:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
currently running '/var/spool/asterisk/outgoing/callfile'
[May 14 15:10:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
currently running '/var/spool/asterisk/outgoing/callfile'




Asterisk code:

   if (o-retries = o-maxretries) {
now += o-retrytime;
if (o-callingpid  (o-callingpid == ast_mainpid)) {
safe_append(o, time(NULL), DelayedRetry);
ast_log(LOG_DEBUG, Delaying retry since we're
currently running '%s'\n, o-fn);
free_outgoing(o);
} else {
/* Increment retries */
o-retries++;
/* If someone else was calling, they're presumably
gone now
   so abort their retry and continue as we were...
*/
if (o-callingpid)
safe_append(o, time(NULL), AbortRetry);

safe_append(o, now, StartRetry);
launch_service(o);
}
return now;
}




Sure, I could just disable the retry check and add :
if (FALSE) {
And it will always expire should this occur...

But I'm not sure if this is a good idea or not, and it would be nice not
having to do that on every upgrade.


Anyone have experience with what's going on?

The file can be written to, since safe_append seems to be able to write to
the file.

This only happens once in a while, which makes it hard to track down.
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Re: [asterisk-users] Call file problem, DelayedRetry/retrying spite MaxRetries: 0

[Mikael Fredin]

Forgot to mention some important things.

Asterisk versions I have tried this one and got the error: 1.8.16 and
1.8.27.

 core show channels will show 0-10 channels when this happens (the true
count), but the core show calls and the call counter for active calls
after core show channels will show a very high amount of calls
(150-250+), this during times when we'd not expect to have close to that
amount.

Googling a bit gives people with the same problem but no solutions, one
with asterisk 1.4 who also reports weird call/channel counts.






On 15 May 2014 13:34, Mikael Fredin mik...@wiraya.com wrote:

 I am using Realtime extensions as well, in case that would matter.

 Following problem arises from time to time, a call will successfully
 terminate:

 [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing
 [t@project_init:1] Hangup(SIP/peer-2-2f7e, )
 [May 14 14:31:41] VERBOSE[3274] pbx.c:   == Spawn extension (project_init,
 t, 1) exited non-zero on 'SIP/peer-2-2f7e'

 bye message,  Really destroying SIP dialog, etc

 This is the call file:

 Channel: SIP/peer-2/00numberhere
 CallerID:  +calleridhere
 Extension: 123
 SetVar: someid=123
 Context: setup
 WaitTime: 30
 MaxRetries: 0
 RetryTime: 300
 Account: 123
 Priority: 1


 Some time after the call has hung up, the call file is still there and
 this is appended to the file:
 StartRetry: 20354 1 (1400070906) (My note: Wed May 14 14:35:06 CEST 2014)

 DelayedRetry: 20354 0 (1400070906) same time...

 DelayedRetry: 20354 0 (1400071206) five minutes...

 DelayedRetry: 20354 0 (1400071506) and so on...

 DelayedRetry: 20354 0 (1400071806) never deleting this file

 DelayedRetry: 20354 0 (1400072106) are we?

 DelayedRetry: 20354 0 (1400072406) nope

 DelayedRetry: 20354 0 (1400072706) waiting for someone

 DelayedRetry: 20354 0 (1400073006) to do manual work




 Asterisk log:
 [May 14 14:35:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
 currently running '/var/spool/asterisk/outgoing/callfile'
 [May 14 14:40:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
 currently running '/var/spool/asterisk/outgoing/callfile'
 [May 14 14:45:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
 currently running '/var/spool/asterisk/outgoing/callfile'
 [May 14 14:50:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
 currently running '/var/spool/asterisk/outgoing/callfile'
 [May 14 14:55:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
 currently running '/var/spool/asterisk/outgoing/callfile'
 [May 14 15:00:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
 currently running '/var/spool/asterisk/outgoing/callfile'
 [May 14 15:05:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
 currently running '/var/spool/asterisk/outgoing/callfile'
 [May 14 15:10:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're
 currently running '/var/spool/asterisk/outgoing/callfile'




 Asterisk code:

if (o-retries = o-maxretries) {
 now += o-retrytime;
 if (o-callingpid  (o-callingpid == ast_mainpid)) {
 safe_append(o, time(NULL), DelayedRetry);
 ast_log(LOG_DEBUG, Delaying retry since we're
 currently running '%s'\n, o-fn);
 free_outgoing(o);
 } else {
 /* Increment retries */
 o-retries++;
 /* If someone else was calling, they're presumably
 gone now
so abort their retry and continue as we were...
 */
 if (o-callingpid)
 safe_append(o, time(NULL), AbortRetry);

 safe_append(o, now, StartRetry);
 launch_service(o);
 }
 return now;
 }




 Sure, I could just disable the retry check and add :
 if (FALSE) {
 And it will always expire should this occur...

 But I'm not sure if this is a good idea or not, and it would be nice not
 having to do that on every upgrade.


 Anyone have experience with what's going on?

 The file can be written to, since safe_append seems to be able to write to
 the file.

 This only happens once in a while, which makes it hard to track down.


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[asterisk-users] .call file retry issue in Asterisk-10.11.1

[pankaj pandey]

Hi,

I am working on Asterisk-10.11.1,I tried to generating outbound call through 
.call file and facing a issue that call retry was happening after call 
Answered.Is it bug in that Version or i missed some thing.
Here is my call file is-

Channel: DAHDI/G1/09990212758
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: menu
Extension: 1234
Priority: 4


Please suggest.


 
Thanks  Regards,
Pankaj Pandey
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Re: [asterisk-users] call file and NFS server

[Steve Edwards]

Please don't top-post.


On Friday 06 July 2012, Chandrakant Solanki wrote:

I have set the folder (callfile/Server{A/B})  permission to 777 as 
well as call file
permission to 777.



On Fri, 6 Jul 2012, A J Stiles wrote:

  (By the way, you should have permissions 666 for a callfile, not 777. 
Callfiles should not be
  executable.)


On Sat, 7 Jul 2012, Chandrakant Solanki wrote:


Once all call completed, I found following error for all files...

[Jul  7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to open
/var/spool/asterisk/outgoing/100097_172.18.100.72.call: No such file or 
directory, deleting
[Jul  7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to open
/var/spool/asterisk/outgoing/100098_172.18.100.72.call: No such file or 
directory, deleting
[Jul  7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to open
/var/spool/asterisk/outgoing/100099_172.18.100.72.call: No such file or 
directory, deleting


Just a SWAG, any chance you set the directory to 666?

Personally, I think AMI would be a better choice for originating calls, 
but that is just a WAG.


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-
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[asterisk-users] call file and NFS server

[Chandrakant Solanki]

Hello,

I have 3 server, 2 running with asterisk and another one generate call
files say some directory callfile/serverA and callfile/serverB (NFS
Sharing) and mounted this directory to respectively on Server A (Asterisk)
and Server B(Asterisk) on /var/spool/asterisk/outgoing.

Server A has Asterisk 1.8.0-rc2 and Server B has asterisk version 1.8.9.0,
and both asterisk compile  ./configure --without-inotify

Callfile will execute call successfully on both machine, but got the
following problem

*[Jul  6 16:15:04] WARNING[26921]: pbx_spool.c:278 safe_append: Unable to
set utime on /var/spool/asterisk/outgoing/15.call: Operation not
permitted
*
I have set the folder (callfile/Server{A/B})  permission to 777 as well as
call file permission to 777.

-- 
Regards,

Chandrakant Solanki
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Re: [asterisk-users] call file and NFS server

[Arstan Jusupov]

Why don't you just generate call files for each of the servers on the same 
server? Anyhow you are not sharing one single pool of call files among servers, 
I suspect that's where network drive would come in handy.

Sent from my iPhone

On Jul 6, 2012, at 6:56 PM, Chandrakant Solanki solanki.chandrak...@gmail.com 
wrote:

 Hello,
 
 I have 3 server, 2 running with asterisk and another one generate call files 
 say some directory callfile/serverA and callfile/serverB (NFS Sharing) and 
 mounted this directory to respectively on Server A (Asterisk) and Server 
 B(Asterisk) on /var/spool/asterisk/outgoing.
 
 Server A has Asterisk 1.8.0-rc2 and Server B has asterisk version 1.8.9.0, 
 and both asterisk compile  ./configure --without-inotify
 
 Callfile will execute call successfully on both machine, but got the 
 following problem
 
 [Jul  6 16:15:04] WARNING[26921]: pbx_spool.c:278 safe_append: Unable to set 
 utime on /var/spool/asterisk/outgoing/15.call: Operation not permitted
 
 I have set the folder (callfile/Server{A/B})  permission to 777 as well as 
 call file permission to 777.
 
 -- 
 Regards,
 
 Chandrakant Solanki
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Re: [asterisk-users] call file and NFS server

[A J Stiles]

On Friday 06 July 2012, Chandrakant Solanki wrote:
 I have 3 server, 2 running with asterisk and another one generate call
 files say some directory callfile/serverA and callfile/serverB (NFS
 Sharing) and mounted this directory to respectively on Server A (Asterisk)
 and Server B(Asterisk) on /var/spool/asterisk/outgoing.
 
 Server A has Asterisk 1.8.0-rc2 and Server B has asterisk version 1.8.9.0,
 and both asterisk compile  ./configure --without-inotify
 
 Callfile will execute call successfully on both machine, but got the
 following problem
 
 *[Jul  6 16:15:04] WARNING[26921]: pbx_spool.c:278 safe_append: Unable to
 set utime on /var/spool/asterisk/outgoing/15.call: Operation not
 permitted
 *
 I have set the folder (callfile/Server{A/B})  permission to 777 as well as
 call file permission to 777.

The problem is that root on one machine doesn't have full root access to other 
users' files on NFS shares.  A user logged in as root on a local machine and 
accessing an NFS share on a remote machine ordinarily has *fewer* privileges, 
and even world write doesn't allow remote root write.  This is by design; as 
otherwise, a local privilege escalation on one machine can lead to a whole-
network privilege escalation.

(By the way, you should have permissions 666 for a callfile, not 777.  
Callfiles 
should not be executable.)

You could either recompile all the NFS stuff  (not really recommended);  or 
have the callfile generated and re-timed by a CGI script on the remote machine  
(where /var/spool/asterisk/outgoing actually is),  fired off by `wget` on the 
local machine.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] call file and NFS server

[Steve Edwards]

On Friday 06 July 2012, Chandrakant Solanki wrote:


I have set the folder (callfile/Server{A/B})  permission to 777 as well 
as call file permission to 777.


On Fri, 6 Jul 2012, A J Stiles wrote:

(By the way, you should have permissions 666 for a callfile, not 777. 
Callfiles should not be executable.)


Whenever I see 777 (or it's Satanic cousin, 666) I see 'I don't really 
understand ownership and permissions so let's just allow everything and 
hope for the best.'


Do you really intend to allow every user and exploited program to be able 
to create call files? (And if you've done this, you've probably created 
other holes in your system's security.)


While 'opening the flood gates' is (IMO) a valid temporary debugging 
technique to identify the source of the problem, the directories and files 
should be owned by the user executing Asterisk and permissions should 
limit reading to only users and groups that need reading and limit writing

to only users and groups that need writing.

I don't have any need or experience with call files on my production 
boxes, but I suspect a successful implementation would include NTP and 
creating the call file in another directory on the shared device and then 
moving the call file to the outgoing spool directory.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] call file and NFS server

[Chandrakant Solanki]

Hi,

I have 100+ call file generated in other directory, and by using program, I
have moved 10-10 files in /var/spool/asterisk/outgoing, and call made
successfully.

Once all call completed, I found following error for all files...

[Jul  7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100097_172.18.100.72.call: No such
file or directory, deleting
[Jul  7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100098_172.18.100.72.call: No such
file or directory, deleting
[Jul  7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100099_172.18.100.72.call: No such
file or directory, deleting


On Fri, Jul 6, 2012 at 8:47 PM, Steve Edwards asterisk@sedwards.comwrote:

 On Friday 06 July 2012, Chandrakant Solanki wrote:


  I have set the folder (callfile/Server{A/B})  permission to 777 as well
 as call file permission to 777.


 On Fri, 6 Jul 2012, A J Stiles wrote:

  (By the way, you should have permissions 666 for a callfile, not 777.
 Callfiles should not be executable.)


 Whenever I see 777 (or it's Satanic cousin, 666) I see 'I don't really
 understand ownership and permissions so let's just allow everything and
 hope for the best.'

 Do you really intend to allow every user and exploited program to be able
 to create call files? (And if you've done this, you've probably created
 other holes in your system's security.)

 While 'opening the flood gates' is (IMO) a valid temporary debugging
 technique to identify the source of the problem, the directories and files
 should be owned by the user executing Asterisk and permissions should limit
 reading to only users and groups that need reading and limit writing
 to only users and groups that need writing.

 I don't have any need or experience with call files on my production
 boxes, but I suspect a successful implementation would include NTP and
 creating the call file in another directory on the shared device and then
 moving the call file to the outgoing spool directory.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000


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-- 
Regards,

Chandrakant Solanki
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Re: [asterisk-users] call file challenge...

[Positively Optimistic]

Thanks for the response..  I just got the opportunity to try this with the
wait time adjusted to 15..  and got the same result...

[2011-06-22 04:42:47] NOTICE[19692]: pbx_spool.c:339 attempt_thread: Call
failed to go through, reason (3) Remote end Ringing

so far I've been unable to identify what generates the Remote end
Ringing message...

