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geisj at pagestation.com
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 Posted: Thu Jan 21, 2016 8:43 am Post subject: [asterisk-users] is there some blocking in 11.21.0 |
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I am using the AMI interface to start calls.
At one point I have a 10 second delay "Wait(10)" in the dialplan...
During this time it "seems" that if I then connect with the manager interface
and place a call that nothing happens till the 10 seconds is done...
I am requesting Async yes...
manager_str Action: Originate[CR ][LF ]Async: Yes[CR ][LF ]Channel: SIP/430[CR ][LF] (this is the first part of the AMI string)
How can I get past that... I want the other call to happen right away.
Thanks,
Jerry |
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asterisk_list at earth...
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| Posted: Thu Jan 21, 2016 8:46 am Post subject: [asterisk-users] is there some blocking in 11.21.0 |
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On Thursday 21 Jan 2016, Jerry Geis wrote:
| Quote: | I am using the AMI interface to start calls.
At one point I have a 10 second delay "Wait(10)" in the dialplan...
During this time it "seems" that if I then connect with the manager
interface
and place a call that nothing happens till the 10 seconds is done...
I am requesting Async yes...
manager_str Action: Originate[CR ][LF ]Async: Yes[CR ][LF ]Channel:
SIP/430[CR ][LF] (this is the first part of the AMI string)
How can I get past that... I want the other call to happen right away.
|
Use Priority: to begin at the dialplan step after the Wait() command.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
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geisj at pagestation.com
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| Posted: Thu Jan 21, 2016 9:03 am Post subject: [asterisk-users] is there some blocking in 11.21.0 |
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| Quote: | >>On Thursday 21 Jan 2016, Jerry Geis wrote:
| Quote: | | Quote: | I am using the AMI interface to start calls.
|
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>>
>> At one point I have a 10 second delay "Wait(10)" in the dialplan...
>> During this time it "seems" that if I then connect with the manager
>> interface
>> and place a call that nothing happens till the 10 seconds is done...
>>
>> I am requesting Async yes...
>> manager_str Action: Originate[CR ][LF ]Async: Yes[CR ][LF ]Channel:
>> SIP/430[CR ][LF] (this is the first part of the AMI string)
>>
>> How can I get past that... I want the other call to happen right away.
| Quote: | | Use Priority: to begin at the dialplan step after the Wait() command. |
Thanks AJS - Sorry, I was not clear enough.
|
Its not the command right after the Wait(10) that is being delayed....
I have another process that issuing the commands to the AMI.
Its basically like this:
AGI( run to bring items into conference SIP/100 & SIP/101 lets say)
Wait(10)
ConfBridge(to take this user into above conference)
But the SIP/100 and SIP/101 calls do not take place until a second delay.
Why are the SIP/100&SIP/101 calls delayed during the Wait(10) ?
Thanks
Jerry |
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tony at softins.co.uk
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| Posted: Thu Jan 21, 2016 10:08 am Post subject: [asterisk-users] is there some blocking in 11.21.0 |
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In article <CABr8-B7F1MVwPiL5Da8=qZGukU8EhvZX6ZyW5bZS55_JnB93_Q@mail.gmail.com>,
Jerry Geis <geisj@pagestation.com> wrote:
| Quote: |
| Quote: | | Quote: | On Thursday 21 Jan 2016, Jerry Geis wrote:
* I am using the AMI interface to start calls.
|
|
*>> >>* At one point I have a 10 second delay "Wait(10)" in the dialplan...
*>>* During this time it "seems" that if I then connect with the manager
*>>* interface
*>>* and place a call that nothing happens till the 10 seconds is done...
*>> >>* I am requesting Async yes...
*>>* manager_str Action: Originate[CR ][LF ]Async: Yes[CR ][LF ]Channel:
*>>* SIP/430[CR ][LF] (this is the first part of the AMI string)
*>> >>* How can I get past that... I want the other call to happen right away.
*
| Quote: | Use Priority: to begin at the dialplan step after the Wait() command.
|
Thanks AJS - Sorry, I was not clear enough.
Its not the command right after the Wait(10) that is being delayed....
I have another process that issuing the commands to the AMI.
