This means that no ethernet interface is found for seeding the global
EID. So you will have to set it manually.
:) Pretty clear.
On Thu, Jul 16, 2009 at 11:08 PM, michel freihamich...@gmail.com wrote:
Hi all,
Can you please let me know what the below issue mean when trying to start
asterisk
Great Job Bogdan
On Tue, Mar 10, 2009 at 12:52 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi,
When trying to cluster Asterisk boxes to gain scalability and more
performance, there is now a new simple and efficient solution for doing it.
OpenSIPS/OpenSER 1.5 can now implement
Hi All
I got Asterisk to run on Solaris however I do need it to run in
realtime mode I.e. with the res_mysql file.
Did anyone succeed in this ?
Regards
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Hi All
I got Asterisk to run on Solaris however I do need it to run in
realtime mode I.e. with the res_mysql file.
Did anyone succeed in this ?
Regards
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asterisk-users mailing
Hi
I do have asterisk running in real time I do want to add voicemail to real
time. I did follow :
http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail
However when I do try to make a voicemail I do get :
[Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible
codecs,
Hi
I am trying to get H323 to run on Asterisk, basically I had Asterisk running
so I followed this tutorial
http://astrecipes.net/index.php?n=286
and got h323 to run on my first server on the second server it is just
throwing the error:
ERROR:Failed to create H323 listener
The whole error is :
Hi
I am using
show cannels verbose
to get info about my current sip calls. However, the time displayed is
always zero.
Any hints ?
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With Regards
Ali Jawad System Administrator
http://www.alijawad.org
Phone : +961-01-559031
Mobile : +961-03-041705
Confidentiality Notice: The contents of this E-mail are intended for the
named recipient only. It may contain confidential
Hi
One of my softphones is supposed to support g711 , however I am getting
these errors and a 404 not found when I try to make a call from it. However
on xlite it works fine using g711.
Below is the log of the phone that is not working.
Content-Type: application/sdp
Content-Length: 1123
P-hint:
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With Regards
Ali Jawad System Administrator
http://www.alijawad.org
Phone : +961-01-559031
Mobile : +961-03-041705
Confidentiality Notice
Hi
I followed this howto
http://www.voip-info.org/wiki/view/MeetMe-Web-Control
and
http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html
to install web meetme with asterisk, I know the meetme module is included
however I need to be able to ban and mute users
Thanks for the hint Patrick I appreciate it.
On Tue, Apr 22, 2008 at 3:02 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Using _. is going to result in warnings. A much better practice is to
use _X.
Ali Jawad wrote:
Thx again patrick it worked, I used
[google-in]
exten = _.,1,Set
Hi All
When I dial a number it reaches the asterisk switch as [EMAIL PROTECTED]@123.com
what I need to do is to parse the abc and send it to my pstn gateway
as in
exten = _.,2,Dial(SIP/[EMAIL PROTECTED])
this does work but I do have a varying number of numbers before the @
exten =
of it, it is causing auto fallthrough with status UNKNOWN
On Tue, Apr 22, 2008 at 1:33 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Ali Jawad schrieb:
When I dial a number it reaches the asterisk switch as [EMAIL
PROTECTED]@123.com
what I need to do is to parse the abc and send it to my pstn gateway
Thx again patrick it worked, I used
[google-in]
exten = _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten = _.,1,Dial(SIP/[EMAIL PROTECTED])
while it should have been
[google-in]
exten = _.,1,Set(dst=${CUT(EXTEN,@,1)})
exten = _.,2,Dial(SIP/[EMAIL PROTECTED])
Dear All
I am using gtalk features with my own XMPP server OpenFire
I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls
from clients registered on my XMPP server to SIP devices by calling the xmpp
accounts registered as clients on asterisk.
So far so good. So if I want
Dear All
I am using gtalk features with my own XMPP server OpenFire
I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls
from clients registered on my XMPP server to SIP devices by calling the xmpp
accounts registered as clients on asterisk.
I have sent a previous email
So should I register directly on the asterisk server or should I send
the voice calls through ejabberd to asterisk ?
On Mon, Mar 31, 2008 at 4:55 PM, Philippe Sultan
[EMAIL PROTECTED] wrote:
Hi Ali,
On Fri, Mar 28, 2008 at 5:31 PM, Ali Jawad [EMAIL PROTECTED] wrote:
Hi All
I am
Hi All
I am developing a client that uses libjingle to do xmpp stuff with
ejabberd. I can also make audio calls between those clients. What I am
trying to archive now is to send calls to pstn using jingle. I was
told in the jingle-dev community that asterisk can do that.
Is there any way to send
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