On Wed, Jun 15, 2011 at 5:39 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:

 Hi,

 I think  you need to update *waittime* parameter in .call file please put
 atleast 10 seconds.
 for more understanding please try to read

 *http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out*

 Regards
 Dhaval

   On Wed, Jun 15, 2011 at 12:15 PM, Positively Optimistic 
 positivelyoptimis...@gmail.com wrote:

  Greetings!!

   We're getting some strange results using call files..  no matter the
 technology, DAHDI, SIP, etc., we get a Call failed to go through, reason
 (3) Remote end Ringing message when attempting to originate a call from a
 call file.  Numbers changed to protect the innocent



 using call file
 //CALL FILE//

  Channel: DAHDI/g1/918005551212
  Callerid: 8002211212
  WaitTime: 2
  MaxRetries: 6
  RetryTime: 8

  Context: xs-globx-ds3
  Extension: 12564286000
  Priority: 1

 //CALL FILE//

 //CLI SNIPPET//

  -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 1)
  -- Requested transfer capability: 0x00 - SPEECH
  -- PROGRESS with cause code 31 received
  -- Hungup 'DAHDI/1-1'
  [2011-06-15 01:35:14] NOTICE[27176]: pbx_spool.c:339 attempt_thread:
 Call failed to go through, reason (3) Remote end Ringing
  -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 2)
  -- Requested transfer capability: 0x00 - SPEECH
  -- PROGRESS with cause code 31 received
  -- Hungup 'DAHDI/1-1'
  [2011-06-15 01:35:24] NOTICE[27177]: pbx_spool.c:339 attempt_thread:
 Call failed to go through, reason (3) Remote end Ringing
  -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 3)
  -- Requested transfer capability: 0x00 - SPEECH
  -- PROGRESS with cause code 31 received
  -- Hungup 'DAHDI/1-1'
  [2011-06-15 01:35:34] NOTICE[27179]: pbx_spool.c:339 attempt_thread:
 Call failed to go through, reason (3) Remote end Ringing
  -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 4)
  -- Requested transfer capability: 0x00 - SPEECH
  -- PROGRESS with cause code 31 received
  -- Hungup 'DAHDI/1-1'
  [2011-06-15 01:35:44] NOTICE[27182]: pbx_spool.c:339 attempt_thread:
 Call failed to go through, reason (3) Remote end Ringing
  -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 5)
  -- Requested transfer capability: 0x00 - SPEECH
  -- PROGRESS with cause code 31 received
  -- Hungup 'DAHDI/1-1'
  [2011-06-15 01:35:54] NOTICE[27183]: pbx_spool.c:339 attempt_thread:
 Call failed to go through, reason (3) Remote end Ringing
  -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 6)
  -- Requested transfer capability: 0x00 - SPEECH
  -- PROGRESS with cause code 31 received
  -- Hungup 'DAHDI/1-1'
  [2011-06-15 01:36:04] NOTICE[27185]: pbx_spool.c:339 attempt_thread:
 Call failed to go through, reason (3) Remote end Ringing
  -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 7)
  -- Requested transfer capability: 0x00 - SPEECH
  -- PROGRESS with cause code 31 received
  -- Hungup 'DAHDI/1-1'
  [2011-06-15 01:36:14] NOTICE[27188]: pbx_spool.c:339 attempt_thread:
 Call failed to go through, reason (3) Remote end Ringing

 //CLI SNIPPET//

 Software Version(s)

 Asterisk 1.6.2.16.1
 DAHDI Version: 2.4.0
  libpri version: 1.4.11.5




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Re: [asterisk-users] call file challenge...

[DHAVAL INDRODIYA]

Hi,

I think  you need to update *waittime* parameter in .call file please put
atleast 10 seconds.
for more understanding please try to read

*http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out*

Regards
Dhaval

On Wed, Jun 15, 2011 at 12:15 PM, Positively Optimistic 
positivelyoptimis...@gmail.com wrote:

 Greetings!!

 We're getting some strange results using call files..  no matter the
 technology, DAHDI, SIP, etc., we get a Call failed to go through, reason
 (3) Remote end Ringing message when attempting to originate a call from a
 call file.  Numbers changed to protect the innocent



 using call file
 //CALL FILE//

 Channel: DAHDI/g1/918005551212
 Callerid: 8002211212
 WaitTime: 2
 MaxRetries: 6
 RetryTime: 8

 Context: xs-globx-ds3
 Extension: 12564286000
 Priority: 1

 //CALL FILE//

 //CLI SNIPPET//

 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 1)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:14] NOTICE[27176]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 2)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:24] NOTICE[27177]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 3)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:34] NOTICE[27179]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 4)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:44] NOTICE[27182]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 5)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:54] NOTICE[27183]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 6)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:36:04] NOTICE[27185]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 7)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:36:14] NOTICE[27188]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing

 //CLI SNIPPET//

 Software Version(s)

 Asterisk 1.6.2.16.1
 DAHDI Version: 2.4.0
 libpri version: 1.4.11.5




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Re: [asterisk-users] call file for page auto-call

[Bryant Zimmerman]

 From: satish patel satish...@hotmail.com
Sent: Tuesday, March 15, 2011 2:31 PM
To: asterisk-users asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call file for page auto-call

 Thanks for you input but how to do  SIPAddHeader(Alert-Info: Ring Answer)   
for auto answer my polycom phones and how to create group in .call file I am 
reading at http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out  but 
didn't found anything related group calling. may be i am missing something 
could point me out..

-S

   Hey Support,

I am planing to implement new page system with asterisk 1.8  we have 200 SIP 
calls and page() will overkill my system if announce in one shot. so i am 
planing to record and play page over 50...50...50 chunk..

I am planing to do with .call file for auto calling after record message but i 
don't know how to call multiple extension ? and how to use page() with .call 
file for auto-answer and auto-call?

Appreciate your help..

-S One suggestion - set up 4 call groups.  Group 1 calls first 50 phones, 
Group 2 51-100, etc.  If you set it up like 601, 602, etc. then in your call 
file you can test with 101 to get what you want, then change it to 601.

satish

We have a page group offering in our systems. We do not use call files to 
handle this we do it as direct processing. If I were to use a call file. I 
would create a custom context to use from the call file. The first thing I 
would do is build a string list of the phones being paged. The second is I 
would add the auto answer headers for the different types of phones that are in 
my network. This process is really quite straight forward.  The flow would be 
somthing like this..

Call Page Record.
 Call in.
 Record Message.
 Select page groups to send the message to.
 Write a call file with the message name, page groups and the page handling 
context.

Call file would contain.
 Custom page handling/processing context.
 List of page groups and message file name stored in vars.

In your Custom page handling/processing context.
 Read and parse the page groups list from a variable set in the .call file
 Read the recorded message from the .call file
Loop for each page group.
 Build your paging group in a string (This should be able to be done 
using some kind of list. Either stright. csv or database you choose)
 Set the correct page headers
 Call the page command with the correct list.
 Play the recorded message
 Hangup
Loop back and do next group.

This is really just a coding project. You have to break the entire issue down 
into it's base parts and then solve each one.

Good luck.



 Bryant Z
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[asterisk-users] call file for page auto-call

[satish patel]

Hey Support,

I am planing to implement new page system with asterisk 1.8  we have 200 SIP 
calls and page() will overkill my system if announce in one shot. so i am 
planing to record and play page over 50...50...50 chunk..

I am planing to do with .call file for auto calling after record message but i 
don't know how to call multiple extension ? and how to use page() with .call 
file for auto-answer and auto-call?

Appreciate your help..

-S
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Re: [asterisk-users] call file for page auto-call

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, March 15, 2011 1:06 PM
To: asterisk-users
Subject: [asterisk-users] call file for page auto-call

 

Hey Support,

I am planing to implement new page system with asterisk 1.8  we have 200 SIP
calls and page() will overkill my system if announce in one shot. so i am
planing to record and play page over 50...50...50 chunk..

I am planing to do with .call file for auto calling after record message but
i don't know how to call multiple extension ? and how to use page() with
.call file for auto-answer and auto-call?

Appreciate your help..

-S

 

One suggestion - set up 4 call groups.  Group 1 calls first 50 phones,
Group 2 51-100, etc.  If you set it up like 601, 602, etc. then in your call
file you can test with 101 to get what you want, then change it to 601.

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Re: [asterisk-users] call file for page auto-call

[satish patel]

Thanks for you input but how to do  SIPAddHeader(Alert-Info: Ring Answer)   for 
auto answer my polycom phones and how to create group in .call file I am 
reading at http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out  but 
didn't found anything related group calling. may be i am missing something 
could point me out..

-S

From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 15 Mar 2011 13:11:16 -0500
Subject: Re: [asterisk-users] call file for page auto-call



























From:
asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of satish patel

Sent: Tuesday, March 15, 2011 1:06
PM

To: asterisk-users

Subject: [asterisk-users] call
file for page auto-call



 

Hey Support,



I am planing to implement new page system with asterisk 1.8  we have 200
SIP calls and page() will overkill my system if announce in one shot. so i am
planing to record and play page over 50...50...50 chunk..



I am planing to do with .call file for auto calling after record message but i
don't know how to call multiple extension ? and how to use page() with .call
file for auto-answer and auto-call?



Appreciate your help..



-S

 

One suggestion – set up 4 “call
groups”.  Group 1 calls first 50 phones, Group 2 51-100, etc.  If you set
it up like 601, 602, etc. then in your call file you can test with 101 to get
what you want, then change it to 601.







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Re: [asterisk-users] call file question

[Jeff LaCoursiere]

On Wed, 30 Jun 2010, Steve Edwards wrote:

 Now I whipped up a C program to create a call file to do the same thing
 from the command line:

 [snip]
 fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n);

 I don't see exten *71 in custom-callfwd.

Doh!  That was the problem.  In FreePBX I made *71 the feature code to 
access that context, and it was still in my head when I made the callfile.


 Why are you using a local channel in your call file?


That was the meat of the question, actually.  I want to create a single 
leg with a callfile - just the outbound call.  All other times I have used 
callfiles I was creating two legs and bridging them.  Is there a better 
way to do what I am attempting?


  fprintf(callfile, Application: Playback\n);
  fprintf(callfile, Data: hello-world\n);
 [snip]

 When I run this it creates the call file and I see this in the console:

 -- Attempting call on Local/*...@custom-callfwd/n for application
 Playback(hello-world) (Retry 1)

 What does the call file look like before you mv it to the spool directory?


Exactly the above fprintf lines...

Thanks,

j

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[asterisk-users] call file question

[Jeff LaCoursiere]

I am sure this is simple, but have been struggling.  I want to create a 
call file that dials out a particular Dahdi channel to enable call 
forwarding on a POTS line.  I have this in extensions.conf:

[custom-callfwd]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*717157750)
exten = s,n,Verbose(${DIALSTATUS})
exten = s,n,Hangup

[custom-callfwdcanc]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*72)
exten = s,n,Verbose(${DIALSTATUS})
exten = s,n,Hangup

Using FreePBX I have setup custom destinations and custom 
applications such that users can dial a code from their desks and enable 
or disable forwarding via the above contexts.  That works fine.

Now I whipped up a C program to create a call file to do the same thing 
from the command line:

[snip]
 fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n);
fprintf(callfile, Application: Playback\n);
fprintf(callfile, Data: hello-world\n);
[snip]

When I run this it creates the call file and I see this in the console:

 -- Attempting call on Local/*...@custom-callfwd/n for application 
Playback(hello-world) (Retry 1)

And that is all... no call actually goes out on the Dahdi line.

I'm sure I am not properly creating the call file to do what I want.  Any 
suggestions?

Thanks,

j

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Re: [asterisk-users] call file question

[Steve Edwards]

On Thu, 1 Jul 2010, Jeff LaCoursiere wrote:

 I am sure this is simple, but have been struggling.  I want to create a 
 call file that dials out a particular Dahdi channel to enable call 
 forwarding on a POTS line.  I have this in extensions.conf:

 [custom-callfwd]
 exten = s,1,Answer
 exten = s,n,Dial(DAHDI/4-1/*717157750)
 exten = s,n,Verbose(${DIALSTATUS})
 exten = s,n,Hangup

 [custom-callfwdcanc]
 exten = s,1,Answer
 exten = s,n,Dial(DAHDI/4-1/*72)
 exten = s,n,Verbose(${DIALSTATUS})
 exten = s,n,Hangup

 Using FreePBX I have setup custom destinations and custom 
 applications such that users can dial a code from their desks and 
 enable or disable forwarding via the above contexts.  That works fine.

 Now I whipped up a C program to create a call file to do the same thing 
 from the command line:

 [snip]
 fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n);

I don't see exten *71 in custom-callfwd.

Why are you using a local channel in your call file?

   fprintf(callfile, Application: Playback\n);
   fprintf(callfile, Data: hello-world\n);
 [snip]

 When I run this it creates the call file and I see this in the console:

 -- Attempting call on Local/*...@custom-callfwd/n for application
 Playback(hello-world) (Retry 1)

What does the call file look like before you mv it to the spool directory?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Call file structure and syntax

[Mike Ely]

Yep, I saw that and it's just not the case.  I was having it dial my desk
extension, which was decidedly not busy at the time...


On 6/28/10 5:30 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:

 Well, I¹ve tried this, and something just isn¹t right.
 
