[asterisk-users] Asterisk Crashes - Segmentation Fault

Manmohan Singh Jandu manmohansj at gmail.com
Sun Aug 8 07:56:41 CDT 2010


Hi Dan,

I was trying to make the Invite working. I am getting following error
when i try to make a call.

[Aug  8 16:55:22] NOTICE[15082]: chan_local.c:534 local_call: No such
extension/context 73281 at default while calling Local channel
[Aug  8 16:55:22] NOTICE[15082]: channel.c:4042
__ast_request_and_dial: Unable to call channel Local/73281
[Aug  8 16:55:22] ERROR[12166]: pbx.c:9301 device_state_cb: Received
invalid event that had no device IE
[Aug  8 16:55:22] ERROR[12166]: app_queue.c:1099 device_state_cb:
Received invalid event that had no device IE

Following is my dialplan in /etc/asterisk/extensions.conf

[outgoing]
exten => _73...,1,Dial(SIP/callman02&SIP/callman01/${EXTEN:2})
exten => _73...,n,Congestion

following is in lib/defines.php

//Outcall defaults
define ("CHAN_TYPE", "Local"); //Use Local to let dialplan decide which chan
define ("OUT_CONTEXT", "outgoing"); //Select a context to place the call from
define ("OUT_PEER", ""); // Use this if not using CHAN_TYPE Local
define ("OUT_CALL_CID", "Parlez <1996>"); // Caller ID for Invites

--Manmohan Singh
On Fri, Aug 6, 2010 at 12:46 AM, Dan Austin <Dan_Austin at phoenix.com> wrote:
> Manmohan wrote:
>> I commented locale.php in defines.php and it perfectly worked well.
>
>> Now i am wondering what is this invite participants for, while adding
>> conference. wherein it asks for first name, lastname, emailaddress &
>> telephone number..
> The 'Invite Others' option is mostly for installs that do not have
> a consistent e-mail environment, and are using the SERVER mailer.
> This feature lets the server send invite emails to multiple parties.
> In my environments we have Exchange and Outlook, so I prefer the CLIENT
> mailer, and I can manage the invitations in my mail client
>
>> Let me brief you how i had done this setup. I had created a SIP trunk
>> between Cisco Call manager and Asterisk server. And i am using webmeetme
>> for Audio conferencing.
> Sounds familiar.  I put this package together after wasting too much
> money and time trying to make an expensive Cisco conferencing solution
> work.
>
>> Other than the invite participants, while the conf call is going on we
>> get couple of more options, when we click to the current ongoing conference
>> number.
>
>> End call -- To end the conference call
> Yes
>
>> Extend -- I am sure this is to extend the time of the call for which its
>> scheduled for, but not sure on how much time does it extends by default
>> OR is there any way to define the customized time on whatever required.
> 10 minutes is the default.  I thought I had made it configurable in lib/defines.php,
> but no I have it hard coded in conf_add (to be fixed in the next release now).
> You can search for +600 and change it to any value you like.
>
>> Invite-- When i click this button it asks me telephone number. I assume this
>> is any number which asterisk server can reach as per the dialplan configured
>> in extension.conf in /etc/asterisk.. Though this invite button looks pretty
>> much interesting to use but whenever i enter any phone number it says
>> "System error" not sure if am understand this wrongly.
> You understand it correctly, but the default settings are likely not working.
> Check out the section 'Outcall defaults' in lib/defines.php.  It is likely you
> need to change the OUT_CONTEXT at a minimum.
>
> Dan
>
> --
> _____________________________________________________________________
> -- 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
>



-- 
Thanks & Regards
Manmohan Singh Jandu



More information about the asterisk-users mailing list