Displaying 20 results from an estimated 300000 matches similar to: "how to make a ip to ip call"
2007 Apr 13
5
SIP REGISTRATION TIME OUT
hi!
First of all i want to tell i have a dedicated server on layeredtech
with direct internet connection and i currently dont use iptables, so
this is not about network configuration =).
well so, i install asterisk-1.4.2 on my server, and next install
asterisk-gui from the digium repository.
next i go to:
http://pbxa.com:8088/asterisk/static/config/cfgbasic.html
and install a default
2005 Jun 08
5
Xlite not communicating with Asterisk
Dear All,
I have downloaded the xlite version 2.0 for windows and I made the
following conf in the xlite itself as the document suggested in order to
make it work with Asterisk but still it doesn't work as a matter of fact
when I tried to make a tcp dump I can see no packets going between the
windows client and the Asterisk server at all, here is the my conf on
the xlite itself:
in the
2011 Feb 24
1
Using a Virtual IP Line
Hello!
I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured.
I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.
2008 Apr 03
0
NAT when outbound call leg is not a local subscriber?
Hi,
I have been experimenting with NAT and Asterisk a bit now. Though I have
made progress along the way, I have come across the following problem. I'll
really appreciate if anyone can provide me any help or pointers. Thanks!
Successful Scenario:
-------------------
All sorts of NAT calls are successful with full two-way media when both end
points are locally subscribed users.
Problem
2005 May 11
1
Trouble Connecting Xlite to Asterisk
I just installed Xorcom Rapid and I'm trying to connect with Xlite.
In my SIP Proxy I have set the Domain/Realm and SIP Proxy as the IP Address
of the new install. I can ping that box.
When I try to connect I get hung on the "Awaiting Proxy login information"
and the log reads:
========================================================================
? 2004 Xten Networks, Inc. All
2005 May 26
1
Asterisk con X-lite : Register Ok but no calls (404 Not found)
Hi all,
I'm working on an implementation of VoIP en Linux.
I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a
Red Hat 9.0 (*.*.*.172) with another softphone X-lite.
Both of the softphones are registering and appear in the peers (sip
show peers) with the good parameters of address and port.
If I try to make a call, * receive the INVITE request and send a 404
NOT FOUND answer.
2009 Oct 30
1
Cannot make calls
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000066">
Hi all,<br>
<br>
I can only get a line signal when I set the phones to not register
with domain . <br>
<br>
All phones are in the same NAT and I cannot make calls.<br>
2005 Feb 26
0
Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite....
XLite does not support transfer... You have to buy their XPro
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mateo
Meier
Sent: Tuesday, February 22, 2005 3:50 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to
forwarda call to X-Lite....
Hey Guys
Im
2005 Feb 28
0
Anybody using X-Lite Softphone ? tryed toforwarda call to X-Lite....
Try the snom soft phone! http://snom.com
CS
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> Dave Chase
> Sent: Saturday, February 26, 2005 12:31 PM
> To: ich@mateo.ch; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Anybody using
2005 Aug 24
0
Re: [Serusers] SER IP PBX for multiple clients
Waldo,
How do you let your customers manage 'their' PBX. I too have a setup
like you. However, I installed a * server for each customer, via
vserver. I'd like to now what kind of software/webbased package you use
for this.
I also have SER installed as a front-end server for the * servers. But,
as I'm still not very into SER, don't know exactly how this fits in.
Should I use
2005 Feb 28
1
Anybody using X-Lite Softphone ? tryedtoforwarda call to X-Lite....
Mateo,
Dialing the extension to your softphone is the same as any hardware
extension.
Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty
exten => 1000,2,Macro(vmessage,1000)
exten => 1000,3,Hangup
Change [mateo01] to [1000] in your sip and you will be saying that ext.
1000 is registered with the specifics you are using.
Update the settings in your softphone to register the name and
2005 Mar 27
1
Asterisk and XLite on same machine (OSX)?
Dear all,
I have tried to run an asterisk instance together with XLite on a single
machine (a PowerBook).
The intent is to take advantage of IAX connections to easily cross NATs
while traveling.
While the IAX setup proved 'easy', just having to fiddle a little with
working configs at both sides, I did not succeed so far in getting XLite
to connect to the local Asterisk server, AND be
2005 Mar 24
1
direct ip-to-ip call
Hello!
I'm searching for a way to call ATA (IAX or SIP) that is not registered
with any server or proxy.
Is it possible to make such a call from a softphone to an ATA just with
IP? Something like (sip:// or iax://)1111@210.12.34.45 (where
210.12.34.45 is ATA's public ip)?
Regards,
CuPoTKa.
2005 Jan 05
1
Can't initiate a call with X-Lite.
Hello,
I'm trying to place a call to asterisk using X-Lite. Asterisk is setup
with some Grandstream phones. I can call from one grandstream extension
to another. When I try to an extension with X-Lite, it comes back with
Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk
extension. It is just sending a sip invite to extension@IP. Does the
X-Lite need to connect to
2007 May 16
0
NO ANSWER, When openser make an oubound SIP call to my asterisk
Hi all,
I try to make a call from my Openser(SIP Proxy) to the asterisk in different
machine.
I use my asterisk as a trunking gateway.
I can make a call from my openser to some trunking gateway such as my cisco
5300 or welltech 5250.
In the same method, I try to make a call to asterisk ( sip listen on udp
5060 )
I use ngrep on my asterisk machine and list as below.
But I can't find any sip
2003 Sep 20
4
Maximum retries exceeded w/SIP
First of all, I'd like to send a big "thank you" to all the folks who have
helped me get this far.
Now on to the next problem. Here's my current network setup:
The Big I ---+--- FreeBSD FW --- * (10.0.0.253) ---- PC (10.0.0.1)
|
+--- Laptop (public IP)
natd is set up with the following rules:
redirect_port udp 10.0.0.253:10000-20000 10000-20000
2006 Jan 31
0
unable to register using SIP
Sorry for the duplicate post but I have hit a brick wall trying to get
this to work. Is there anyone who can help me?
I am having trouble trying to register with a voip
provider using sip. I am able to connect using xlite softphone. in
xlite i use
domain/realm: providerdomain.com
sip proxy: host.providerdomain.com:9000
this difference in domain and sip proxy host is whats causing
2005 May 16
2
NAT and sip issues
I have an asterisk server behind NAT - no audio on the test external calls I
have tried making so far.
Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution
evident from there, sounds like I have case 9. I would have thought that all I
would have to do is port foward and have the external IP on the asterisk server,
which I have done
I have fowared 5060UDP, 8000UDP, and
2017 May 22
3
SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)
Hello List
I work at an SIP Provider and we have added and SBC in front of our
Voice Switch to protect it.
This requires all our SIP Trunk customers to register via a 'proxy'.
I struggle with Asterisk to work over a proxy.
This is what I have done so far.
register => username at sip.example.com:password at sbc.example.com
This works fine, asterisk is sending registrations via the
2005 Jan 21
0
Cisco 7960 can't make/receive calls
I've got three 7960s running v6 SIP firmware. My Asterisk setup has
worked fine with grandstream devices, and basically, we're just
upgrading to use nicer phones.
Whilst I can make/receive calls from the 7960 to/from gossiptel).
When I try to place a call, I get the following
Jan 21 11:09:23 NOTICE[19688]: chan_sip.c:7271 handle_request: Failed to
authenticate user "30"