Displaying 5 results from an estimated 5 matches for "ext200".
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ext203
2010 Aug 19
3
Calling Line Identity - any ideas
Hi list
I have a requirement that I just don't know how to address - I don't think
its strange but can't find any pointers anywhere.
I have a user that wishes to have a "multi phone" divert. By that I mean
"calls made to his extension say Ext200 can be redirected to a different
extension say Ext400 and also to his home landline.
Doing the dial is fine using Dial(SIP/Ext400&SIP/TheWorld/441234567890)
The problem is CLID -
At the moment internal calls (Ext to Ext) show a CLID "EXTxxx" and External
Calls show the received CL...
2007 Jun 28
2
CDR and call transfer
Hello,
I'm using digium E1 cards and serving SIP users at Asterisk. After the
following call (see below) CDR shows two records. First looks as
outbound call, but the second - as inbound call. Is it a bug or intended
behavior?
Call flow:
SIP (ext: 100) -> ZAP (national number)
SIP (ext: 100) transfers to SIP (ext: 200)
SIP (ext: 200) -> ZAP (national number).
In CDR it looks like
2005 Feb 22
2
Custom Menu Not Working
Greetings *`s,
I am having what appears to be a small problem, but the frustration is
erally getting to me, what am I doing wrong here ?
I used AMP to set up a custom menu, so if caller presses 1 it goes to
ext200, if caller presses 2 it goes to ext201 etc etc...
Now I have created a third option that when the caller presses 3 it must
play a sound and hang up.
No rocket science yet.
When adding the details in AMP for when caller dials 3, I have
referenced it using 'custom-myapp,s,1', and if I go to...
2005 Jun 24
0
How to setup two Asterisk boxes - keeping theregistration
You'll need to create a trunk between the two systems
Then configure your out bound routing to use that trunk
Eg if you use 1 as the prefix for the trunk
BOX A Ext200 will call box B EXT201 by dialing 1201
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ronald
Wiplinger
Sent: Sunday, 19 June 2005 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:...
2013 Apr 11
1
"Dropping call because extensions '200', 's' and 'i' doesn't exists"
hello all
i,m newbie in asterisk and now want to sip and h323 connection.
this is my scenario:
phone(ext100)--->freepbx---sip--->system1---H323--->system2--->freepbx--->phone(ext200)
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says "service unavailable".
i debug asterisk in my system 2 and see below message:
"Dropping call because extensions '200', 's' and 'i' doesn't exists
in...