 Look here:
 
 Event: Hangup
 Channel: SIP/ShoreTel-1-0004
 Cause: 17   
 Cause-txt: User busy
 


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Re: [asterisk-users] Call file structure and syntax

[Mike Ely]

 Of Mike Ely
 Sent: Tuesday, June 22, 2010 12:02 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Call file structure and syntax
  
 Hi there,
 
 I¹ve been looking to do an outbound dialer for systems alerting, etc. and have
 in large part followed the recipe here:
 http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
 
 That and the associated pages at voip-info give a basic set of recipes for
 callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen,
 Madsen,  Smith can I find a detailed discussion of what goes into a callfile,
 how to get it to do things like interact with the shell (I¹d like ³Press 2² in
 my outbound call to do something of value), etc.  I¹ve googled around but
 haven¹t found what I¹m looking for, just other people¹s ³Hello World²
 callfiles.  As of now, I can make outbound calls well enough, but want more...
 
 Can someone point me in the right direction for this?
 
 Thanks,
 Mike 
 
 


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Re: [asterisk-users] Call file structure and syntax

[Philipp von Klitzing]

 Well, I¹ve tried this, and something just isn¹t right.

Look here:

 Event: Hangup
 Channel: SIP/ShoreTel-1-0004
 Cause: 17  
 Cause-txt: User busy


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[asterisk-users] Call file structure and syntax

[Mike Ely]

Hi there,

I¹ve been looking to do an outbound dialer for systems alerting, etc. and
have in large part followed the recipe here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

That and the associated pages at voip-info give a basic set of recipes for
callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen,
Madsen,  Smith can I find a detailed discussion of what goes into a
callfile, how to get it to do things like interact with the shell (I¹d like
³Press 2² in my outbound call to do something of value), etc.  I¹ve googled
around but haven¹t found what I¹m looking for, just other people¹s ³Hello
World² callfiles.  As of now, I can make outbound calls well enough, but
want more...

Can someone point me in the right direction for this?

Thanks,
Mike
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Re: [asterisk-users] Call file structure and syntax

[Danny Nicholas]

#1 - once you've got to this point, AMI would be a better option than a call
file

#2 -  using AMI or a call file, you are going to want to use the
context-based method instead of application to get the most bang for your
buck

 

I use a bigger instance of this to play a message and accept 1 or 2 from the
user

; this context is used by AMI to play a message

[accept]

exten = s,1,Answer

exten = s,n,Background(important)

exten = s,n,WaitExten(5,m)

exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))

exten = 1,n,Background(${Data})

exten = 1,n,Background(repeatmsg)

exten = 1,n,WaitExten(5,m)

exten = 1,n,Hangup

exten = 2,1,Background(calllater)

exten = 2,n,ForkCDR(v,s(reject=${Data}))

exten = 2,n,Hangup

exten = 3,1,Goto(accept|1|2)

exten = *,1,Goto(accept|s|1)

exten = i,1,Goto(accept|s|1)

exten = t,1,Goto(accept|s|1)

 

here's the call file

Action = 'Originate',

  Channel = DAHDI/1,

  Variable = Data=/tmp/test.gsm,

  Exten = 'SIP/170',

  Context = 'accept',

  priority = 1,

  Number = 5551212

Using the accept context, 5551212 is called on DAHDI/1 and user hears
important.gsm.  then they press 1 to hear test.gsm or 2 to hear it later.

 

Hope this is helpful.

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Ely
Sent: Tuesday, June 22, 2010 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call file structure and syntax

 

Hi there,

I've been looking to do an outbound dialer for systems alerting, etc. and
have in large part followed the recipe here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

That and the associated pages at voip-info give a basic set of recipes for
callfiles, but nowhere there or in my copy of the O'Reilly book by Meggelen,
Madsen,  Smith can I find a detailed discussion of what goes into a
callfile, how to get it to do things like interact with the shell (I'd like
Press 2 in my outbound call to do something of value), etc.  I've googled
around but haven't found what I'm looking for, just other people's Hello
World callfiles.  As of now, I can make outbound calls well enough, but
want more...

Can someone point me in the right direction for this?

Thanks,
Mike 

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Re: [asterisk-users] Call file structure and syntax

[Mike Ely]

That¹s a good start.  In my case, I want it to dial a round-robin queue (set
up separately) and if the user presses 2, stop dialing the queue and log
which user acknowledged the alarm.  If the user presses 1, repeat the
message, if no key is pressed before a timeout, hang up and dial the next
user in the queue.  Or something like that.  One of the things I also want
to be able to do with this is echo out something to the shell, either a
textfile or an actual command so that I can trigger some other actions not
necessarily related to Asterisk.

It¹s a fun project except for the knowledge that successful completion is
going to mean it wakes me up some night at 3am.



On 6/22/10 10:31 AM, Danny Nicholas da...@debsinc.com wrote:

 #1 ­ once you¹ve got to this point, AMI would be a better option than a call
 file
 #2 -  using AMI or a call file, you are going to want to use the context-based
 method instead of application to get the most ³bang for your buck²
  
 I use a bigger instance of this to play a message and accept 1 or 2 from the
 user
 ; this context is used by AMI to play a message
 [accept]
 exten = s,1,Answer
 exten = s,n,Background(important)
 exten = s,n,WaitExten(5,m)
 exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))
 exten = 1,n,Background(${Data})
 exten = 1,n,Background(repeatmsg)
 exten = 1,n,WaitExten(5,m)
 exten = 1,n,Hangup
 exten = 2,1,Background(calllater)
 exten = 2,n,ForkCDR(v,s(reject=${Data}))
 exten = 2,n,Hangup
 exten = 3,1,Goto(accept|1|2)
 exten = *,1,Goto(accept|s|1)
 exten = i,1,Goto(accept|s|1)
 exten = t,1,Goto(accept|s|1)
  
 here¹s the call file
 Action = 'Originate',
  Channel = DAHDI/1,
  Variable = Data=/tmp/test.gsm²,
  Exten = 'SIP/170',
  Context = 'accept',
  priority = 1,
  Number = 5551212
 Using the accept context, 5551212 is called on DAHDI/1 and user hears
 important.gsm.  then they press 1 to hear test.gsm or 2 to hear it later.
  
 Hope this is helpfulŠ
 
 
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Ely
 Sent: Tuesday, June 22, 2010 12:02 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Call file structure and syntax
  
 Hi there,
 
 I¹ve been looking to do an outbound dialer for systems alerting, etc. and have
 in large part followed the recipe here:
 http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
 
 That and the associated pages at voip-info give a basic set of recipes for
 callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen,
 Madsen,  Smith can I find a detailed discussion of what goes into a callfile,
 how to get it to do things like interact with the shell (I¹d like ³Press 2² in
 my outbound call to do something of value), etc.  I¹ve googled around but
 haven¹t found what I¹m looking for, just other people¹s ³Hello World²
 callfiles.  As of now, I can make outbound calls well enough, but want more...
 
 Can someone point me in the right direction for this?
 
 Thanks,
 Mike 
 
 
 

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Re: [asterisk-users] Call file structure and syntax

[Zeeshan Zakaria]

Danny, you replies are the best on this list, with working dialplans, what
could be a better reply.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-22 2:01 PM, Mike Ely mike...@amyskitchen.net wrote:

 That’s a good start.  In my case, I want it to dial a round-robin queue
(set up separately) and if the user presses 2, stop dialing the queue and
log which user acknowledged the alarm.  If the user presses 1, repeat the
message, if no key is pressed before a timeout, hang up and dial the next
user in the queue.  Or something like that.  One of the things I also want
to be able to do with this is echo out something to the shell, either a
textfile or an actual command so that I can trigger some other actions not
necessarily related to Asterisk.

It’s a fun project except for the knowledge that successful completion is
going to mean it wakes me up some night at 3am.





On 6/22/10 10:31 AM, Danny Nicholas da...@debsinc.com wrote:

 #1 – once you’ve got to this point, AMI would be a better option than a
call file
 #2 -  using AM...

--



 From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] ...

--


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[asterisk-users] Call File Channel

[David Gibbons]

I know I'm missing something here (been a long day)...

How can I specify more than one channel in a call file?

I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1...

Thanks
Dave
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Re: [asterisk-users] Call File Channel

[Danny Nicholas]

Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call File Channel

 

I know I'm missing something here (been a long day).

 

How can I specify more than one channel in a call file?

 

I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1.

 

Thanks

Dave

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Re: [asterisk-users] Call File Channel

[David Gibbons]

Thanks Danny,

I do have a dial cmd with multiple arguments in my normal outgoing context. I 
guess my question really is:

How do I tell the call file using Channel: XXX to use my outgoing context 
instead of Zap/g1/xx or sip/trunk_x/xx directly?

-Dave

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, August 12, 2009 5:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60)


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call File Channel

I know I'm missing something here (been a long day)...

How can I specify more than one channel in a call file?

I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1...

Thanks
Dave
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Re: [asterisk-users] Call File Channel

[Danny Nicholas]

Ok.  Here's how you would do that:

 

Channel: SIP/170 (some local extension)

CallerID: SIP/104 (another local extension)

MaxRetries: 1

WaitTime: 60

retryTime: 5

Context: your_context

Extension: s

 

This should create an extension call using your context.  The context can
then dial out as you write it.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 4:10 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

 

Thanks Danny,

 

I do have a dial cmd with multiple arguments in my normal outgoing context.
I guess my question really is:

 

How do I tell the call file using Channel: XXX to use my outgoing context
instead of Zap/g1/xx or sip/trunk_x/xx directly?

 

-Dave

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, August 12, 2009 5:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

 

Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call File Channel

 

I know I'm missing something here (been a long day).

 

How can I specify more than one channel in a call file?

 

I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1.

 

Thanks

Dave

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Re: [asterisk-users] Call File Channel

[David Gibbons]

Context: is what the call is dumped into after it is answered, at extension 
Extension:. I don't think it's related to how the call is placed.

I can dial the local extension SIP/170 but I'm not sure where that gets me.

Basically I want to have the same failover that I have for all other outgoing 
calls on these automatic calls...

Thanks
Dave

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, August 12, 2009 5:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

Ok.  Here's how you would do that:

Channel: SIP/170 (some local extension)
CallerID: SIP/104 (another local extension)
MaxRetries: 1
WaitTime: 60
retryTime: 5
Context: your_context
Extension: s

This should create an extension call using your context.  The context can then 
dial out as you write it.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 4:10 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

Thanks Danny,

I do have a dial cmd with multiple arguments in my normal outgoing context. I 
guess my question really is:

How do I tell the call file using Channel: XXX to use my outgoing context 
instead of Zap/g1/xx or sip/trunk_x/xx directly?

-Dave

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, August 12, 2009 5:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60)


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call File Channel

I know I'm missing something here (been a long day)...

How can I specify more than one channel in a call file?

I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1...

Thanks
Dave
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Re: [asterisk-users] Call File Channel

[Danny Nicholas]

Your'e wanting control of the call from a call file.  The way to do that is
to call using a context instead of a Technology/number.  

When you call SIP/trunk_1, you are using the default context and therefore
don't have any fallthrough options unless you wrote them into your default
context.  If your default context allows dynamic handling on fallthrough,
you would probably still want to call number 1 using a context.  I used
SIP/170 as an example; you could use SIP/trunk1/#1 just as easily.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 4:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

 

Context: is what the call is dumped into after it is answered, at extension
Extension:. I don't think it's related to how the call is placed.

 

I can dial the local extension SIP/170 but I'm not sure where that gets me.

 

Basically I want to have the same failover that I have for all other
outgoing calls on these automatic calls.

 

Thanks

Dave

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, August 12, 2009 5:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

 

Ok.  Here's how you would do that:

 

Channel: SIP/170 (some local extension)

CallerID: SIP/104 (another local extension)

MaxRetries: 1

WaitTime: 60

retryTime: 5

Context: your_context

Extension: s

 

This should create an extension call using your context.  The context can
then dial out as you write it.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 4:10 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

 

Thanks Danny,

 

I do have a dial cmd with multiple arguments in my normal outgoing context.
I guess my question really is:

 

How do I tell the call file using Channel: XXX to use my outgoing context
instead of Zap/g1/xx or sip/trunk_x/xx directly?

 

-Dave

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, August 12, 2009 5:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call File Channel

 

Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call File Channel

 

I know I'm missing something here (been a long day).

 

How can I specify more than one channel in a call file?

 

I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1.

 

Thanks

Dave

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Re: [asterisk-users] Call File Channel

[Duncan Turnbull]

If you use a Local channel to dial it then it will fall under the same rules

Channel: Local/numbertod...@the-context-you-want

This gets a CDR produced, it does pay to check everything works the same 
but it should be fine

Cheers Duncan

David Gibbons wrote:

 Context: is what the call is dumped into after it is answered, at 
 extension Extension:. I don’t think it’s related to how the call is 
 placed.

 I can dial the local extension SIP/170 but I’m not sure where that 
 gets me.

 Basically I want to have the same failover that I have for all other 
 outgoing calls on these automatic calls…

 Thanks

 Dave

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny 
 Nicholas
 *Sent:* Wednesday, August 12, 2009 5:17 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Ok. Here’s how you would do that:

 Channel: SIP/170 (some local extension)

 CallerID: SIP/104 (another local extension)

 MaxRetries: 1

 WaitTime: 60

 retryTime: 5

 Context: your_context

 Extension: s

 This should create an extension call using your context. The context 
 can then dial out as you write it.

 

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David 
 Gibbons
 *Sent:* Wednesday, August 12, 2009 4:10 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Thanks Danny,

 I do have a dial cmd with multiple arguments in my normal outgoing 
 context. I guess my question really is:

 How do I tell the call file using “Channel: XXX” to use my outgoing 
 context instead of Zap/g1/xx or sip/trunk_x/xx directly?

