Thanks Dave and Geraint for the reply,
I'll be really specific: What does the realm= and the domain= in
sip.conf actually control?? And how do they relate into Guest INVITE
messages ?
Dave - yes you've got it pretty right:
I'm basically dialling a number (5550) from a sip client to
David Backeberg schrieb:
On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote:
all server are in one rack in our datacenter and are connected to an HP
Procurve 2650 switch, which has been setup around 3 months ago, cause of
the old switch died silent in the night.
all server
I'd like to see that link too!
I use Cisco 7940s at the moment, and would like to see how to hook them into AD
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews
Sent: 26 May 2009 15:56
To: Asterisk
Hi All,
Can I use Asterisk IPV6 build for making an PSTN gateway? I read in the
asterisk ipv6 web site that this build would support IPv6 for SIP protocol only
and not h323 . Could anyone please tell me if this means that asterisk IPv6
build cannot work properly as a PSTN gateway?
On Thu, May 28, 2009 at 10:49:38AM +0200, Stefan Schmidt wrote:
David Backeberg schrieb:
On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote:
all server are in one rack in our datacenter and are connected to an HP
Procurve 2650 switch, which has been setup around 3 months
Hi,
I finally solved the problem.
As I mentioned in one of my earlier postings, I forgot to install libpri
when I compiled the dahdi package for the first time. I fixed that but
did not compile asterisk again. Therefore the chan-dahdi.so obviously
did not contain the necessary code to react
Alex Samad schrieb:
Hi
Hi Alex,
I am new to asterisk so my suggestions might be a bit silly.
Why not setup a iax2 connection bettween the asterisk servers, because
its a lower overhear and more efficient.
We had changed from iax connections to sip connections cause we had
timing
Hi,
sip.conf is missing in /etc/asterisk after installing the package AsteriskNow
1.5 Beta release?
Is there any guide for FreePbx Administration?
Thanks
Explore and discover exciting holidays and getaways with Yahoo! India
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Alright, by popular demand, here they are:
All of the scripts are written in PHP (so I'm kind of partial :) and you'll
need to have it compiled on the asterisk box with cli and ldap to hook into AD
and be useable from the command line. Beyond the auto-provisioning script, the
remote reboot
Hi,
I have been trying to get T.38 to work with clients behind NAT for the past
week but with no success.
I have an asterisk server on the public internet and several Grandstream (I
tried Linksys too) HT502 ATAs behind NAT in different locations.
I tried every possible combination of NAT,
Stefan,
I'm not sure if you've considered the underlying hardware, firmware, and device
drivers yet, but I was brought in to evaluate a site under heavy load and was
able to stabilize things by applying vendor supplied drivers and firmware to
the system.
What is the underlying hardware (E.g.
On Fri, 29 May 2009 01:52:08 Antoine Megalla wrote:
Hi,
I have been trying to get T.38 to work with clients behind NAT for the past
week but with no success.
I have an asterisk server on the public internet and several Grandstream (I
tried Linksys too) HT502 ATAs behind NAT in different
I never tried it with Asterisk, but from what you describe you have
got either a codec issue or NAT issue.
On Mon, May 18, 2009 at 6:57 AM, Si Tai Fan s...@hktelecoms.com wrote:
Has anyone tried the Panasonic KX-HTG100CE with asterisk?
Mine works when I call between extensions but when I call
It has been suggested that I should do my Asterisk tutorial
(http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html)
using newer software, OK.
I hope this is not opening a big can of worms, as I am sure there are a
lot of different opinions about this, but:
For a low/no growth
Since you opened this Can-O-Worms, Digium implicitly endorses
Scientific Linux and SVN branches using Zaptel, based on my findings from
SwitchVox. This being said, I'd probably go with 1.4.21.X since anything
above that replaces zaptel with DAHDI. There are still a lot of things To
be worked
Hello all,
I have a need to be able to use the originate AMI command to dial out to
the PSTN, have that person answer and then have the second PSTN
connection dialed out.
I have tried to use:
Action: Originate
Channel: sip/number@provider
Context: default
Exten: othernumber
Priority: 1
users.conf
[108]
username = 108
transfer = yes
mailbox = 108
call-limit = 100
fullname = General Messages
registersip = no
host = dynamic
callgroup = 1
context = DLPN_DialPlan1
cid_number = 108
hasvoicemail = yes
vmsecret = 1234
email = du...@dummy.com
threewaycalling = no
hasdirectory = no
On Thu, 2009-05-28 at 12:58 -0500, Danny Nicholas wrote:
This being said, I’d probably go with 1.4.21.X since anything above
that replaces zaptel with DAHDI. There are still a lot of things “To
be worked out” in DAHDI – Zaptel is a pretty solid standard.
