Re: [asterisk-users] Missed call
There is no signal that is sent to display a missed call. Your cell phone does that. If it rings and is not answered it counts that as a miss. The only way to avoid it is to not ring it. So instead of simultaneous ringing you can do sequential. -- Mehmet Avcioglu meh...@activecom.net > On Jun 4, 2015, at 11:21 PM, Luca Bertoncello wrote: > > Hi list! > > I configured Asterisk to forward the incoming call for a number to both > phones. > I wrote that: > > exten => _0049351222,n,Dial(SIP/0049351222&SIP/004935,,R) > > of course it works... > Now the problem is, that when a phone get the call, on the other phone I get > "1 missed call"... > Is it possible to avoid that and signaling the other phone, that the call was > not "missed"? > > Thanks a lot > Luca Bertoncello > (lucab...@lucabert.de) > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ControlPlayback unable to play MixMonitor files, Premature EOF
I have a strange problem with MixMonitor and ControlPlayback, I am recording wav files MixMonitor(/path/filename.wav,b) When I try to play them back, I get no audio, dialplan continues to the next line as if the sound file is 0 seconds long ControlPlayback(/path/filename) I can play the file in other programs and verify there is sound there If I touch the file with sox, it will start working sox /path/filename.wav /path/filename-trim.wav trim 1 ControlPlayback(/path/filename-trim) ; Works Both files look the same /path/filename.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz /path/filename-trim.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz However the sox command above gives a "Premature EOF" warning message for the input wav file. I suspect this is the problem. Why would I be creating files using MixMonitor that have a premature end of file and what can I do to avoid this? Using Asterisk 1.8.19. Thanks -- Mehmet Avcioglu meh...@activecom.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the maximum number of meetme's allowed?
What architecture are you using? You would need to be running a 64 bit OS to take advantage of the higher memory. -- Mehmet Avcioglu meh...@activecom.net On Dec 24, 2012, at 5:13 PM, Deepesh D wrote: > Hello, > > What is the maximum number of meetme's allowed by asterisk. > > On my server with an 8 GB memory, I start getting the following error > after 150-160 meetme's are created > > WARNING[3485]: app_meetme.c:1820 conf_run: Unable to open DAHDI pseudo > channel: Cannot allocate memory > > At this time the server still has about 6 GB of free memory. I even > tried this on a server with higher memory, it gives the same result. > > I am using asterisk 1.4.44. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting OLD Voicemails
You can delete old files, it won't break anything. Also to prevent saving files in multiple formats, edit voicemail.conf and change format parameter under general. -- Mehmet Avcioglu meh...@activecom.net On May 23, 2012, at 1:03 AM, Danny Dias wrote: > Thanks Jason, > > But how to delete them? there are a lot of old voicemails, but i don't want > to break the app_voicemail. > > > > 2012/5/22 Jason Parker > On 05/22/2012 04:54 PM, Danny Dias wrote: > > There are 4 files for each voicemail: > > > > msg.gsm > > msg.txt > > msg.wav > > msg.WAV > > > > That is perfectly normal. The .txt file is metadata that contains things like > caller ID and duration. Asterisk will also save voicemails into every format > you have specified in voicemail.conf. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > www.danntel.net > sip:danny4...@thesipschool.com > sip:dann...@opensips.org > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk voicemail
To turn of the after recording messages, in voicemail.conf change review=yes to
review=no
--
Mehmet Avcioglu
meh...@activecom.net
On May 21, 2012, at 6:11 PM, Danny Nicholas wrote:
> Your "other option" would be to set CHANNEL(language) to your native
> language. The ,s option only suppresses the initial greeting; you are
> wanting to suppress the "after recording" instructions.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bogdan
> Sent: Monday, May 21, 2012 10:07 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] asterisk voicemail
>
> I am using that but does not seem to work.