 -Dave

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny 
 Nicholas
 *Sent:* Wednesday, August 12, 2009 5:05 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60)

 

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David 
 Gibbons
 *Sent:* Wednesday, August 12, 2009 3:59 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Call File Channel

 I know I’m missing something here (been a long day)…

 How can I specify more than one channel in a call file?

 I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1…

 Thanks

 Dave

 

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Re: [asterisk-users] Call File Channel

[David Gibbons]

Duncan and Danny--

Thank you! I believe the Local/ is what I was missing with ex...@context.

-Dave

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Duncan Turnbull 
[dun...@e-simple.co.nz]
Sent: Wednesday, August 12, 2009 5:42 PM
To: Asterisk Users Mailing List - 
Non-Cohttps://mail.videon-central.net/owa/?ae=PreFormActiont=IPM.Notea=Replyid=RgDvdntYewg%2bRopom4XHVQiWBwDABk4e%2fzVQQKMcsNSFUOsuAE10SQAHAAD54%2bBr%2fe7oQrgyh88yX6qLANRp8a4EAAAJ#mmercial
 Discussion
Subject: Re: [asterisk-users] Call File Channel

If you use a Local channel to dial it then it will fall under the same rules

Channel: Local/numbertod...@the-context-you-want

This gets a CDR produced, it does pay to check everything works the same
but it should be fine

Cheers Duncan

David Gibbons wrote:

 Context: is what the call is dumped into after it is answered, at
 extension Extension:. I don’t think it’s related to how the call is
 placed.

 I can dial the local extension SIP/170 but I’m not sure where that
 gets me.

 Basically I want to have the same failover that I have for all other
 outgoing calls on these automatic calls…

 Thanks

 Dave

 *From:* asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny
 Nicholas
 *Sent:* Wednesday, August 12, 2009 5:17 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Ok. Here’s how you would do that:

 Channel: SIP/170 (some local extension)

 CallerID: SIP/104 (another local extension)

 MaxRetries: 1

 WaitTime: 60

 retryTime: 5

 Context: your_context

 Extension: s

 This should create an extension call using your context. The context
 can then dial out as you write it.

 

 *From:* asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David
 Gibbons
 *Sent:* Wednesday, August 12, 2009 4:10 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Thanks Danny,

 I do have a dial cmd with multiple arguments in my normal outgoing
 context. I guess my question really is:

 How do I tell the call file using “Channel: XXX” to use my outgoing
 context instead of Zap/g1/xx or sip/trunk_x/xx directly?

 -Dave

 *From:* asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny
 Nicholas
 *Sent:* Wednesday, August 12, 2009 5:05 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Call File Channel

 Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60)

 

 *From:* asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David
 Gibbons
 *Sent:* Wednesday, August 12, 2009 3:59 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Call File Channel

 I know I’m missing something here (been a long day)…

 How can I specify more than one channel in a call file?

 I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1…

 Thanks

 Dave

 

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Re: [asterisk-users] call file concurrency

[Christian Victor]

2009/2/27 Bill Michaelson b...@cosi.com

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?


Afaik only by limiting the number of call files in the directory.
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Re: [asterisk-users] call file concurrency

[Danny Nicholas]

Some variant of the ulimit command would accomplish this but YMMV and
Caveat Emptor.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill
Michaelson
Sent: Thursday, February 26, 2009 7:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] call file concurrency

Is there a convenient way to limit the number of call files (outgoing 
directory) that are processed concurrently?


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Re: [asterisk-users] call file concurrency

[Steve Edwards]

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?

On Fri, 27 Feb 2009, Danny Nicholas top posted:

 Some variant of the ulimit command would accomplish this but YMMV and
 Caveat Emptor.

Which one?

-fs::sedwards:~$ ulimit -a
core file size  (blocks, -c) 0
data seg size   (kbytes, -d) unlimited
file size   (blocks, -f) unlimited
pending signals (-i) 1024
max locked memory   (kbytes, -l) 32
max memory size (kbytes, -m) unlimited
open files  (-n) 1024
pipe size(512 bytes, -p) 8
POSIX message queues (bytes, -q) 819200
stack size  (kbytes, -s) 10240
cpu time   (seconds, -t) unlimited
max user processes  (-u) 16114
virtual memory  (kbytes, -v) unlimited
file locks  (-x) unlimited

Limiting the number of open files or file locks would not have the 
intended effect.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] call file concurrency

[Danny Nicholas]

Here is a link to a better, but possibly dangerous answer.

http://www.netadmintools.com/art295.html

Since a typical linux box probably allows about 250K files to be
simultaneously open, and you need about 2K for system and * overhead, by
cutting the max number of files down to about 3K, you would limit the number
of calls to about 1K, assuming that each open call is one file handle.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, February 27, 2009 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call file concurrency

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?

On Fri, 27 Feb 2009, Danny Nicholas top posted:

 Some variant of the ulimit command would accomplish this but YMMV and
 Caveat Emptor.

Which one?

-fs::sedwards:~$ ulimit -a
core file size  (blocks, -c) 0
data seg size   (kbytes, -d) unlimited
file size   (blocks, -f) unlimited
pending signals (-i) 1024
max locked memory   (kbytes, -l) 32
max memory size (kbytes, -m) unlimited
open files  (-n) 1024
pipe size(512 bytes, -p) 8
POSIX message queues (bytes, -q) 819200
stack size  (kbytes, -s) 10240
cpu time   (seconds, -t) unlimited
max user processes  (-u) 16114
virtual memory  (kbytes, -v) unlimited
file locks  (-x) unlimited

Limiting the number of open files or file locks would not have the 
intended effect.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] call file concurrency

[Eric Wieling, Asteria Solutions Group]

Set the ctime of the spool file in the future and Asterisk will not 
process the file until that time.

Danny Nicholas wrote:
 Here is a link to a better, but possibly dangerous answer.
 
 http://www.netadmintools.com/art295.html
 
 Since a typical linux box probably allows about 250K files to be
 simultaneously open, and you need about 2K for system and * overhead, by
 cutting the max number of files down to about 3K, you would limit the number
 of calls to about 1K, assuming that each open call is one file handle.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
 Sent: Friday, February 27, 2009 11:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] call file concurrency
 
 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?
 
 On Fri, 27 Feb 2009, Danny Nicholas top posted:
 


-- 
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com

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Re: [asterisk-users] call file concurrency

[Steve Edwards]

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
 Sent: Friday, February 27, 2009 11:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] call file concurrency

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?

 On Fri, 27 Feb 2009, Danny Nicholas top posted:

 Some variant of the ulimit command would accomplish this but YMMV and
 Caveat Emptor.

On Fri, 27 Feb 2009, Steve Edwards wrote:

 Which one?

   -fs::sedwards:~$ ulimit -a
   core file size  (blocks, -c) 0
   data seg size   (kbytes, -d) unlimited
   file size   (blocks, -f) unlimited
   pending signals (-i) 1024
   max locked memory   (kbytes, -l) 32
   max memory size (kbytes, -m) unlimited
   open files  (-n) 1024
   pipe size(512 bytes, -p) 8
   POSIX message queues (bytes, -q) 819200
   stack size  (kbytes, -s) 10240
   cpu time   (seconds, -t) unlimited
   max user processes  (-u) 16114
   virtual memory  (kbytes, -v) unlimited
   file locks  (-x) unlimited

 Limiting the number of open files or file locks would not have the
 intended effect.

On Fri, 27 Feb 2009, Danny Nicholas top posted:

 Here is a link to a better, but possibly dangerous answer.

 http://www.netadmintools.com/art295.html

 Since a typical linux box probably allows about 250K files to be 
 simultaneously open, and you need about 2K for system and * overhead, by 
 cutting the max number of files down to about 3K, you would limit the 
 number of calls to about 1K, assuming that each open call is one file 
 handle.

I think proposing to control the number of concurrently processed call 
files by inducing file descriptor exhaustion is about 32 days premature.

Calls would fail at random and you may or may not be able to log in or 
even execute a command line depending on if you were currently exhausted 
at any particular instant.

I think the OP is looking for some knob to turn in pbx_spool.c

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] call file concurrency

[Steve Edwards]

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
 Sent: Friday, February 27, 2009 11:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] call file concurrency

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?

On Fri, 27 Feb 2009, Eric Wieling, Asteria Solutions Group top posted:

 Set the ctime of the spool file in the future and Asterisk will not 
 process the file until that time.

This only controls when, not how many.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] call file concurrency

[Danny Nicholas]

Agreed, but the OP seemed to be looking for a command-line solution, not a
C one.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, February 27, 2009 2:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call file concurrency

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
 Sent: Friday, February 27, 2009 11:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] call file concurrency

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?

 On Fri, 27 Feb 2009, Danny Nicholas top posted:

 Some variant of the ulimit command would accomplish this but YMMV and
 Caveat Emptor.

On Fri, 27 Feb 2009, Steve Edwards wrote:

 Which one?

   -fs::sedwards:~$ ulimit -a
   core file size  (blocks, -c) 0
   data seg size   (kbytes, -d) unlimited
   file size   (blocks, -f) unlimited
   pending signals (-i) 1024
   max locked memory   (kbytes, -l) 32
   max memory size (kbytes, -m) unlimited
   open files  (-n) 1024
   pipe size(512 bytes, -p) 8
   POSIX message queues (bytes, -q) 819200
   stack size  (kbytes, -s) 10240
   cpu time   (seconds, -t) unlimited
   max user processes  (-u) 16114
   virtual memory  (kbytes, -v) unlimited
   file locks  (-x) unlimited

 Limiting the number of open files or file locks would not have the
 intended effect.

On Fri, 27 Feb 2009, Danny Nicholas top posted:

 Here is a link to a better, but possibly dangerous answer.

 http://www.netadmintools.com/art295.html

 Since a typical linux box probably allows about 250K files to be 
 simultaneously open, and you need about 2K for system and * overhead, by 
 cutting the max number of files down to about 3K, you would limit the 
 number of calls to about 1K, assuming that each open call is one file 
 handle.

I think proposing to control the number of concurrently processed call 
files by inducing file descriptor exhaustion is about 32 days premature.

Calls would fail at random and you may or may not be able to log in or 
even execute a command line depending on if you were currently exhausted 
at any particular instant.

I think the OP is looking for some knob to turn in pbx_spool.c

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] call file concurrency

[Steve Edwards]

 Sent: Friday, February 27, 2009 11:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] call file concurrency

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?

 On Fri, 27 Feb 2009, Danny Nicholas top posted:

 Some variant of the ulimit command would accomplish this but YMMV and
 Caveat Emptor.

 On Fri, 27 Feb 2009, Steve Edwards wrote:

 I think proposing to control the number of concurrently processed call
 files by inducing file descriptor exhaustion is about 32 days premature.

 Calls would fail at random and you may or may not be able to log in or
 even execute a command line depending on if you were currently exhausted
 at any particular instant.

 I think the OP is looking for some knob to turn in pbx_spool.c

On Fri, 27 Feb 2009, Danny Nicholas top posted:

 Agreed, but the OP seemed to be looking for a command-line solution, 
 not a C one.

The OP didn't specify what kind of solution they were looking for.

I wouldn't have considered introducing instability as a solution.

It seems a reasonable request. Maybe the OP would like to request a 
feature or offer a bounty?

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] call file concurrency

[James Sneeringer]

On Fri, Feb 27, 2009 at 4:14 AM, Christian Victor
christ...@victormedia.de wrote:
 2009/2/27 Bill Michaelson b...@cosi.com
 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?

 Afaik only by limiting the number of call files in the directory.

If you can get the outgoing directory (or a reaonable parent) on its
own mountable partition or volume, you could accomplish this with disk
quotas. It won't control how many Asterisk processes at once (does it
even handle them in parallel?), but it will control how many can
possibly be queued up waiting to be processed.

-James

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Re: [asterisk-users] call file concurrency

[Steve Edwards]

On Fri, 27 Feb 2009, James Sneeringer wrote:

 If you can get the outgoing directory (or a reaonable parent) on its own 
 mountable partition or volume, you could accomplish this with disk 
 quotas. It won't control how many Asterisk processes at once (does it 
 even handle them in parallel?), but it will control how many can 
 possibly be queued up waiting to be processed.

I'm thinking bad things could happen if a call fails (causing the call 
file to be appended) when you are at the limit. Also, this implies that 
the process creating the call files can handle the quota error. This also 
creates a bit of a land mine for the next admin when he replaces the 
failed disk with one without the quota.

I think it should be handled by munging the code in pbx_spool.c.

I took a casual peek at the (1.2) code this morning, so don't hold me to 
my opinions :)

The call file directory is scanned every once in a while and for each 
eligible call file, a detached thread is kicked off to handle it.

Limiting the number of concurrent threads (call files) would mean 
incrementing a [locked] counter as each thread is created and decrementing 
the [locked, non-zero] counter as each thread finishes.

Then, in the loop that scans the directory, if the counter is greater than 
the desired limit, exit the loop.

I'm sure there are more details to be worked out, but I think this could 
be done easily.

At the same time, it might be nice to add a feature to throttle the 
thread creation so that if a bunch of call files are dumped into the 
directory Asterisk doesn't spike trying to create concurrent-limit 
threads at once.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] call file concurrency

[Mik Cheez]

Steve Edwards wrote:
 On Fri, 27 Feb 2009, James Sneeringer wrote:
 
 If you can get the outgoing directory (or a reaonable parent) on its own 
 mountable partition or volume, you could accomplish this with disk 
 quotas. It won't control how many Asterisk processes at once (does it 
 even handle them in parallel?), but it will control how many can 
 possibly be queued up waiting to be processed.
 