It continues to amaze me when I
There are over 100 open items on the Digium board related to DAHDI. I'm
sure quite a few are chair-to-keyboard issues, but here are two real ones:
1. DAHDI will not look for a connection when dialing, so if I make a call
and play a file, X percent of the file plays before the person actually (if
Danny Nicholas wrote:
users.conf
[108]
username = 108
transfer = yes
mailbox = 108
call-limit = 100
fullname = General Messages
registersip = no
host = dynamic
callgroup = 1
context = DLPN_DialPlan1
cid_number = 108
hasvoicemail = yes
vmsecret = 1234
email = du...@dummy.com
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any potential sniffers or inside hackers
Please help or suggest any solution that you feel may help
Kind regards
Sam
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any potential sniffers or inside hackers
I have reviewed and shall soon try:
Hi,
Like me, some of you probably remember Jim as one of the pioneers
along with Leif and Jarod. These guys wrote the book, literally. Jim
is our guest tomorrow and he'll be talking about system building,
among other things. We always have a good time AND get stuff done on
the Conference so come
On Thu, May 28, 2009 at 02:00:15PM -0500, resea...@businesstz.com wrote:
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any potential sniffers or inside hackers
On Thu, May 28, 2009 at 01:32:58PM -0500, Danny Nicholas wrote:
There are over 100 open items on the Digium board related to DAHDI.
Which is because people use it. I wonder which of those are actually
regressions from before the DAHDI times (not that others are less
serious. But this is what i
Hello,
I am installing asterisk, libpri and zaptel.
I have it setup for EM wink.
incoming calls are working.
outgoing calls are not.
zttool shows TxA as 1100
RxA shows
This doesnt seem like em_w signalling?
Seems like the PBX is not setup for EM_w.
Is that the case?
Jerry
There are actually 793 items, so I guess a lot of folks are using it. The
bug number for #1 is 14935. I developed a similar app that was working
great in the 1.4.21/Zaptel environment, but is now iffy at best in the
1.4.25/1.6 environment.
https://issues.asterisk.org/view.php?id=14935
I'll
Probably not, otherwise incoming shouldn't work. What are you doing for a
Dial command (exten = s,1,Dial(Zap/1,w5551212,60) )?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, May 28,
Probably not, otherwise incoming shouldn't work. What are you doing for a
Dial command (exten = s,1,Dial(Zap/1,w5551212,60) )?
My dial command is a Zap/1/91317506 for my cell.
It says attempting call...
unable to
Couple of things:
Does native zap dial work (exten = s,1,Dial(Zap/1) ) ?
You should put a w in front of the number (wait .5 seconds before
proceeding) and give the call at least 30 seconds to connect.
What do you get from zap show channels and zap show status?
_
From:
On Thu, 2009-05-28 at 14:47 -0500, Danny Nicholas wrote:
The bug number for #1 is 14935. I developed a similar app that was working
great in the 1.4.21/Zaptel environment, but is now iffy at best in the
1.4.25/1.6 environment.
If this is an analog line connected to an FXO port, then Asterisk
Clearly the problem is related to option #1. Change confuses some people. :-)It continues to amaze me when I hear this, as there really isn't much
difference between Zaptel and DAHDI. In fact, the only two differences
I know about are:
1) The name change
2) Making software echo can modules able
On Thu, May 28, 2009 at 02:15:08PM +0200, Stefan Schmidt wrote:
Alex Samad schrieb:
Hi
Hi Alex,
I am new to asterisk so my suggestions might be a bit silly.
Why not setup a iax2 connection bettween the asterisk servers, because
its a lower overhear and more efficient.
We
Hello
Please help me, i need transfer a call in asterisk to other telco number and
free the channel. Can i do with any q931 function?.
Thanks a lot
Aris...
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Are conference bridges and other resources going to work with SRTP ?
I'm wondering what enabling SRTP will break in Asterisk. It breaks
several things in Cisco CallManager. Also wondering what make/model SIP
phone you are using for SRTP and what experience other having using that
make/model for
Hi guys, I am new here but I have a quick question.
I have an incoming trunk that sends me calls from various usernames I have
with them. Only trouble is they send invites as s...@my.ip.addr, not as the
username I have with them. So I cannot match extensions like I would want
to.
Here is a
Good morning
How I use the described commands below to limit the number of simultaneous
calls saw voip providers that they can be effected and be received in the
trunk in the Freepbx?
I verified the commands incominglimit and call-limit as I can use asterisk
is version 1.4!
It would like to
Hi all,
I have installed asterisk latest stable version 1.6.1.0, with dahdi
driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now
it wont work with 1.6.
I managed to register my phone on asterisk. but i cant hear any dial
tone on my phone. these are my configs. it will
Hi,
in Asterisk 1.4 to limit the simoultaneous calls I use the following
parameters:
[general]
...
limitonpeers=yes
notifyringing=yes
[phone]
...
host=dynamic
username=phone
call-limit=2
So I can receive and make max 2 calls simoultaneous.
Fo me that's work fine.
2009/5/29 Yuri
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