Hi,


Where in Asterisk @Home does the configuration for Outgoing SIP calls to other services get placed?

Ie. To reach 20805 or [EMAIL PROTECTED] how do you configure Asterisk @Home to allow this SIP address to

Be called?

 

Bruce

 

-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent:
Sunday, March 05, 2006 5:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; [email protected]
Subject: [asterisk-biz] Signate Intro to * -
London Training March 21-23

 

We still have a seat open in the London Introduction to Asterisk class.

 

TKS

 

Paul

 

 

Paul Mahler

[EMAIL PROTECTED]

 

 


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of McQuiggan, Mark xt46480
Sent:
Sunday, March 05, 2006 12:20 PM
To: [email protected]
Subject: [Asterisk-Users] Problem with libpri?

 

While testing a problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon this problem:

 

Program received signal SIGSEGV, Segmentation fault.

[Switching to Thread -1210770512 (LWP 11346)]

0x002e3fe1 in pri_release_timeout (data="" at q931.c:2589

2589 q931.c: No such file or directory.

in q931.c

         

q931.c is in libpri, function pri_release_timeout, and line 2589 reads: 

        if (pri->debug & PRI_DEBUG_Q931_STATE)
                pri_message(pri, "Timed out looking for release complete\n");

 

PRI Debug was not on in the asterisk console. 

Any ideas?  My asterisk restarts about twice a day, and drops any current calls in the process.

Regards,

Mark McQuiggan

 

 



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