 I'm thinking bad things could happen if a call fails (causing the call 
 file to be appended) when you are at the limit. Also, this implies that 
 the process creating the call files can handle the quota error. This also 
 creates a bit of a land mine for the next admin when he replaces the 
 failed disk with one without the quota.
 
 I think it should be handled by munging the code in pbx_spool.c.
 
 I took a casual peek at the (1.2) code this morning, so don't hold me to 
 my opinions :)
 
 The call file directory is scanned every once in a while and for each 
 eligible call file, a detached thread is kicked off to handle it.
 
 Limiting the number of concurrent threads (call files) would mean 
 incrementing a [locked] counter as each thread is created and decrementing 
 the [locked, non-zero] counter as each thread finishes.
 
 Then, in the loop that scans the directory, if the counter is greater than 
 the desired limit, exit the loop.
 
 I'm sure there are more details to be worked out, but I think this could 
 be done easily.
 
 At the same time, it might be nice to add a feature to throttle the 
 thread creation so that if a bunch of call files are dumped into the 
 directory Asterisk doesn't spike trying to create concurrent-limit 
 threads at once.
 
 Thanks in advance,
 
 Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline Fax: +1-760-731-3000
 
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Just my 2c, but what I've done in the past is modify the sleep function 
in asterisk from one based on seconds to one based on either 
milliseconds or nanoseconds (don't remember which).  Then I have a 
background daemon which looks to see how many files are in the 
directory, and if it's under threshold it pushes a new file from a queue 
into the directory.

Then, as they say above, you set the ulimit to something like 'ulimit -n 
10' or whatever it is you want.

Of course, the purpose for sending out a bazillion calls is another 
question . . . play nice.



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Re: [asterisk-users] call file concurrency

[Lenz Emilitri]

IIRC, some early dialler of the pre-AMI era used this technique to control
the number of calls placed simoultaneously - they just counted the number of
call files in the spool dir. As they are deleted when the call is over, this
was a simple way to do the throttling.

You could use a similar technique; have call files written to a staging
directory and then use a simple process to transfer them to the actual spool
dir so that there are never more than N in the spool dir.

Thanks

l.


2009/2/27 Bill Michaelson b...@cosi.com

 Is there a convenient way to limit the number of call files (outgoing
 directory) that are processed concurrently?

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[asterisk-users] call file concurrency

[Bill Michaelson]

Is there a convenient way to limit the number of call files (outgoing 
directory) that are processed concurrently?


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[asterisk-users] call file FXO channel problem

[Ray Chen]

I have problem of using call file to make auto outbound dial through FXO
channel. I put Channel: DAHDI/1/xx (xx is the
destination PSTN number to dial). For some reason asterisk did not dial
the number but the control came to the context that I defined in the call
file as if the peer had answed the call. It works if I change the channel
from DAHDI to a SIP channel like SIP/4567 or I dial DAHDI/1/xx
from a SIP channel. I am using asterisk1.4.23.1. Is it a bug in this
release? Thanks Ray

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Re: [asterisk-users] call file FXO channel problem

[Eric Wieling, Asteria Solutions Group]

Ray Chen wrote:
 I have problem of using call file to make auto outbound dial through FXO 
 channel. I put Channel: DAHDI/1/xx (xx is the destination 
 PSTN number to dial). For some reason asterisk did not dial the number but 
 the control came to the context that I defined in the call file as if the 
 peer had answed the call. It works if I change the channel from DAHDI to a 
 SIP channel like SIP/4567 or I dial DAHDI/1/xx from a SIP channel. I 
 am using asterisk1.4.23.1. Is it a bug in this release?
   
Analog FXO ports are considered answered as soon as dialing is 
finished.  The telco does not provide a signal to the calling device to 
indicate the far end answered the phone.  This does not apply to PRI or 
FXS.  Virtually all SIP service providers use PRIs.  If the service 
provider used analog you would also experience this when dialing using SIP.

-- 
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com


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[asterisk-users] call file bug?

[Ray Chen]

I have a problem of using call file to make an auto dial out call through
FXO channel. I defined the channel in the call file as Channel:
DAHDI/1/8775203463 When I put the call file under the
/var/spool/asterisk/outgoing dir it did not call out but came to the
context I defined in extensions.conf as if the callee had answered the
call. If I make a call from an extension to DAHDI/1/8775203463 it'll
success. . If I change the channel to SIP/8000 and put the call file
under /var/spool/asterisk/outgoing it is also success - it calls the
extension 8000 and the controle goes to the context after the extension
8000 answers the call. I am using asterisk 1.4.23.1. Is it a bug
introduced in this release? Thanks.

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Re: [asterisk-users] call file bug?

[Danny Nicholas]

You should post the call file.  Also, I'd use DAHDI/G1 instead of DAHDI/1 as
that ties the call to a specific port/line (perhaps what you want to do?)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ray Chen
Sent: Tuesday, February 17, 2009 2:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] call file bug?

 

I have a problem of using call file to make an auto dial out call through
FXO channel. I defined the channel in the call file as Channel:
DAHDI/1/8775203463 When I put the call file under the
/var/spool/asterisk/outgoing dir it did not call out but came to the context
I defined in extensions.conf as if the callee had answered the call. If I
make a call from an extension to DAHDI/1/8775203463 it'll success. . If I
change the channel to SIP/8000 and put the call file under
/var/spool/asterisk/outgoing it is also success - it calls the extension
8000 and the controle goes to the context after the extension 8000 answers
the call. I am using asterisk 1.4.23.1. Is it a bug introduced in this
release?

 

Thanks.

 


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Re: [asterisk-users] call file bug?

[Ex Vito]

On Tue, Feb 17, 2009 at 8:04 PM, Ray Chen ray1...@techie.com wrote:
 I have a problem of using call file to make an auto dial out call through
 FXO channel. I defined the channel in the call file as Channel:
 DAHDI/1/8775203463 When I put the call file under the
 /var/spool/asterisk/outgoing dir it did not call out but came to the context
 I defined in extensions.conf as if the callee had answered the call. If I
 make a call from an extension to DAHDI/1/8775203463 it'll success. . If I
 change the channel to SIP/8000 and put the call file under
 /var/spool/asterisk/outgoing it is also success - it calls the extension
 8000 and the controle goes to the context after the extension 8000 answers
 the call. I am using asterisk 1.4.23.1. Is it a bug introduced in this
 release?


  It's probably the result of FXO lines having very little signalling. IOW,
  asterisk picks up the FXO line and dials the number... By then it has
  no way of knowing wheather the other party answered or not. That's
  probably why your getting your other leg too early in the process.

  Maybe you could try to Wait() for a few seconds on your dialplan.
--
  exvito

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Re: [asterisk-users] Call file in the future

[didier.cuffaut]

First, thanks for your help

Ok, i going to do a script and call ot with only one 'System' (cf Gordon 
Henderson) and take a look to 'incron' (T Cohen)

Just need some explanations:

1) If the call file 'failed', an 'exitstatus' is happendGood 
How to check/get these $ and put in in an * $ ? (of course, the call file have 
to have archive= yes and go to 'outgoing-done')
sorry, i'm not a linux guru and it's not a pure Asterisk pb. Anyway, could 
someone show me the complete exact way and syntax to do this?

Using something as: $ egrep -vw (^#|^) file | awk -F   '{ print $2 }'  
(or some use of awk)

2) From my first post, are these lines  OK or wrong? (syntax error?)
  tmsp = the delay in future.. say 100 seconds

  exten= ra,n,System(NOW='date %S')

  exten= ra,n,System(let NOW=$NOW+$tmsp)

  exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec GMT+1 
+%Y%m%d%H%M. %S)NOTE THE 'M. %S'



  *

  or this way ?

   

  exten= ra,n,Set(touchtime=$[${EPOCH} + ${tmsp}])

  exten= ra,n,Set(TOUCH_TMSP=${STRFM(${touchtime},GMT+1,%C%y%m%d%H%M%S)

  * 


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Re: [asterisk-users] Call file in the future

[Steve Edwards]

On Mon, 19 Jan 2009, didier.cuffaut wrote:

 2) From my first post, are these lines OK or wrong? (syntax error?)

  tmsp = the delay in future.. say 100 seconds
  exten= ra,n,System(NOW='date %S')
  exten= ra,n,System(let NOW=$NOW+$tmsp)
  exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec GMT+1 
 +%Y%m%d%H%M. %S)   NOTE THE 'M. %S'

  *

  or this way ?

  exten= ra,n,Set(touchtime=$[${EPOCH} + ${tmsp}])
  exten= ra,n,Set(TOUCH_TMSP=${STRFM(${touchtime},GMT+1,%C%y%m%d%H%M%S)

  *

Each invocation of system() executes a separate process. The environment 
variables do not survive across processes. This method will not work.

Setting a channel variable and then passing it will work.

Your choices are to use system() or agi(). I'm leaning towards system() 
because the script/executable does not interact with Asterisk and may have 
value to you as a stand-alone command line utility.

You can write either in whatever language you are comfortable with. My 
sharpest tool is C but if execution speed is not important any scripting 
language (like shell) will do.

I'm a big fan of the getopt facility as it does all the nasty command line 
parsing for you so your utilities have a consistent, self-documenting look 
and feel. I even use it when I write AGIs. A year from now, which would 
you rather re-discover in your dialplan:

exten = 
s,n,agi(schedule-future-call,--archive,--max-retries=2,--offset=${TMSP},--retry-time=60,--wait-time=20)

or

exten = s,n,agi(schedule-future-call,${TMSP},60,,,20,a,,2)

I think I would pass the offset rather than the absolute. I don't like to 
clutter up my dialplans too much.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Call file in the future

[Tzafrir Cohen]

On Sat, Jan 17, 2009 at 08:06:08PM +0100, randulo wrote:
 On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr 
 wrote:
  May i have some comment/opinion on these two ways below to place a call file
  in the future ? (from the wiki and the asterisk book but added typos and
  stupidity come from me)
 
  The best is ?  (and should work ?)
 
 This is just me, but if I were going to program calls in the future I
 would just name them with the time (2009-01-17-20-08.call for four
 minutes from now, for example) and put them in a directory. The I'd
 have a cron job running that looked once per minute in that dir and
 did the mv if found file with that name.
 
 Does that make sense?

Run a program when something in the filesystem changes?

Looks like inotify can help you there. E.g.
http://packages.debian.org/sid/incron

-- 
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icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Call file in the future

[didier.cuffaut]

Hello,
 I read a thread on the asterisk dev list (call file handling suggestion)

May i have some comment/opinion on these two ways below to place a call file in 
the future ? (from the wiki and the asterisk book but added typos and stupidity 
come from me)

The best is ?  (and should work ?)

tmsp = the delay in future.. say 100 seconds

exten= ra,n,System(NOW='date %S')

exten= ra,n,System(let NOW=$NOW+$tmsp)

exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec GMT+1 
+%Y%m%d%H%M. %S)



*

or this way ?



exten= ra,n,Set(touchtime=$[${EPOCH} + ${tmsp}])

exten= ra,n,Set(TOUCH_TMSP=${STRFM(${touchtime},GMT+1,%C%y%m%d%H%M%S)

*



next step:

exten= ra,n,System(touch -t $TOUCH_TMSP /tmp/${idclient}.call))

exten= ra,n,System(mv /tmp/${idclient}.call /var/spool/asterisk/outgoing)





Thanks for your attention, happy 2009. and perhaps a reply ?
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Re: [asterisk-users] Call file in the future

[randulo]

On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr wrote:
 May i have some comment/opinion on these two ways below to place a call file
 in the future ? (from the wiki and the asterisk book but added typos and
 stupidity come from me)

 The best is ?  (and should work ?)

This is just me, but if I were going to program calls in the future I
would just name them with the time (2009-01-17-20-08.call for four
minutes from now, for example) and put them in a directory. The I'd
have a cron job running that looked once per minute in that dir and
did the mv if found file with that name.

Does that make sense?

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Re: [asterisk-users] Call file in the future

[Gordon Henderson]

On Sat, 17 Jan 2009, randulo wrote:

 On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr 
 wrote:
 May i have some comment/opinion on these two ways below to place a call file
 in the future ? (from the wiki and the asterisk book but added typos and
 stupidity come from me)

 The best is ?  (and should work ?)

 This is just me, but if I were going to program calls in the future I
 would just name them with the time (2009-01-17-20-08.call for four
 minutes from now, for example) and put them in a directory. The I'd
 have a cron job running that looked once per minute in that dir and
 did the mv if found file with that name.

 Does that make sense?

Not to me.

Cron jobs can be delayed, servers can be rebooted and you're suggesting a 
solution for a problem that already has a solution - ie. set the access 
time of the file in the future which is what didier.cuffaut is trying to 
do.

For didier.cuffaut: the Asterisk command System() calls the system routine 
system() which will fork a shell which will then fork to execute your 
command. It's more efficient to put all the commands in one script and 
then use System() to execute that file.

   System(/path/to/script 100)

Although personally, I'd probably write a C (or php or perl) program to 
make things as efficient as possible than use a shell script (or shell 
commands), to avoid all the forking, but maybe that's just me.

Gordon

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Re: [asterisk-users] Call file in the future

[Tilghman Lesher]

On Saturday 17 January 2009 13:49:16 Gordon Henderson wrote:
 On Sat, 17 Jan 2009, randulo wrote:
  On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr 
wrote:
  May i have some comment/opinion on these two ways below to place a call
  file in the future ? (from the wiki and the asterisk book but added
  typos and stupidity come from me)
 
  The best is ?  (and should work ?)
 