>
> On 05/21/2012 05:53 PM, Danny Nicholas wrote:
>> That's what the ,s switch on voicemail() is for.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bogdan
>> Sent: Monday, May 21, 2012 9:52 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] asterisk voicemail
>>
>> Hello,
>>
>> I am using asterisk 1.4 and I have the following issue: I would like
>> to turn off the voicemail option that asks you to press 1 ("to accept
>> this message"), 2 (" to listen to it") and 3 ("to rerecord your
>> message")when you make the record since they are not in my native
>> language. Does anyone know how to do this?Thank you!
>>
>> Bogdan
>>
>> --
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>>
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>
>
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Re: [asterisk-users] Slow AMI Originate
Since this didn't ring a bell with anybody, it must not be related to asterisk. Digging further into my code, I may have found the reason for this. Note to others who may find this post later, looks like asterisk can take Originate actions as fast as you can give it, look other places..:) -- Mehmet Avcioglu meh...@activecom.net On May 19, 2012, at 12:26 PM, Mehmet Avcioglu wrote: > > Hello, > > We use AMI to originate calls. Sometimes, lately every morning, the AMI > Originate process operates extremely slowly. I cannot see the calls in "core > show channels verbose", I don't know where they are, what state they are in, > after 2-3 minutes the calls go through one after the other. As mentioned, it > usually happens in the morning as soon as people start their workday, where > there are a lot of logins and calls being made, but no where close to a peak > in terms of simultaneous channels, etc. In some cases restarting asterisk, in > others just taking the storm and waiting it out solves the problem. Having a > hard time coming up with something to troubleshoot this. Any ideas would be > appreciated. > > -- > Mehmet Avcioglu > meh...@activecom.net > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slow AMI Originate
Hello, We use AMI to originate calls. Sometimes, lately every morning, the AMI Originate process operates extremely slowly. I cannot see the calls in "core show channels verbose", I don't know where they are, what state they are in, after 2-3 minutes the calls go through one after the other. As mentioned, it usually happens in the morning as soon as people start their workday, where there are a lot of logins and calls being made, but no where close to a peak in terms of simultaneous channels, etc. In some cases restarting asterisk, in others just taking the storm and waiting it out solves the problem. Having a hard time coming up with something to troubleshoot this. Any ideas would be appreciated. -- Mehmet Avcioglu meh...@activecom.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Problem
Danny's suggestion of using System instead of AGI was correct, it was wrong of me to use AGI here as there is no communication. The same script does communicate with asterisk when executed with different options, so I had just kept the same line using different options at different places. Today I changed the AGI calls to System at places where there is no response back to asterisk. Kept everything the same, instead of AGI(script.php,var1,var2) made it System(/path/script.php "var1" "var2") and it worked without a problem. This way I have been able to fix the problem where this caused a deadlock requiring human intervention. The lines in the dialplan that continue to use the AGI call still experience the same problem as before. Today it happened 9 times out of 10486 executions, but these are not as fatal as before, so they can almost be ignored. > Are you reading the AGI environment in the script? Yes > How are you passing variables from Asterisk? As channel variables or as > command line parameters? > What is the dialplan statement that executes your script? Command line parameters, explained above. > Can you post the script source? Yes but I really cannot see how the script could have a problem, it is around 2000 lines and since my problem has come to a more manageable level, I would rather not bug you with debugging the whole thing:) > If you captured the AGI dialog from a call when your script failed and you > feed the same values (including the AGI environment) into your script from a > shell does it fail? Yes it runs just fine. It even runs fine at different times from within the asterisk using AGI when the same number calls or is called. So the only variable that is different between executions is time. > On a Unix system level, a 4 means 'Interrupted system call' -- assuming this > is what your '4' signifies. BTW, I'm a 1.2 Luddite and the AGI return value > was meaningless back then. Running these on 1.8.5 I'll migrate this setup to a different environment, perhaps try at different asterisk and try the signal catching mentioned before and if I get any more data will update the list. Thanks a lot. -- Mehmet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Problem