Its basically like this:
AGI( run to bring items into conference SIP/100 & SIP/101 lets say)
Wait(10)
ConfBridge(to take this user into above conference)
But the SIP/100 and SIP/101 calls do not take place until a second delay.
Why are the SIP/100&SIP/101 calls delayed during the Wait(10) ?
|
Are you saying that this worked in earlier versions but you started to
get the delay when you updated to 11.21.0? Or just that you happened to
be using 11.21.0 the first time you tried this scenario?
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
--
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geisj at pagestation.com
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| Posted: Thu Jan 21, 2016 10:11 am Post subject: [asterisk-users] is there some blocking in 11.21.0 |
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| Quote: | >Are you saying that this worked in earlier versions but you started to
| Quote: | get the delay when you updated to 11.21.0? Or just that you happened to
be using 11.21.0 the first time you tried this scenario?
|
I should have said "first time" trying this.Any thoughts? Jerry |
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tony at softins.co.uk
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| Posted: Thu Jan 21, 2016 10:50 am Post subject: [asterisk-users] is there some blocking in 11.21.0 |
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In article <CABr8-B5HQBy1vU2i=6iV5TbC0-yNG0HxM8c-M2O1XC7YDMvE-A@mail.gmail.com>,
Jerry Geis <geisj@pagestation.com> wrote:
| Quote: |
| Quote: | Are you saying that this worked in earlier versions but you started to
get the delay when you updated to 11.21.0? Or just that you happened to
be using 11.21.0 the first time you tried this scenario?
|
I should have said "first time" trying this.
Any thoughts?
|
Not really. Very little info to go on so far. You need to give us
more detail of what you are doing with AGI and AMI.
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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geisj at pagestation.com
Guest
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| Posted: Thu Jan 21, 2016 1:14 pm Post subject: [asterisk-users] is there some blocking in 11.21.0 |
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| Quote: | >Not really. Very little info to go on so far. You need to give us
| Quote: | more detail of what you are doing with AGI and AMI.
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Sorry - let me try again...
I am basically doing the following:
1) calling a phone SIP/401 upon answer run an AGI for voice prompts etc... to select AUDIO groups
2) when done setup to return to the dialplan - exit AGI
3) issue AGI that calls those groups selected (SIP/430 & SIP/431) at the moment to bring into a conference.
4) Wait 10 seconds
5) jump SIP/401 into conference
6) speak live to endpoints.
However my issue is such that step 3 above "seems" to block until after step 5.
My Manager AMI connection reports success in step 3:
21-Jan-16 01:02 pm asterisk_pa_list() manager_str Action: Originate[CR ][LF ]Async: Yes[CR ][LF ]Channel: SIP/430[CR ][LF ] (truncated)
21-Jan-16 01:02 pm asterisk_pa_list() manager_str return Response: Success[CR ][LF ]Message: Originate successfully queued[CR ][LF ][CR ][LF ]
I was expecting the jump into conference from step 3 to go ahead and do the request - not wait till after step 5.
Does this help? What am I not doing right so the calls in step 3 happen WAY before the 10 seconds is complete?
Jerry |
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tony at softins.co.uk
Guest
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| Posted: Thu Jan 21, 2016 2:13 pm Post subject: [asterisk-users] is there some blocking in 11.21.0 |
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In article <CABr8-B5ejk4Q5tfU4hYGpgq=0pp=rXUKz+fCUesEeNQimmKgxw@mail.gmail.com>,
Jerry Geis <geisj@pagestation.com> wrote:
| Quote: |
| Quote: | Not really. Very little info to go on so far. You need to give us
more detail of what you are doing with AGI and AMI.
|
Sorry - let me try again...
I am basically doing the following:
1) calling a phone SIP/401 upon answer run an AGI for voice prompts etc...
to select AUDIO groups
2) when done setup to return to the dialplan - exit AGI
3) issue AGI that calls those groups selected (SIP/430 & SIP/431) at the
moment to bring into a conference.
4) Wait 10 seconds
5) jump SIP/401 into conference
6) speak live to endpoints.
However my issue is such that step 3 above "seems" to block until after
step 5.