  This is just me, but if I were going to program calls in the future I
  would just name them with the time (2009-01-17-20-08.call for four
  minutes from now, for example) and put them in a directory. The I'd
  have a cron job running that looked once per minute in that dir and
  did the mv if found file with that name.
 
  Does that make sense?

 Not to me.

 Cron jobs can be delayed, servers can be rebooted and you're suggesting a
 solution for a problem that already has a solution - ie. set the access
 time of the file in the future which is what didier.cuffaut is trying to
 do.

Almost.  It's actually the modified timestamp, not the access timestamp.  You
cannot usually alter the access timestamp on filesystems, as this is part of
the security audit trail.  You can use the touch(1) system utility to set the
modified timestamp to whatever you like.

-- 
Tilghman

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[asterisk-users] .call file not updating MySQL CDR's

[cbbs70a]

All;
   I have implemented an autodialer solution where I create .call files for 
each number to be dialed. The .call file is very simple in design:

Channel: SIP/2405551...@broadvoice-outbound
Context: autodial
MaxRetries: 5
RetryTime: 600
WaitTime: 60
CallerID: Hildas Cleaning 410555
Extension: 2405551212
Priority: 1
Account: 999
Archive: yes


A problem recently started where the MySQL records were not being written 
(after 6 months of working flawlessly) although all other CDR records were 
fine. The solution turned out to be where I had to *downgrade* Asterisk from 
1.4.22 to 1.4.20.1. Nothing else in the configuration changed. Did I come 
across a bug, a feature, or what? Did I miss something obvious?
Thanks
FSD
 


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Re: [asterisk-users] .call file not updating MySQL CDR's

[Leif Madsen]

You can follow this issue here:

http://bugs.digium.com/view.php?id=14167

Your best bet in the future is to check the bug tracker for any issues 
you may have to see if it has already been reported.

Thanks!
Leif Madsen.

cbbs...@hotmail.com wrote:
 A problem recently started where the MySQL records were not being 
 written (after 6 months of working flawlessly) although all other CDR 
 records were fine. The solution turned out to be where I had to 
 *downgrade* Asterisk from 1.4.22 to 1.4.20.1. Nothing else in the 
 configuration changed. Did I come across a bug, a feature, or what? Did 
 I miss something obvious?


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[asterisk-users] Call file IAX Trunk: Call Failed, Reason 0

[Raveen Siddiqui]

Hello,

I am new to Asterisk; I did go through a lot of documentation, wikis, 
and the O'Reilly book and have most of what I need now working well. 
I do have a problem that I keep bumping heads against, however: I can 
dial out very well through a IAX2 trunk (9 followed by number), but 
if I specify the same IAX2 trunk in the Channel of a .call file, the 
call does not go through.

Here is my test call file:

Channel: IAX2/providername/14165551212
MaxRetries: 2
RetryTime: 20
WaitTime: 30
Application: Playback
Data: hello-world

The call is said to have failed with reason 0; the provider lists a 
0-second call. Does anyone know what's going on ? (of course, the 
very same call file with an internal extension works perfectly well)

Thank you !


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Re: [asterisk-users] Call file IAX Trunk: Call Failed, Reason 0

[Raveen Siddiqui]

Please disregard. I have taken two debug dumps on a successful and a 
failed communication through the trunk; the only difference was in 
the absence of CallerID ! Now, this works well. I can say hello to 
the whole world.

At 06:01 PM 8/12/2007, you wrote:

Hello,

I am new to Asterisk; I did go through a lot of documentation, wikis,
and the O'Reilly book and have most of what I need now working well.
I do have a problem that I keep bumping heads against, however: I can
dial out very well through a IAX2 trunk (9 followed by number), but
if I specify the same IAX2 trunk in the Channel of a .call file, the
call does not go through.

Here is my test call file:

Channel: IAX2/providername/14165551212
MaxRetries: 2
RetryTime: 20
WaitTime: 30
Application: Playback
Data: hello-world

The call is said to have failed with reason 0; the provider lists a
0-second call. Does anyone know what's going on ? (of course, the
very same call file with an internal extension works perfectly well)

Thank you !


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Re: [asterisk-users] .call file and logging

[Vieri]

EDIT:
It seems that if the call fails then I can see the
number in the cdr-lastdata (and lastapp) fields (eg.
Dial Zap/g1/7032).
However, if the call is answered, there's no trace of
the number if Zap was used (there's a trace only if
SIP is used).
So, how can I hack this so that I can set the number
up myself so that if a Zap call succeeds then
cdr-lastadata and/or lastapp will somehow contain the
extension that was dialed through this channel?
Right now, cdr-lastapp is Hangup, as expected, and
cdr-lastdata is empty. How could I add the number
that was dialed to the empty cdr-lastdata?

--- Vieri [EMAIL PROTECTED] wrote:

 I am writing a cron script to check if certain
 extensions are online and if they aren't then
 Asterisk
 creates a couple of .call files to notify another
 set
 of extensions or external numbers.
 
 It works fine except for logging information.
 
 What I'm doing in the script is setting a fake
 caller ID (as it's generated by Asterisk, not by a
 user) and calling out real users.
 
 So the user's extension is specified in the
 Channel:
  field. When the user picks the phone up, asterisk
 drops into the custom_NOTIFY context which plays a
 menu.
 
 My problem is that when I check the logs in /var/log
 or in the MySQL CDR database, I can't always
 demonstrate that Asterisk actually called a specific
 number (in the code below, the number I need to log
 is
 $alerts).
 
 If I use a SIP extension in the Channel field then
 the
 logging works for me because I can see that the
 SIP/EXTEN was used (see below).
 
 However, if I use a Zap extension then only the Zap
 channel number is logged but the extension's number
 isn't (in the example below, 7022 does not appear in
 the logs).
 
 Any suggestions as to how I can solve this?
 Maybe by changing the Extension: line or setting
 variables. A quick simple example would be
 appreciated.
 
 Thanks,
 
 Vieri
 
 Code snippet:
 
 $ftime = time();
 $fname = /tmp/asterisk_.$ftime..call;
 $fname_call =

/var/spool/asterisk/outgoing/asterisk_.$ftime..call;
 $fd = fopen($fname, 'w');
 fwrite($fd, Channel: .$alerts.\n);
 fwrite($fd, Callerid: IT 7021\n);
 fwrite($fd, Set: FHMNUM=.$FAILURES.\n);
 fwrite($fd, MaxRetries: 2\n);
 fwrite($fd, RetryTime: 20\n);
 fwrite($fd, WaitTime: 40\n);
 fwrite($fd, Context: custom-NOTIFY\n);
 fwrite($fd, Extension: s\n);
 fwrite($fd, Priority: 1\n);
 fclose($fd);
 chown($fname,asterisk);
 chgrp($fname,asterisk);
 rename($fname,$fname_call);
 
 # cat cdr_custom.conf
 ;
 ; Mappings for custom config file
 ;
 [mappings]
 Master.csv =

${CDR(clid)},${CDR(src)},${CDR(dst)},${CDR(dcontext)},${CDR(channel)},${CDR(dstchannel)},${CDR(lastapp)},${CDR(lastdata)},${CDR(start)},${CDR(answer)},${CDR(end)},${CDR(duration)},${CDR(billsec)},${CDR(disposition)},${CDR(amaflags)},${CDR(accountcode)},${CDR(uniqueid)},${CDR(userfield)}
 
 If $alerts is Zap/g1/7022 Then:
 # tail /var/log/asterisk/cdr-csv/Master.csv
 ,7021,s,custom-NOTIFY,IT
 7021,Zap/2-1,,Hangup,,2007-08-07
 13:30:19,2007-08-07 13:30:19,2007-08-07
 13:30:26,7,7,ANSWERED,DOCUMENTATION
 
 If $alerts is SIP/4053 Then:
 # tail /var/log/asterisk/cdr-csv/Master.csv
 ,7021,s,custom-NOTIFY,IT

7021,SIP/4053-0829b6a0,,Hangup,,2007-08-07
 12:58:02,2007-08-07 12:58:02,2007-08-07
 12:58:15,13,13,ANSWERED,DOCUMENTATION
 
 (This is an Asterisk/FreePBX system)



   

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[asterisk-users] .call file and logging

[Vieri]

I am writing a cron script to check if certain
extensions are online and if they aren't then Asterisk
creates a couple of .call files to notify another set
of extensions or external numbers.

It works fine except for logging information.

What I'm doing in the script is setting a fake
caller ID (as it's generated by Asterisk, not by a
user) and calling out real users.

So the user's extension is specified in the Channel:
 field. When the user picks the phone up, asterisk
drops into the custom_NOTIFY context which plays a
menu.

My problem is that when I check the logs in /var/log
or in the MySQL CDR database, I can't always
demonstrate that Asterisk actually called a specific
number (in the code below, the number I need to log is
$alerts).

If I use a SIP extension in the Channel field then the
logging works for me because I can see that the
SIP/EXTEN was used (see below).

However, if I use a Zap extension then only the Zap
channel number is logged but the extension's number
isn't (in the example below, 7022 does not appear in
the logs).

Any suggestions as to how I can solve this?
Maybe by changing the Extension: line or setting
variables. A quick simple example would be
appreciated.

Thanks,

Vieri

Code snippet:

$ftime = time();
$fname = /tmp/asterisk_.$ftime..call;
$fname_call =
/var/spool/asterisk/outgoing/asterisk_.$ftime..call;
$fd = fopen($fname, 'w');
fwrite($fd, Channel: .$alerts.\n);
fwrite($fd, Callerid: IT 7021\n);
fwrite($fd, Set: FHMNUM=.$FAILURES.\n);
fwrite($fd, MaxRetries: 2\n);
fwrite($fd, RetryTime: 20\n);
fwrite($fd, WaitTime: 40\n);
fwrite($fd, Context: custom-NOTIFY\n);
fwrite($fd, Extension: s\n);
fwrite($fd, Priority: 1\n);
fclose($fd);
chown($fname,asterisk);
chgrp($fname,asterisk);
rename($fname,$fname_call);

# cat cdr_custom.conf
;
; Mappings for custom config file
;
[mappings]
Master.csv =
${CDR(clid)},${CDR(src)},${CDR(dst)},${CDR(dcontext)},${CDR(channel)},${CDR(dstchannel)},${CDR(lastapp)},${CDR(lastdata)},${CDR(start)},${CDR(answer)},${CDR(end)},${CDR(duration)},${CDR(billsec)},${CDR(disposition)},${CDR(amaflags)},${CDR(accountcode)},${CDR(uniqueid)},${CDR(userfield)}

If $alerts is Zap/g1/7022 Then:
# tail /var/log/asterisk/cdr-csv/Master.csv
,7021,s,custom-NOTIFY,IT
7021,Zap/2-1,,Hangup,,2007-08-07
13:30:19,2007-08-07 13:30:19,2007-08-07
13:30:26,7,7,ANSWERED,DOCUMENTATION

If $alerts is SIP/4053 Then:
# tail /var/log/asterisk/cdr-csv/Master.csv
,7021,s,custom-NOTIFY,IT
7021,SIP/4053-0829b6a0,,Hangup,,2007-08-07
12:58:02,2007-08-07 12:58:02,2007-08-07
12:58:15,13,13,ANSWERED,DOCUMENTATION

(This is an Asterisk/FreePBX system)



   

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[asterisk-users] .call file problem

[Nitesh Divecha]

Hello All,

Something strange I found that my .call file is running twice...
Just after 60 sec it will run again, without any application invoking it.

This is my .call file: -
=
Channel: SIP/xo-out/19097773456
Callerid: 9097773456
MaxRetries: 3
RetryTime: 30
WaitTime: 15
Context: custom-900
Extension: 900
Priority: 1

I am running Asterisk 1.2.18 on CentOS 4.5.

Anyone can help?

Cheers,
Nitesh



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Re: [asterisk-users] .call file problem

[Atis]

Is your .call file writable by asterisk?

$ chmod 777 sample.call

On 7/31/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
 Hello All,

 Something strange I found that my .call file is running twice...
 Just after 60 sec it will run again, without any application invoking it.

 This is my .call file: -
 =
 Channel: SIP/xo-out/19097773456
 Callerid: 9097773456
 MaxRetries: 3
 RetryTime: 30
 WaitTime: 15
 Context: custom-900
 Extension: 900
 Priority: 1

 I am running Asterisk 1.2.18 on CentOS 4.5.

 Anyone can help?

 Cheers,
 Nitesh



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-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org

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Re: [asterisk-users] .call file problem

[Nitesh Divecha]

Thanks Atis,

Yes and the .call executes fine... but after 60 seconds it executes 
again automatically without any application executing it.

Cheers,
Nitesh



Atis wrote:
 Is your .call file writable by asterisk?

 $ chmod 777 sample.call

 On 7/31/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
   
 Hello All,

 Something strange I found that my .call file is running twice...
 Just after 60 sec it will run again, without any application invoking it.

 This is my .call file: -
 =
 Channel: SIP/xo-out/19097773456
 Callerid: 9097773456
 MaxRetries: 3
 RetryTime: 30
 WaitTime: 15
 Context: custom-900
 Extension: 900
 Priority: 1

 I am running Asterisk 1.2.18 on CentOS 4.5.

 Anyone can help?

 Cheers,
 Nitesh



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Re: [asterisk-users] .call file problem

[Eric \ManxPower\ Wieling]

Make sure you have a blank line at the end of your .call file.

