Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] what is the possible cause of maximum pbx stack exceeded


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
covici at ccs.covici.com
Guest





PostPosted: Wed Dec 04, 2013 4:27 am    Post subject: [asterisk-users] what is the possible cause of maximum pbx s Reply with quote

Hi. I am using asterisk 11 svn r401076M and I am getting this warning
at times. I can't find much doing a google search, so anyone with any
ideas?

I have looked at the logs, but can find no particular pattern to
indicate where this is happening and the system appears to be otherwise
working, but I am still wondering if something is wrong. I am also
using freepbx in case there are known issues there -- because some of
these occur during their dialout trunk code.

Any suggestions would be appreciated.

--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?

John Covici
covici@ccs.covici.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
Back to top
rnewton at digium.com
Guest





PostPosted: Tue Dec 10, 2013 11:03 am    Post subject: [asterisk-users] what is the possible cause of maximum pbx s Reply with quote

I'm not a developer, but from comments in the code, it looks like that
warning is generated when Asterisk dialplan processing exceeds a
certain depth of includes.

Seeing as it is possibly a dialplan related issue, and FreePBX is
writing your dialplan, you may have the best odds of getting a
relevant answer by asking on the FreePBX forums (and giving them
access to a copy of your logs to examine)

That's all I got! Smile

On Wed, Dec 4, 2013 at 3:27 AM, <covici@ccs.covici.com> wrote:
Quote:
Hi. I am using asterisk 11 svn r401076M and I am getting this warning
at times. I can't find much doing a google search, so anyone with any
ideas?

I have looked at the logs, but can find no particular pattern to
indicate where this is happening and the system appears to be otherwise
working, but I am still wondering if something is wrong. I am also
using freepbx in case there are known issues there -- because some of
these occur during their dialout trunk code.

Any suggestions would be appreciated.

--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?

John Covici
covici@ccs.covici.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



--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
covici at ccs.covici.com
Guest





PostPosted: Tue Dec 10, 2013 1:51 pm    Post subject: [asterisk-users] what is the possible cause of maximum pbx s Reply with quote

OK, thanks.

Rusty Newton <rnewton@digium.com> wrote:

Quote:
I'm not a developer, but from comments in the code, it looks like that
warning is generated when Asterisk dialplan processing exceeds a
certain depth of includes.

Seeing as it is possibly a dialplan related issue, and FreePBX is
writing your dialplan, you may have the best odds of getting a
relevant answer by asking on the FreePBX forums (and giving them
access to a copy of your logs to examine)

That's all I got! Smile

On Wed, Dec 4, 2013 at 3:27 AM, <covici@ccs.covici.com> wrote:
Quote:
Hi. I am using asterisk 11 svn r401076M and I am getting this warning
at times. I can't find much doing a google search, so anyone with any
ideas?

I have looked at the logs, but can find no particular pattern to
indicate where this is happening and the system appears to be otherwise
working, but I am still wondering if something is wrong. I am also
using freepbx in case there are known issues there -- because some of
these occur during their dialout trunk code.

Any suggestions would be appreciated.

--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?

John Covici
covici@ccs.covici.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



--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?

John Covici
covici@ccs.covici.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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services