Displaying 20 results from an estimated 200 matches similar to: "sip.conf"
2003 Jun 05
1
dl102s again
Please I need help, I don't know why,almost every time I dial on my dect
phones, the dialtone doesn't go off and * doesn't recognise anything!!!! I'm
using two dlink voip gateways, MGCP: DL102s. Any ideas?
thanks in advance
michelle matis
-----
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2003 Jun 11
1
some sip questions
<P>I write the email again, cause the first one I have had problems while sending it. Here is the email again:</P>
<P>Hi everybody,</P>
<P>I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! </P>
<P>1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two
2003 Jun 11
4
some sip questions AGAIN
I write the email again, the third time!!, cause the other two ones, I have
had problems while sending them. I hope this time it works. Here is the
email again:
Hi (and sorry) everybody
I'm starting with SIP and I wanted to ask some questions, perhaps silly
ones, but I hope people can answer me!
1) Which codecs may I use? I want the SIP phones to call to the PSTN
above all, but I have
2003 Jun 16
2
The same SIP problems...SORRY!
Hi eveybody again!
I don't want to be annoying, but if nobody can help me with this, I'll have to
desist of working with SIP.I have some questions about SIP, as I wrote in
another mail. I have a SIP Gateway and I have two phones (an analog one
and a DECT one) conected to it.Also, I have two Dlink dg102s with four
phones conected to them. The main problems are two.
Calls between the
2003 Jun 16
8
SIP REGISTER
Hi!
I have a new problem with my SIP device.I have done some changes and
now I receive continuosly a SIP message: "501" "Not impelmented" back
from the SIP Gateway. I can see that it doesn't register to Asterisk.
I have in the SIP device:
Registrar 1: UnRegistered to: 2222
registrar: 188.208.12.237 5060 expires: 2000
name: gateway passwd: 123
My
2005 Feb 08
1
Fastagi question
Hi All,
I have a question about Fastagi because I can't get
it to work for some reason. Everytime I execute the
fastagi command, i get an error:
my extensions.conf:
..
exten => 1000,1,agi(agi://some_ip_address)
..
output from asterisk console:
-- Executing AGI("Zap/1-1",
"agi://some_ip_address") in new stack
-- Launched AGI Script
2012 Jan 05
2
suppressing openssh server identification
With all of the discussions regarding getting p3wned, I am feeling paranoid and can't seem to figure out how to suppress this...
telnet $SOME_CENTOS_5_SERVER 22
Trying $SOME_IP_ADDRESS...
Connected to $SOME_CENTOS_5_SERVER.
Escape character is '^]'.
SSH-2.0-OpenSSH_4.3
'Banner no' doesn't do it. Is it possible to suppress the version?
--
Craig White
2003 Jun 05
0
dl102S
I'm using * as a Call Agent for two DL102S but I have some problems, like the tones not being sending from the phone to the *. I have not changed the configuration of the DL, except the IP and the Notify Entity (*). Must I change another thing in * or in the device? Thanks very much michelle
<br><br>
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2003 Jun 11
0
(no subject)
<P>Hi everybody</P>
<P>I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! </P>
<P>1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't
2005 Jan 21
1
Where is the * servers IP defined for sip phones?
This I am sure is a very easy question, but I can't seem to find the
answer.
Here is the scenario:
cisco 7940g phone has SIP 6.3 firmware applied
the file SIP<mac>.cnf does not seem to have a place for it:
image_version: P0S3-06-3-00
#line 1 settings
line1_name: "5010" ; Line 1 ExtensionUser ID
line1_displayname: "5010" ; Line 1 Display
2003 Apr 28
4
adsi phones
Can anyone recommend some phone sets that are adsi compliant and work well
with asterisk?
2005 May 12
4
gnugk
Hi
I've a problem with a gnugkv2.0.7
I've compiled gnugk successfully
I've installed PWlib-1.6.6 and openh323-1.13.5 libraries successfully
When i run gnugk i have this error:
error while loading shared libraries liboh323_linux_x86_r.so.1.13.5 cannot
open shared object file No such file or directory
I try to use command export:
export
2003 Jun 15
1
SIP REGISTER behavior change: specific domains possible in REGISTER
Mark has fixed the REGISTER issues to be more RFC compliant. I've
created a new thread so that those of you who got bored with the old
thread might read this new one. The feature that has just been added
was added a while ago, but now it actually seems to _work_. :-)
If you have a SIP server to which you are trying to REGISTER, and
they demand valid domain (the part after the
2004 Apr 01
1
samba oplocks ...
I've tried to configure samba to lock files bewteen windows and linux but i couldn't i've
read a lot of messages here, but trere is no one that have something about the file smb.conf.
i have this in my global secction but i doesn't work
[global]
workgroup =3D GMC
create mask =3D 0777
os level =3D 16
directory mask =3D 0777
hosts allow =3D
2003 Apr 29
28
H323
Is H323 built into the current CVS? If so, could someone give me an idea of a simple config?
thanks,
darran
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2005 Mar 10
0
Re: Polycom phones do not talk to each other
>Also, I'm sure you've probably checked on this one,
>but are the phones registered with asterisk?
>You can make outbound calls on them without them
>actually being registered. I'm assuming you can
>still get in and see the CLI. What does "sip show peers"
>look like? What does "sip show peer xxx" show?
>What does the CLI show when you
2009 Jan 21
0
About Asterisk 1.6.0.1
Hi asterisk users,
I am in need of information about how to configure the
sip.conf and extension.conf for subscribers to support the dialog event
package rfc 4235. I am using asterisk 1.6.0.1 version.
The below are the configuration of sip.conf and extension.conf files
which I have done.
I have three subscribers as one from my application(App) and other are
x-lite1 and
2017 Aug 22
1
Error in .jnew(“java/io/FileOutputStream”, jFile)
I don't know R absolutely, but I have to do this work for my diploma. So I'm sorry for strange message below. Please, help me anybody decides this issue.
If you need any information, I'll show you all what you need.
**error**:<br>
> #save
> #csv
>file.create("C:\\Users\\Sapl\\Desktop\\NATA\\code\\Results\\Created by R\\ab_ret_banks_short_form_10.05.2006.csv")
2009 Jan 22
0
Query About Asterisk 1.6.0.1 Dialog Event Package.
Hi asterisk users,
I am in need of information about how to configure the
sip.conf and extension.conf for subscribers to support the dialog event
package rfc 4235. I am using asterisk 1.6.0.1 version.
The below are the configuration of sip.conf and extension.conf files
which I have done.
I have three subscribers as one from my application(App) and other are
x-lite1 and
2005 Mar 11
1
Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI.
I am using PBXware for configuring users and extensions.
Pbxware uses Internal script called init.sh to process the calls
based on its own version of extensions.conf defined in the GUI.
I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51.
I have used IAX2 extension 101 and dialed SIP Extension 51
But the PBXWare's Init.sh AGI command identifies the DNIS
as another IAX