Re: [Asterisk-Users] What do you name yours
mine, on the stars of saturn options: Dione, Rhea, Titan, Mimas, Enceladus, Tethys, Hyperion, Iapetus, and Phoebe Abhishek -- Drishti-Soft Solutions Pvt Ltd http://www.drishti-soft.com On 5/12/05, Christopher Stephens [EMAIL PROTECTED] wrote: Mine is called 'blacksun', as that's where it's colo'd. (idiocy in a naming convention, I know.) On Wed, 11 May 2005 19:55:36 -0700 (PDT), Matt Klein [EMAIL PROTECTED] said: Mine is named spike... On Thu, 12 May 2005, Paul Hales wrote: We bought one of those books on the worst cars ever made...every page has great names... PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Thursday, 12 May 2005 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Andrew Latham Subject: Re: [Asterisk-Users] What do you name yours On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to disconnect a call manually
soft hangup channel name -Abhishek Drishti-Soft Solutions Pvt Ltd http://www.drishti-soft.com On 5/2/05, Asterisk guy [EMAIL PROTECTED] wrote: 1 after giving command oh323 show channels, i want to disconnect a call, is there any command to disconnect a call? 2 how asterisk kill a hung/dead call ? for most commercial softswitch, there are a setting for maximum duration for a call. they will hang up it l if its duration reachs the limit. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I eveluate trailing numbers in extensions.conf?
Hi,
you can use StripMSD if you are sure of the number of dialled digits.
or you can use {$EXTEN:-1:1}, where {$EXTEN:a:b} means first b digits
starting from a, from the front or back depending whether a 0 or a 0.
additionally, can have your own variables, just look at
pbx_retrieve_variable in pbx.c
-Abhishek
Drishti Soft
www.drishti-soft.com
On Sun, 13 Mar 2005 10:01:27 +, Umar Sear [EMAIL PROTECTED] wrote:
Checkout
http://www.voip-info.org/wiki-Asterisk+variables
I believe that should have the answer for you.
furthermore assuming that your number is always going to be 12 digits.
exten = _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number.
Hope this helps.
Umar
On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz [EMAIL PROTECTED] wrote:
Hi,
this message seems to not have made it to the list the first time - sorry
if it did.
My SIP provider includes trailing numbers to my account just fine, like
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc62e.644d2d75.0
From: Anonymous sip:[EMAIL PROTECTED];tag=as4b25d20f
Call-ID: [EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as406f4254
CSeq: 102 ACK
User-Agent: sipgate ser
Content-Length: 0
where 498645342456 is my SIP account phone number that can be reached from
the outside just fine. My question is, how can I evaluate the trailing 2
in my extensions.conf? This would be ideal for direct dialing to an attached
phone, and not be restricted to a single digit.
asterisk-dev The full number is included in the SIP message but does *
keep it
somewhere internally so that one could maybe add another externally usable
* variable? I browsed the source code but could not find anything... TIA!
Ciao,
hm
--
Today is the first day of the rest of the mess
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[Asterisk-Users] Asterisk with mediant 2000 - facing problems
Hi, I have been using/working on asterisk for some time now and presently was trying to configure asterisk to work with digium cards. It worked fine with the fxo/fxs cards, but now i'm trying to get it working by interfacing it with mediant t1 port. no avail ... anyone out there got it working, what particular configuration used on mediant (isdn signalling, framing, coding etc ??) and/or what configuration on asterisk side ?? if anyone has it working, please send me the ini file for the mediant and the zaptel.conf etc. would be extremely thankful. Regards Abhishek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
