Hi,
I think you want this to be 102 since a busy returns n+101 n being the
priority your Dial function was called.
exten = _X.,101,Busy
should be
exten = _X.,102,Busy
HTH
-b
Quoting Eric Wieling [EMAIL PROTECTED]:
Nicklas Bondesson wrote:
Just like this? It doesn't seem to work though.
Hi,
The first thing is I dont see a username=username in your config.I
recommend
keeping the username the same as what you have in the brackets, in this case
[201]
If you are behind a NAT device make sure to add nat=yes
And if you dont have it configured in the [general] context be sure to
Yes you can, It's called DISA. Realize that using DISA has it's potential
security concerns.
From the asterisk console type show application DISA for more information.
DISA = Direct Inward System Access
Ciao,
-bh
Quoting Angus Berry [EMAIL PROTECTED]:
I haven't found this in any docs or faqs
Curious if anyone else has run into this.
I am testing this with a Sipura-2000 with firmware rev 1.0.33. On FC1 with
Asterisk 1.0 Stable branch 4/1/2004 CVS (no April fools jokes please :) )
The Sipura has the ability so when you dial *67 it turns ON CID block and *68
turns it back off. (This is
Hi,
Thanks for the reply!
I am still having troubles
I did try:
disallow=all
allow=g726
And still get:
Mar 30 15:49:29 WARNING[-1137677392]: chan_sip.c:2128 process_sdp: No
compatible codecs!
-b
Quoting Michael Manousos [EMAIL PROTECTED]:
Hi,
Bill Hamel wrote:
Hi,
I am running
Perhaps the 7960 has Call Waiting set to YES - this being the case, you're
really not hitting a busy but the 15 sec timeout instead.
Try setting CallWaiting=No in the 7960 you should then get a 'busy'.
Oh, another thing. If you have multiple line appearences configured as the same
SIP phone on
The firebox has the UDP timeout set pretty low by default, this is a good thing
to help prevent DOS attacks, but isn't a really good thing for a SIP device.
There is no option in the GUI to set this.
However you can go into the config file itself and modify the following:
Hi,
I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of
this morning 3/30/04 of asterisk, zap and libpri.
The SIP device I am using is a Sipura SPA-2000 with G726-32 Forced.
When I 'make clean and recompiled zaptel, libpri, asterisk and start asterisk I
can see:
Hi,
First let me apologize if I sent this to the list twice.
Is it possible to kick a caller out of a queue after 5 minutes and goto the
next priority in the context where they were assigned to the queue ?
My desired result is that even though one agent is dynamically logged into the
queue and
Hi,
Is it possible to kick a caller out of a queue after 5 minutes and goto the
next priority in the context where they were assigned to the queue ?
My desired result is that even though one agent is dynamically logged into the
queue and is on a call, I would like the 2nd caller to stay in the
Hi,
Have you checked for IRQ conflicts ?
-b
Quoting Steve Foy [EMAIL PROTECTED]:
Hi,
On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote:
Steve,
this really is a FAQ. You need add to EACH (!) sip user something like
disallow=all
allow=ulaw
allow=alaw
Shot in the dark here ...
Do you have:
canreinvite=no
Set in sip.conf for the SIP phones in question ?
Ciao,
-b
Quoting Steve Foy [EMAIL PROTECTED]:
Hi,
I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a
Steve Foy [EMAIL PROTECTED]:
Bill,
On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote:
Shot in the dark here ...
Do you have:
canreinvite=no
Set in sip.conf for the SIP phones in question ?
No, I don't.
All I have in sip.conf is the general stuff like
I can only speak for the SIP IOS load on the 7960's (We're running 6.1 ) but if
you add:
[EMAIL PROTECTED]
It should work
Note: 7188 being the mail box number and ContextInVoicemailConf
being the context in the voicemail.conf file where the mail box 7188
exists.
Example:
[7188]
type=friend
Curious what your iax.conf looks like.
Also FWIW - if you are connecting directly to VoicePulse with a SIP phone,
wouldn't that mean that you have a SIP account and not an IAX account ?
-b
Quoting yair hakak [EMAIL PROTECTED]:
hello,
after playing with an asterisk configuration for voip
Hi,
Looking around I can't seem to find a way to show the number of agents currently
logged into a queue and if possible who they are. Is there a way to do this ?
Thanks
-b
--
This message was sent using IMP, the Internet
Hello,
I have agents / queues working to the extent that agents can login, logout and I
can send a caller into the queue and the logged in agent's phones will ring.
Maybe I've spent to much time googleing and reading and my eyes are crossing
now, but what I am trying to do is this but cannot
or anything on you phones ?
Thanks
-bh
On Fri, 16 Jan 2004, Bill Hamel wrote:
Hi,
Just got some CISCO 7960 phones. They seem to work great except if I make
any
SIP call using the speaker phone (leaving the hand set in the cradle)the
call
will disconnect in about 5 or so seconds
Atually with the root servers dropping their domain name announcement nothing
would have helped. Well, except for hard codeing the IP rather than using fqdn
in the config. Or making a static entry in the local hosts file ( both have
it's issues)
I prefer to use IP rather than fqdns when
Hi,
Just got some CISCO 7960 phones. They seem to work great except if I make any
SIP call using the speaker phone (leaving the hand set in the cradle)the call
will disconnect in about 5 or so seconds. If I pick up the hand set and make a
call, it's fine.
Has anyone else run into this ? Any
Hi,
FWIW
This issue had been resolved. The fix is nothing to speak of except that maybe
this post may be informative for someone out there.
It turned out to be a hardware issue in the PC, after swapping the Zaptel cards
to another PC, it has been up and running with no ZT_CHANCONFIG failed on
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