Nitesh Divecha wrote:
 Hello All,
 
 Something strange I found that my .call file is running twice...
 Just after 60 sec it will run again, without any application invoking it.
 
 This is my .call file: -
 =
 Channel: SIP/xo-out/19097773456
 Callerid: 9097773456
 MaxRetries: 3
 RetryTime: 30
 WaitTime: 15
 Context: custom-900
 Extension: 900
 Priority: 1
 
 I am running Asterisk 1.2.18 on CentOS 4.5.
 
 Anyone can help?
 
 Cheers,
 Nitesh
 
 
 
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Re: [asterisk-users] .call file problem

[Balu Raman]

No, Atis means 'make it writable'. .call should be removable after
execution.
- balu raman

On 7/31/07, Nitesh Divecha [EMAIL PROTECTED] wrote:

 Thanks Atis,

 Yes and the .call executes fine... but after 60 seconds it executes
 again automatically without any application executing it.

 Cheers,
 Nitesh



 Atis wrote:
  Is your .call file writable by asterisk?
 
  $ chmod 777 sample.call
 
  On 7/31/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
 
  Hello All,
 
  Something strange I found that my .call file is running twice...
  Just after 60 sec it will run again, without any application invoking
 it.
 
  This is my .call file: -
  =
  Channel: SIP/xo-out/19097773456
  Callerid: 9097773456
  MaxRetries: 3
  RetryTime: 30
  WaitTime: 15
  Context: custom-900
  Extension: 900
  Priority: 1
 
  I am running Asterisk 1.2.18 on CentOS 4.5.
 
  Anyone can help?
 
  Cheers,
  Nitesh
 
 
 
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Re: [asterisk-users] .call file problem

[Nitesh Divecha]

Thanks Eric,

It solved the problem by having a blank line... I wonder why we need a 
blank line...

Cheers,
Nitesh



Eric ManxPower Wieling wrote:
 Make sure you have a blank line at the end of your .call file.

 Nitesh Divecha wrote:
   
 Hello All,

 Something strange I found that my .call file is running twice...
 Just after 60 sec it will run again, without any application invoking it.

 This is my .call file: -
 =
 Channel: SIP/xo-out/19097773456
 Callerid: 9097773456
 MaxRetries: 3
 RetryTime: 30
 WaitTime: 15
 Context: custom-900
 Extension: 900
 Priority: 1

 I am running Asterisk 1.2.18 on CentOS 4.5.

 Anyone can help?

 Cheers,
 Nitesh



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Re: [asterisk-users] .call file problem

[Eric \ManxPower\ Wieling]

Config file parsers frequently read until end of line where end of 
line is a CR or LF.  No CR or LF, no last line read. I think that bug 
was fixed fairly recently in Asterisk.

Other software has similar issues in the past.

Nitesh Divecha wrote:
 Thanks Eric,
 
 It solved the problem by having a blank line... I wonder why we need a 
 blank line...
 
 Cheers,
 Nitesh
 
 
 
 Eric ManxPower Wieling wrote:
 Make sure you have a blank line at the end of your .call file.

 Nitesh Divecha wrote:
   
 Hello All,

 Something strange I found that my .call file is running twice...
 Just after 60 sec it will run again, without any application invoking it.

 This is my .call file: -
 =
 Channel: SIP/xo-out/19097773456
 Callerid: 9097773456
 MaxRetries: 3
 RetryTime: 30
 WaitTime: 15
 Context: custom-900
 Extension: 900
 Priority: 1

 I am running Asterisk 1.2.18 on CentOS 4.5.

 Anyone can help?

 Cheers,
 Nitesh



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Re: [asterisk-users] .call file

[Kevin Smith]

Paul wrote:
 I'm going to top post in this situation.

 Kevin - Commands that operate on the channel variables won't help if we
 are using a call file. We will have a new channel.
   
Agreed, I misread and thought he was trying to generate a call file.

-Kevin
 This syntax works with asterisk 1.2.x for me:

 Application: AGI
 Data: say_it.php|call_status_message

 I have done other things where a bunch of parameters are stored in
 postgres or mysql and the only parameter I pass via the call file is the
 record key. The php script receives the key as a parameter and gets
 everything else from the db. Something like this:

 Application: AGI
 Data: inform.php|68456943

 Kevin Smith wrote:

   
 Nitesh Divecha wrote:
  

 
 Hello All,

 Is there any way to pass additional parameters while calling AGI from 
 *.call file?

 Channel: Local/[EMAIL PROTECTED]
 MaxRetries: 0
 RetryTime: 15
 WaitTime: 15
 Application: AGI
 Data: recordvoice.php

 Something like Data: recordvoice.php?id=3453name=asterisk

 Cheers,
 Nitesh



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 I'm not 100% sure if you can pass it directly, but you can use the set 
 option in the call file to set local variables within Asterisk and then 
 pass them to the AGI script. So for your example it would be.

 Set: name=asterisk

 This will set the variable ${name} in asterisk and depending how your 
 script was created you should be able to grab the variable to use within 
 the script. If you are using say the PHP AGI you can use something like 
 the following:

 $var = $agi-get_variable(name);

 This will create an array with $var['data'] holding 'asterisk';

 Now one more thing I am not sure of is for multiple variables (haven't 
 tried it yet ;D ). You may have to do it one of two ways.

 Set: name=asterisk, id=3453

 or

 Set: name=asterisk
 Set: id=3453

 and if those don't work, just format it so you can filter it out with PHP.

 Hopefully this will help.

 Kevin



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Re: [asterisk-users] .call file

[Kevin Smith]

Nitesh Divecha wrote:
 Hello All,

 Is there any way to pass additional parameters while calling AGI from 
 *.call file?

 Channel: Local/[EMAIL PROTECTED]
 MaxRetries: 0
 RetryTime: 15
 WaitTime: 15
 Application: AGI
 Data: recordvoice.php

 Something like Data: recordvoice.php?id=3453name=asterisk

 Cheers,
 Nitesh



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I'm not 100% sure if you can pass it directly, but you can use the set 
option in the call file to set local variables within Asterisk and then 
pass them to the AGI script. So for your example it would be.

Set: name=asterisk

This will set the variable ${name} in asterisk and depending how your 
script was created you should be able to grab the variable to use within 
the script. If you are using say the PHP AGI you can use something like 
the following:

$var = $agi-get_variable(name);

This will create an array with $var['data'] holding 'asterisk';

Now one more thing I am not sure of is for multiple variables (haven't 
tried it yet ;D ). You may have to do it one of two ways.

Set: name=asterisk, id=3453

or

Set: name=asterisk
Set: id=3453

and if those don't work, just format it so you can filter it out with PHP.

Hopefully this will help.

Kevin



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Re: [asterisk-users] .call file

[Paul]

I'm going to top post in this situation.

Kevin - Commands that operate on the channel variables won't help if we
are using a call file. We will have a new channel.

This syntax works with asterisk 1.2.x for me:

Application: AGI
Data: say_it.php|call_status_message

I have done other things where a bunch of parameters are stored in
postgres or mysql and the only parameter I pass via the call file is the
record key. The php script receives the key as a parameter and gets
everything else from the db. Something like this:

Application: AGI
Data: inform.php|68456943

Kevin Smith wrote:

Nitesh Divecha wrote:
  

Hello All,

Is there any way to pass additional parameters while calling AGI from 
*.call file?

Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php

Something like Data: recordvoice.php?id=3453name=asterisk

Cheers,
Nitesh



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I'm not 100% sure if you can pass it directly, but you can use the set 
option in the call file to set local variables within Asterisk and then 
pass them to the AGI script. So for your example it would be.

Set: name=asterisk

This will set the variable ${name} in asterisk and depending how your 
script was created you should be able to grab the variable to use within 
the script. If you are using say the PHP AGI you can use something like 
the following:

$var = $agi-get_variable(name);

This will create an array with $var['data'] holding 'asterisk';

Now one more thing I am not sure of is for multiple variables (haven't 
tried it yet ;D ). You may have to do it one of two ways.

Set: name=asterisk, id=3453

or

Set: name=asterisk
Set: id=3453

and if those don't work, just format it so you can filter it out with PHP.

Hopefully this will help.

Kevin



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[asterisk-users] .call file

[Nitesh Divecha]

Hello All,

Is there any way to pass additional parameters while calling AGI from 
*.call file?

Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php

Something like Data: recordvoice.php?id=3453name=asterisk

Cheers,
Nitesh



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[asterisk-users] .call file

[Jerry Geis]

You can certainly use variables in the call file that get passed to the AGI.
SetVar: MyVar=44


jerry

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Re: [asterisk-users] .call file

[Nitesh Divecha]

Thanks Jerry,

But how can I access the Set variable in my AGI file?

Like I do for callerId $cidnum = $agi-request['agi_callerid'];

Is there any for Set?

Cheers,
Nitesh



Jerry Geis wrote:
 You can certainly use variables in the call file that get passed to the AGI.
 SetVar: MyVar=44


 jerry

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[asterisk-users] .call file

[Jerry Geis]

Thanks Jerry,

But how can I access the Set variable in my AGI file?

Like I do for callerId $cidnum = $agi-request['agi_callerid'];

Is there any for Set?

Cheers,
Nitesh

I dont use that programming (php) - I use C.
I ask the AGI printf(Get variable name\n\r);
and if gives it back to me.

use voip-info.org search for setvar and agi.

Jerry





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[asterisk-users] call file vs. originate

[Nathan Bell]

I'm having trouble getting the manager interface to behave properly; 
specifically the Originate event.


If I create an originate event as below, the calling phone will 
auto-answer (as it's supposed to) but the receiving phone never rings. 
It will timeout at 20 seconds.


Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe 201

This is rather odd, as if I create a nearly identical call file in 
/var/spool/asterisk/outgoing (below) the receiving phone rings correctly.


Channel: Local/[EMAIL PROTECTED]
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe 201

They both use the same contexts, but the result is drastically 
different. Any thoughts on how to remedy the problem?


Here is are the two contexts from extensions.conf:

; from sip lines
[from-sip]
include = internal

[from-sip2]
exten = _X.,1,SIPAddHeader(Alert-Info: AA)
exten = _X.,n,Dial(SIP/${EXTEN},200,o)
exten = _X.,n,Hangup()

; generic interal route
[internal]
exten = s,1,Answer()
exten = 500,1,Macro(voicemail)
include = parkedcalls
include = cac-ext
include = sip-ext
include = intertel-ext
include = to-ptsn

(cac-ext, sip-ext, intertel-ext and to-ptsn route the calls to our 
channel bank, sip phones, intertel pbx, and the outside world respectively.)


Below lies the results given over the manager interface:

Response: Success
Message: Originate successfully queued

Event: Newchannel
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
State: Ring
CallerID: unknown
CallerIDName: unknown
Uniqueid: 1175271459.2289

Event: Newchannel
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
State: Down
CallerID: unknown
CallerIDName: unknown
Uniqueid: 1175271459.2288

Event: Newcallerid
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2288
CID-CallingPres: 0 (Presentation Allowed, Not Screened)

Event: Newcallerid
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2288
CID-CallingPres: 0 (Presentation Allowed, Not Screened)

Event: Newcallerid
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2289
CID-CallingPres: 0 (Presentation Allowed, Not Screened)

Event: Newexten
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
Context: from-sip2
Extension: 201
Priority: 1
Application: SIPAddHeader
AppData: Alert-Info: AA
Uniqueid: 1175271459.2289

Event: Newexten
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
Context: from-sip2
Extension: 201
Priority: 2
Application: Dial
AppData: SIP/201|200|o
Uniqueid: 1175271459.2289

Event: Newchannel
Privilege: call,all
Channel: SIP/201-08217eb0
State: Down
CallerID: unknown
CallerIDName: unknown
Uniqueid: 1175271459.2290

Event: Dial
Privilege: call,all
Source: Local/[EMAIL PROTECTED],2
Destination: SIP/201-08217eb0
CallerID: 201
CallerIDName: Fake Name
SrcUniqueID: 1175271459.2289
DestUniqueID: 1175271459.2290

Event: Newstate
Privilege: call,all
Channel: SIP/201-08217eb0
State: Ringing
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2290

Event: Newstate
Privilege: call,all
Channel: SIP/201-08217eb0
State: Up
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2290

Event: Newstate
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
State: Up
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2289

Event: Link
Privilege: call,all
Channel1: Local/[EMAIL PROTECTED],2
Channel2: SIP/201-08217eb0
Uniqueid1: 1175271459.2289
Uniqueid2: 1175271459.2290
CallerID1: 201
CallerID2: 201

Event: Newstate
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
State: Up
CallerID: 201
CallerIDName: Fake Name
Uniqueid: 1175271459.2288

Event: Newexten
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1
Context: from-sip
Extension: s
Priority: 1
Application: Answer
AppData:
Uniqueid: 1175271459.2288

Event: Rename
Privilege: call,all
Oldname: SIP/201-08217eb0
Newname: SIP/201-08217eb0MASQ
Uniqueid: 1175271459.2290

Event: Rename
Privilege: call,all
Oldname: Local/[EMAIL PROTECTED],1
Newname: SIP/201-08217eb0
Uniqueid: 1175271459.2288

Event: Rename
Privilege: call,all
Oldname: SIP/201-08217eb0MASQ
Newname: Local/[EMAIL PROTECTED],1ZOMBIE
Uniqueid: 1175271459.2290

Event: Unlink
Privilege: call,all
Channel1: Local/[EMAIL PROTECTED],2
Channel2: Local/[EMAIL PROTECTED],1ZOMBIE
Uniqueid1: 1175271459.2289
Uniqueid2: 1175271459.2290
CallerID1: 201
CallerID2: 201

Event: Hangup
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],1ZOMBIE
Uniqueid: 1175271459.2290
Cause: 16
Cause-txt: Normal Clearing

Event: Hangup
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
Uniqueid: 1175271459.2289
Cause: 16
Cause-txt: Normal Clearing

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Re: [asterisk-users] call file vs. originate

[Richard Lyman]

Nathan Bell wrote:
I'm having trouble getting the manager interface to behave properly; 
specifically the Originate event.