Thanks for the response. Although I had already gone thorough a lot of these types of QAs during my own problem solving, your suggestions did bring a couple that I had not. > In no particular order... > > Earlier, you suspected receiving a signal was causing your problem. > Instead of ignoring the signal, how setting up a handler and logging the > reception? The program is written in a very top to bottom way (himm stateless? non-oo? not sure what to call it) and would be really hard for me to catch those and figure out where it happened and clean things up. So I have been reluctant to do this. But it is one of those tests that I'll eventually hit if I cannot solve the problem. > How about 'agi set debug on'? I was doing extensive logging in my scripts and had turned on full logging for asterisk but I had not turned agi debuging on. I did and carefully inspected the output. And there is nothing out of ordinary between successful executions and unsuccessful ones. > Whose AGI library did you use? If you 'rolled your own' maybe you are > violating the protocol. Nobody gets it right the first time :) > What does the AGI do? > Can you execute it outside of Asterisk by feeding it the appropriate cruft > via STDIN? I am using my own. Yes agreed, but this doesn't even interact with asterisk at all. Just receives a couple values and updates jobs, database accordingly. Does not return anything back to asterisk. Yes executes fine outside of asterisk and even executes fine within asterisk 99.9% of the time. > Are there any characteristics common to the 48? Same ANI? Same DNIS?, Same > path through your dialplan? I have not been able to find any difference. So "returning 4" doesn't mean anything special. It is just the return code received from the AGI application that was forked. What about the fact that asterisk does not continue with the dialplan after receiving 4, is that expected behavior? I have not been able to find what would make a php cli application exit with 4 other than exit(4). php does not log anything when this happens, doesn't write anything to stderr (I specifically remained on the console I started * to see stderr). So this can only be a crash of some sort of php cli binary. What type of a crash would cause it to return 4? I will keep digging. Thanks -- Mehmet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server
Actually it doesn't say "AGI(async:script)" it says "AGI(async:agi)" and than continues further to setting up an AMI user so the script is executed through the manager interface?? Than it says "AGI(agi:async)".?? Well most importantly it says "Cons of async AGI: It is the most complex method of using AGI to implement." ..:) I have been interested in Async AGI as well and after reading your post looked into the link you provided, seems different than what we immediately think, a background process. Perhaps just start the script normally "AGI(script.sh)" and than inside it run your background process "background-script.sh > /dev/null 2>&1 < /dev/null &" or fork a new process, detach, run in background, etc... Hopefully somebody else can point us towards the right direction in setting up a real asterisk asynchronous AGI application. -- Mehmet On Sep 25, 2011, at 2:00 AM, Randall Degges wrote: > Hi Everyone, > > I've been trying to get asynchronous AGIs working in some Asterisk code I > have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and AGI > scripting overall. Here's my problem: I can't get Asterisk to execute *any* > AGIs asynchronously. > > Firstly, I discovered asynchronous AGIs via "Asterisk: The Definitive Guide". > The asynchronous AGI information I read can be found online, here: > http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the > section titled "Async AGI--AMI Controlled AGI"). > > According to the book, since Asterisk 1.6.0 the AGI dialplan application has > been able to execute AGI scripts asynchronously, via the syntax: > > exten => s,1,AGI(async:script) > > According to the book, using the "async:" prefix should have Asterisk run the > AGI script in the background and instantly continue executing dialplan code. > > So here's my Asterisk dialplan code that's being run: > > [hangup] > exten => s,1,AGI(async:/etc/asterisk/scripts/hangup.py) > exten => s,n,Return() > > Pretty simple context--essentially my AGI script just does some call clean up > logic before a caller hangs up, talking to a few web servers and generating > statistics for later usage. What happens when Asterisk executes this context, > is: > > WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute > '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does > not exist. > > As you can see, Asterisk is ignoring the async: directive, and treating it as > part of the AGI script path. > > Is there anyway for me to make asynchronous AGIs work? I've tried searching > online to no avail. > > I'd greatly appreciate any responses, thanks for your time. > > -Randall > > -- > Randall Degges > http://rdegges.com/ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Problem
Thanks for the help. 48 times out of 18000 should be ignorable in most cases
but it required human intervention 48 times.:) Obviously if there is no other
way I'll find a way to automate that human intervention somehow.. Thanks again.