My Manager AMI connection reports success in step 3:
21-Jan-16 01:02 pm asterisk_pa_list() manager_str Action: Originate[CR ][LF
]Async: Yes[CR ][LF ]Channel: SIP/430[CR ][LF ] (truncated)
21-Jan-16 01:02 pm asterisk_pa_list() manager_str return Response:
Success[CR ][LF ]Message: Originate successfully queued[CR ][LF ][CR ][LF ]
I was expecting the jump into conference from step 3 to go ahead and do the
request - not wait till after step 5.
Does this help? What am I not doing right so the calls in step 3 happen WAY
before the 10 seconds is complete?
|
The problem we have is you are asking us to debug a description. That's
usually almost impossible. Much easier to debug code, as it may or may not
match your description! So we can offer more insight if you include
information such as:
1. Relevant section of your dialplan.
2. The AGI code that does the origination. Presumably it is calling the AMI?
In your example above, the Originate AMI has a Channel, but doesn't show
any Context, Extension and Priority. Where is the channel supposed to go
once the call to SIP/430 is answered?
3. The Asterisk "full" log, with at least verbose level 3, encompassing
your attempt.
4. Anything else that you yourself would need to look at to debug the issue.
Cheers
Tony
--
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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asterisk_list at earth...
Guest
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| Posted: Fri Jan 22, 2016 4:15 am Post subject: [asterisk-users] is there some blocking in 11.21.0 |
|
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On Thursday 21 Jan 2016, Jerry Geis wrote:
| Quote: | | Quote: | Not really. Very little info to go on so far. You need to give us
more detail of what you are doing with AGI and AMI.
|
Sorry - let me try again...
I am basically doing the following:
1) calling a phone SIP/401 upon answer run an AGI for voice prompts etc...
to select AUDIO groups
2) when done setup to return to the dialplan - exit AGI
3) issue AGI that calls those groups selected (SIP/430 & SIP/431) at the
moment to bring into a conference.
4) Wait 10 seconds
5) jump SIP/401 into conference
6) speak live to endpoints.
However my issue is such that step 3 above "seems" to block until after
step 5.
My Manager AMI connection reports success in step 3:
21-Jan-16 01:02 pm asterisk_pa_list() manager_str Action: Originate[CR ][LF
]Async: Yes[CR ][LF ]Channel: SIP/430[CR ][LF ] (truncated)
21-Jan-16 01:02 pm asterisk_pa_list() manager_str return Response:
Success[CR ][LF ]Message: Originate successfully queued[CR ][LF ][CR ][LF ]
I was expecting the jump into conference from step 3 to go ahead and do the
request - not wait till after step 5.
Does this help? What am I not doing right so the calls in step 3 happen WAY
before the 10 seconds is complete?
|
It's not *that* helpful, because you are playing your cards way too close to
your chest; we don't even know what language you are using for your AGI. So
the following may be no help at all to you, but is included anyway for the
benefit of anyone else searching the archives a few years down the line.
AGI scripts that do not need to interact with the dialplan anymore should
fork() and then, in the parent process, exit(0). The child process must then
close STDIN, STDOUT and STDERR as soon as possible; the dialplan will carry on
executing when they are closed, and the child process is free to carry on
doing its own stuff independently.
In Perl, do *not* do it like this:
fork && exit 0;
close STDIN;
close STDOUT;
close STDERR;
in case fork fails and returns an undefined value; the "child process" will
continue under the parent's PID. Do it properly:
if ($child_pid = fork) { # parent process
exit 0;
}
elsif (defined $child_pid) { # child process
close STDIN;
close STDOUT;
close STDERR;
# carry on doing stuff
}
else { # fork failed
die "Could not fork";
# tidy up mess as best we can
};
PHP's pcntl_fork() returns -1 if it fails (process numbers are always
positive). So the equivalent code in PHP would be:
$child_pid = pcntl_fork();
if ($child_pid < 0) { // fork failed
die("Could not fork");
}
elseif ($child_pid) { // parent process
exit(0);
}
else { // child process
fclose(STDIN);
fclose(STDOUT);
fclose(STDERR);
// carry on doing stuff
};
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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