If I create an originate event as below, the calling phone will 
auto-answer (as it's supposed to) but the receiving phone never rings. 
It will timeout at 20 seconds.


Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe 201

This is rather odd, as if I create a nearly identical call file in 
/var/spool/asterisk/outgoing (below) the receiving phone rings correctly.


Channel: Local/[EMAIL PROTECTED]
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe 201

*snipped

depending on your manager.c

you will find that manager originate need 'Exten: ..' not 'Extension: ..'
meaning, if you attempt to use 'Extension: ..' it will autofallthru (if 
set) to 's' extension in dialplan.


good luck

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Re: [asterisk-users] call file vs. originate

[Nathan Bell]

After fixing some issues with our pbx dial plan that worked great.

Thanks,

Nathan Bell
IT Engineer
Action Target, Inc.

Richard Lyman wrote:


Nathan Bell wrote:

I'm having trouble getting the manager interface to behave properly; 
specifically the Originate event.


If I create an originate event as below, the calling phone will 
auto-answer (as it's supposed to) but the receiving phone never 
rings. It will timeout at 20 seconds.


Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe 201

This is rather odd, as if I create a nearly identical call file in 
/var/spool/asterisk/outgoing (below) the receiving phone rings 
correctly.


Channel: Local/[EMAIL PROTECTED]
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe 201


*snipped

depending on your manager.c

you will find that manager originate need 'Exten: ..' not 'Extension: ..'
meaning, if you attempt to use 'Extension: ..' it will autofallthru 
(if set) to 's' extension in dialplan.


good luck

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[asterisk-users] Call file: CallerID problem

[Arik Raffael Funke]

Hello,

I have the following call file:

Channel: Local/[EMAIL PROTECTED]/n
Callerid: 27
MaxRetries: 2
RetryTime: 10
Context: test2
Extension: s

And the following dialplan:

[test1]
exten = s,1,NoOp(${CALLERIDNUM})


But my CALLERIDNUM and CALLERIDNAME variables are both empty. I tried 
without success following alternatives in the call file:

- Callerid: Someone 27
- SetVar: CALLERID(number)=27

Can anybody tell me, how to set the callerID for outgoing calls realised 
with call files? I am using asterisk 1.2.


Thanks in advance for any hints.

Regards,
Arik

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Re: [asterisk-users] Call file: CallerID problem

[Doug Lytle]

Arik Raffael Funke wrote:

Hello,

I have the following call file:

Channel: Local/[EMAIL PROTECTED]/n
Callerid: 27


Caller id format is:

CallerID: SomeName SomeNumber  For example, I use:

CallerID: VM-System 4200

Doug

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[asterisk-users] call file mechanism

[K Kuo]

Hi list,
I have a call file as following and it works. But, I don't really understand 
its mechanism.
The SIP/voipbuster is a sip trunk which I set up in freePBX with voipbuster 
account. And 2874 is one of my extension which was assigned to x-lite 
client.
When I place this call file in outgoing folder, it is able to dial out my 
home phone at 001xx. However, the Dst in call logs show 2874 or s 
instead of my phone number. Why sometimes 2874, sometimes s? and why not 
my phone number?


My interpretation is the call file actually call extension 2874 and place 
a out going call via 2874. If I am right, does it mean any outgoing call has 
to be placed through an extension. How can I manipulate this call file in 
order to show my home phone as destination instead of extension number.

Thank you very much.

Channel: SIP/voipbuster/001xx
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: outgoing
Extension: 2874
Priority: 1

Thanks in advance!!

_
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Re: [asterisk-users] Call file do 2 outbound call

[Jean-Michel Hiver]

Daniel Hikel a écrit :


Hello,

I am not so really familar with asterisk at the moment, but i am working
hard on it. Please could anybody advise me how to write a call file for the
queue to do 2 outbounds call and connect both via my SIP interface. 
 


Why not write two .call files if you want two calls?

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[asterisk-users] Call file do 2 outbound call

[Daniel Hikel]

Hello,

I am not so really familar with asterisk at the moment, but i am working
hard on it. Please could anybody advise me how to write a call file for the
queue to do 2 outbounds call and connect both via my SIP interface. 

Thanks in advance.

Daniel


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[asterisk-users] Call file do 2 outbound call

[Daniel Hikel]

Hello,

I am not so really familar with asterisk at the moment, but i am working
hard on it. Please could anybody advise me how to write a call file for the
queue to do 2 outbounds call and connect both via my SIP interface. 

Thanks in advance.

Daniel


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[Asterisk-Users] call file result

[Mimmus]

Hi,
is there a way to 'manage' result of a call file (NOANSWER, BUSY, max
attempts, etc) put under /var/spool/asterisk/outgoing?

Thanks
Mimmus

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Re: [Asterisk-Users] call file result

[Ben Higley]

You could just route the call files through their own context, and using
some smart scripting, just write the output of DIALSTATUS to a file ..
thus it would only write to a file somewhere, when something happends in
that context.. keeping it separate from the rest of your dialplan.



 Hi,
 is there a way to 'manage' result of a call file (NOANSWER, BUSY, max
 attempts, etc) put under /var/spool/asterisk/outgoing?

 Thanks
 Mimmus

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RE: [Asterisk-Users] call file result

[Mimmus]

Thank you for your response.
I found also this:
 If the call is not answered, and the standard extension failed with
priority 1 exists in the same context, control will jump there.
in thw wiki: http://www.voip-info.org/wiki-Asterisk+auto-dial+out


M.


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ben Higley
 Sent: Saturday, January 14, 2006 4:54 PM

 You could just route the call files through their own 
 context, and using some smart scripting, just write the 
 output of DIALSTATUS to a file ..
 thus it would only write to a file somewhere, when something 
 happends in that context.. keeping it separate from the rest 
 of your dialplan.
 
 
 
  Hi,
  is there a way to 'manage' result of a call file (NOANSWER, 
 BUSY, max 
  attempts, etc) put under /var/spool/asterisk/outgoing?
 
  Mimmus

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[Asterisk-Users] Call file always redials (grrrrr)

[Remco Barende]

Hi list!

Our CRM app is creating call files for outgoing calls which is working
great I just have one problem.

I am using this as my call file:
Channel: SIP/228(my phone)
MaxRetries: 0
Context: from-internal  (the context to dial from)
Extension: 003120531234 (the phone number)
Priority: 1
Callerid: Myfinecustomer 003120531234

so the external number is connected to my sip phone. However after 
speaking for approx 5 minuted, Asterisk always does a retry and I 
see the external number in my display on the second line. It does

this on every call. When I'm finished I also see 2
records in the log files.

Any idea why Asterisk is trying to place the call again even though the 
first attempt was succesful and the call is still in progress?


I didn't specify a redial anywhere. I'm running the latest cvs stable (of 
this morning),


Thanks!
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[Asterisk-Users] Call file ][ Unable to request channel ZAP/g1/0123456789 ][ Call failed to go through, reason 0

[Paul van Brouwershaven]

Hello,

I have a problem with sending multiple calls from the outgoing call queue.

Everything is going ok, but when I move more then 5 files in the queue the 
following notice messages are shown on the screen:


Jul 13 07:58:32 NOTICE[7597]: channel.c:1817 __ast_request_and_dial: 
Unable to request channel ZAP/g1/0123456789


Jul 13 07:58:32 NOTICE[7597]: pbx_spool.c:232 attempt_thread: Call failed 
to go through, reason 0


We have an Digium ZAP card on a E1 connection.

Paul
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[Asterisk-Users] Call file calling twice

[Remco Barende]

Hi list!

The call files are working really great I just have one problem.

I am using this as my call file:
Channel: SIP/228
Context: from-internal
Extension: 0090
Priority: 1
Callerid:  0090

so the external number is connected to my sip phone. However after 
speaking for approx 30 seconds, Asterisk does a retry and I see the 
external number in my display on the second line. It does this on 
every call. When I'm finished I also see 2 records in the log files.


This is from the event log:
Jun 21 14:08:15 asterisk[1760]: Queued call to SIP/228 expired without 
completion after 0 attempt(s)

Jun 21 14:08:16 asterisk[1760]: Queued call to SIP/228 completed


Any idea why Asterisk is trying to place the call again even though the 
first attempt was succesfull and the call is still in progress?


I didn't specify a redial anywhere.

Thanks!


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[Asterisk-Users] call file ignored?

[Remco Barende]

Hi list!

I'm trying to use call files to place outgoing calls.

I want to schedule an outbound call and want it to ring on my sip phone. 
My sip phone is SIP/228 and the call should go out according to the LCR 
rules as defined in the dialplan. I don't mind waiting for the call to be 
answered on my phone so I don't need the functionality that my phone will 
ring only when the call is answered.


This is what I have in the call file:

Channel: SIP/228(my sip phone)
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: from-internal
Extension: 003120123456
Priority: 1


When I set the permissions and move the file to 
/var/spool/asterisk/outgoing nothing happens. I guess * does find the file 
because it is gone immediately but I don't even get an error on the 
console.


What am I doing wrong?  Can I snatch a working call file from the 
outgoing directory?


Thanks!
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RE: [Asterisk-Users] call file ignored?

[jurczak]

Hello,

I just tried it, and it worked fine for me. Of course the context and the
Extension where different. Is the Channel correct?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Monday, June 20, 2005 2:12 PM
To: Asterisk Users List
Subject: [Asterisk-Users] call file ignored?

Hi list!

I'm trying to use call files to place outgoing calls.

I want to schedule an outbound call and want it to ring on my sip phone. 
My sip phone is SIP/228 and the call should go out according to the LCR 
rules as defined in the dialplan. I don't mind waiting for the call to be 
answered on my phone so I don't need the functionality that my phone will 
ring only when the call is answered.

This is what I have in the call file:

Channel: SIP/228(my sip phone)
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: from-internal
Extension: 003120123456
Priority: 1


When I set the permissions and move the file to 
/var/spool/asterisk/outgoing nothing happens. I guess * does find the file 
because it is gone immediately but I don't even get an error on the 
console.

What am I doing wrong?  Can I snatch a working call file from the 
outgoing directory?

Thanks!
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RE: [Asterisk-Users] call file ignored?

[Remco Barende]

Yes, the channel should be correct.

I'm using AMP and from-internal is the context the sip phones are normally 
in.


Do you see anything on the console even if you dial a number that isn't 
answered?


Thanks!

On Mon, 20 Jun 2005, jurczak wrote:


Hello,

I just tried it, and it worked fine for me. Of course the context and the
Extension where different. Is the Channel correct?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Monday, June 20, 2005 2:12 PM
To: Asterisk Users List
Subject: [Asterisk-Users] call file ignored?

Hi list!

I'm trying to use call files to place outgoing calls.

I want to schedule an outbound call and want it to ring on my sip phone.
My sip phone is SIP/228 and the call should go out according to the LCR
rules as defined in the dialplan. I don't mind waiting for the call to be
answered on my phone so I don't need the functionality that my phone will
ring only when the call is answered.

This is what I have in the call file:

Channel: SIP/228(my sip phone)
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: from-internal
Extension: 003120123456
Priority: 1


When I set the permissions and move the file to
/var/spool/asterisk/outgoing nothing happens. I guess * does find the file
because it is gone immediately but I don't even get an error on the
console.

What am I doing wrong?  Can I snatch a working call file from the
outgoing directory?

Thanks!
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RE: [Asterisk-Users] call file ignored?

[jurczak]

I tried also with wrong channel, and after a while the file was disappeared
and asterisk said that he was unable to call that channel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Monday, June 20, 2005 4:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] call file ignored?

Yes, the channel should be correct.

I'm using AMP and from-internal is the context the sip phones are normally 
in.

Do you see anything on the console even if you dial a number that isn't 
answered?

Thanks!

On Mon, 20 Jun 2005, jurczak wrote:

 Hello,

 I just tried it, and it worked fine for me. Of course the context and the
 Extension where different. Is the Channel correct?


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
 Sent: Monday, June 20, 2005 2:12 PM
 To: Asterisk Users List
 Subject: [Asterisk-Users] call file ignored?

 Hi list!

 I'm trying to use call files to place outgoing calls.

 I want to schedule an outbound call and want it to ring on my sip phone.
 My sip phone is SIP/228 and the call should go out according to the LCR
 rules as defined in the dialplan. I don't mind waiting for the call to be
 answered on my phone so I don't need the functionality that my phone will
 ring only when the call is answered.

 This is what I have in the call file:

 Channel: SIP/228  (my sip phone)
 MaxRetries: 1
 RetryTime: 60
 WaitTime: 30
 Context: from-internal
 Extension: 003120123456
 Priority: 1


 When I set the permissions and move the file to
 /var/spool/asterisk/outgoing nothing happens. I guess * does find the file
 because it is gone immediately but I don't even get an error on the
 console.

 What am I doing wrong?  Can I snatch a working call file from the
 outgoing directory?

 Thanks!
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