--
Mehmet
On Sep 24, 2011, at 4:34 PM, Sam Govind wrote:
> Should we consider it ignorable or you are still interested in resolving !
> Best of Luck in any case !
>
> On Sat, Sep 24, 2011 at 6:16 PM, Mehmet Avcioglu wrote:
>
> Out of about 18000 instances this problem occurred 48 times yesterday. So it
> has been pretty much impossible for me to replicate it under test conditions.
>
> Thanks
>
> --
> Mehmet
>
> On Sep 24, 2011, at 4:00 PM, Sam Govind wrote:
>
>> Thats wicked !! hmmm stop your asterisk (if u can afford) and run it like
>> "asterisk -cvg" and then make a call.. see whats your AGI doing in there
>> !!
>>
>> On Sat, Sep 24, 2011 at 5:43 PM, Mehmet Avcioglu
>> wrote:
>>
>> Thanks for the response.
>>
>> Asterisk logs the execution of the AGI and script completed messages within
>> the same second, so less than a second.
>>
>> --
>> Mehmet
>>
>> On Sep 24, 2011, at 3:34 PM, Sam Govind wrote:
>>
>>> How much time your AGI is taking? Check if it is completing its task and
>>> not killed by asterisk. I guess we've 6~7 seconds before asterisk kills all
>>> call channel and related tasks.
>>>
>>> On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu
>>> wrote:
>>>
>>> On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
>>> > I have an AGI script that occasionally disappears without completing its
>>> > action and asterisk logs the following.
>>> >
>>> > AGI Script script.php completed, returning
>>> > 4
>>> > Spawn extension (context, 0123456, 2) exited non-zero on
>>> > 'Local/0123456@context-f46e;1'
>>>
>>> I also changed the dialplan and added a line to print AGISTATUS, but when
>>> this "returning 4" happens, asterisk stops there and doesn't execute any
>>> further dialplan actions, so I don't even see AGISTATUS value.
>>>
>>> exten => h,1,AGI(script.php,${ANSWEREDTIME},,,)
>>> exten => h,n,NoOp(${AGISTATUS})
>>>
>>> Executing [h@context:1] AGI("Local/0123456@context-4b79;1",
>>> "script.php,13,,,") in new stack
>>> AGI Script script.php completed, returning 4
>>> Spawn extension (context, h, 1) exited non-zero on
>>> 'Local/0123456@context-4b79;1'
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
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Re: [asterisk-users] AGI Problem
Out of about 18000 instances this problem occurred 48 times yesterday. So it
has been pretty much impossible for me to replicate it under test conditions.
Thanks
--
Mehmet
On Sep 24, 2011, at 4:00 PM, Sam Govind wrote:
> Thats wicked !! hmmm stop your asterisk (if u can afford) and run it like
> "asterisk -cvg" and then make a call.. see whats your AGI doing in there
> !!
>
> On Sat, Sep 24, 2011 at 5:43 PM, Mehmet Avcioglu wrote:
>
> Thanks for the response.
>
> Asterisk logs the execution of the AGI and script completed messages within
> the same second, so less than a second.
>
> --
> Mehmet
>
> On Sep 24, 2011, at 3:34 PM, Sam Govind wrote:
>
>> How much time your AGI is taking? Check if it is completing its task and not
>> killed by asterisk. I guess we've 6~7 seconds before asterisk kills all call
>> channel and related tasks.
>>
>> On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu
>> wrote:
>>
>> On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
>> > I have an AGI script that occasionally disappears without completing its
>> > action and asterisk logs the following.
>> >
>> > AGI Script script.php completed, returning 4
>> > Spawn extension (context, 0123456, 2) exited non-zero on
>> > 'Local/0123456@context-f46e;1'
>>
>> I also changed the dialplan and added a line to print AGISTATUS, but when
>> this "returning 4" happens, asterisk stops there and doesn't execute any
>> further dialplan actions, so I don't even see AGISTATUS value.
>>
>> exten => h,1,AGI(script.php,${ANSWEREDTIME},,,)
>> exten => h,n,NoOp(${AGISTATUS})
>>
>> Executing [h@context:1] AGI("Local/0123456@context-4b79;1",
>> "script.php,13,,,") in new stack
>> AGI Script script.php completed, returning 4
>> Spawn extension (context, h, 1) exited non-zero on
>> 'Local/0123456@context-4b79;1'
>
> --
> _
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Re: [asterisk-users] AGI Problem
Thanks for the response.
Asterisk logs the execution of the AGI and script completed messages within the
same second, so less than a second.
--
Mehmet
On Sep 24, 2011, at 3:34 PM, Sam Govind wrote:
> How much time your AGI is taking? Check if it is completing its task and not
> killed by asterisk. I guess we've 6~7 seconds before asterisk kills all call
> channel and related tasks.
>
> On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu wrote:
>
> On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
> > I have an AGI script that occasionally disappears without completing its
> > action and asterisk logs the following.
> >
> > AGI Script script.php completed, returning 4
> > Spawn extension (context, 0123456, 2) exited non-zero on
> > 'Local/0123456@context-f46e;1'
>
> I also changed the dialplan and added a line to print AGISTATUS, but when
> this "returning 4" happens, asterisk stops there and doesn't execute any
> further dialplan actions, so I don't even see AGISTATUS value.
>
> exten => h,1,AGI(script.php,${ANSWEREDTIME},,,)
> exten => h,n,NoOp(${AGISTATUS})
>
> Executing [h@context:1] AGI("Local/0123456@context-4b79;1",
> "script.php,13,,,") in new stack
> AGI Script script.php completed, returning 4
> Spawn extension (context, h, 1) exited non-zero on
> 'Local/0123456@context-4b79;1'
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Re: [asterisk-users] AGI Problem
On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
> I have an AGI script that occasionally disappears without completing its
> action and asterisk logs the following.
>
> AGI Script script.php completed, returning 4
> Spawn extension (context, 0123456, 2) exited non-zero on
> 'Local/0123456@context-f46e;1'
I also changed the dialplan and added a line to print AGISTATUS, but when this
"returning 4" happens, asterisk stops there and doesn't execute any further
dialplan actions, so I don't even see AGISTATUS value.
exten => h,1,AGI(script.php,${ANSWEREDTIME},,,)
exten => h,n,NoOp(${AGISTATUS})
Executing [h@context:1] AGI("Local/0123456@context-4b79;1", "script.php,13,,,")
in new stack
AGI Script script.php completed, returning 4
Spawn extension (context, h, 1) exited non-zero on
'Local/0123456@context-4b79;1'
--
Mehmet
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Re: [asterisk-users] AGI Problem
On Sep 23, 2011, at 8:07 PM, Danny Nicholas wrote: > Just a WAG - 4 is the error level returned by your php script, where it > normally returns 0. Yes would thing so. But at no place in my script I intentionally exit with 4. I believe 4 is SIGILL (Illegal Instruction) so my script might be seg faulting somewhere? Should I be going after this? It is a php script and php doesn't log anything for these instances. Thanks -- Mehmet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Problem
Hello, I have an AGI script that occasionally disappears without completing its action and asterisk logs the following. AGI Script script.php completed, returning 4 Spawn extension (context, 0123456, 2) exited non-zero on 'Local/0123456@context-f46e;1' I figured this was due to channel hanging up and * sending a SIGHUP to the script and added a catch and ignore for SIGHUP and SIGPIPE. But I still have instances where AGI script gets lost. I am running 1.8. Any ideas what "returning 4" really means, where should I concentrate? Thanks -- Mehmet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
