Displaying 20 results from an estimated 30000 matches similar to: "[RTP] PSTN -> Gateway -> Phone"
2006 Nov 20
2
How to secure access to PSTN line through Linksys gateway?
Hello
I successfully hooked up a Linksys 3102 SIP gateway
(http://www.voip-info.org/users/683/21683/images/716/SPA3102_lrg.jpg) to an
Asterisk server, but since it's connected to a PSTN line, I must make sure
it cannot be used by unauthorized users from the Net. Actually, even legit
users with an account on the Asterisk server shouldn't be able to use it
(outgoing calls should go
2005 Aug 25
1
VoIP Mythbusters Help!! - NATed phone-phone connection without proxy? Possible? Yes/No
Hello,
All I'm looking for is a yes/no answer here. I have heard that the
following scenario is possible (reasonably easy to implement as well) . but
I just don't get it :-) . if it is possible I'll go ahead and learn on my
own, I just don't want to waste time on something that will not work.
Scenario:
2x VoIP phones
- Each phone is configured to register
2007 Jul 05
1
SIP / STUN / Network - Help!!
Hi Everyone.
I'm in a quandry & don't know which way to go. - Obviously I'm an Asterisk
newbie although I've been watching this list for over 2 years now.
I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running
here at home. - It's on my home LAN - NAT'ed behind my LinkSys router. - On
the same LAN I've got a Cisco 7940, 7960, and
2008 Feb 03
1
Multiple SIP phones behind a Linksys firewall
And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall?
In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio.
-----Original
2005 May 31
2
Sipura 2000 behind NAT issue, Vonage is working
Hi,
I'm trying to configure Sipura 2000 (behind NAT) which connects to
Asterisk (public IP, no NAT) and having interesting results. When Sipura
is behind Linux/NAT firewall it works great and no special NAT settings
on Sipura are necessary. The issue I'm having is when Sipura is behind
Linksys broadband NAT router. Sipura gets registered with Asterisk just
fine, but I can't hear
2009 Mar 05
0
Stun with hosted asterisk solution???
Howdy,
I have the following issue and would like to know if anyone has got around this before.
IP Phones - Linksys 942
Sip server - Asterisk 1.4.13
Stun server - Vovida
Ok heres the issue. We have multiple client phones on their own network behind a natted connection. We have setup the phones to be natted and also pointing to our stun server. Now when the phones make an outside call to the PSTN
2006 Nov 29
2
Setting RTP ports for Asterisk?
Hello
When I make calls from home to the PSTN by going through the Net ->
Asterisk -> the Net -> VoIP provider -> PSTN, I get no sound either way. I
assume it's because I must tell Asterisk to use fixed ranges of UDP ports
and map ports accordingly on the NAT firewall under which it is located on
the LAN at work.
Here's the schema:
home > NAT > Internet > NAT
2007 Dec 10
1
T.38 fax solution, opinions?
Hi,
I'm putting together a fax solution for my company that utilizes T.38. I
wanted to throw out my plan and get some feedback if anyone is doing
something similar or sees a blatant problem with it.
We're currently rolling out SPA-942 phones for the standard desk phone with
vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls
for satellite offices are handled by
2010 Dec 25
1
Remote VOIP/SIP Phones through two routers
So, assuming your Asterisk box is behind one firewall (Linksys/Tomato
Software) and your Wireless SIP phone is behind another firewall
(SonicWall 1260 Enhanced). Is there anything special that I have to do
to the firewalls. I do have the Asterisk firewall configured to work
(ports 5060 & 10001-20000). But I'm not sure about the other end. Do I
need STUN at the SIP Phone end? Do I
2003 Sep 12
3
7206 as SIP->PSTN Gateway?
All,
I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway.
Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know
which cards, if any, exist for a 7206VXR to act in a similar capacity,
either as a T1/PRI, DS3, or POTS FXO/FXS?
What other Cisco routers can act as SIP gateways today?
Thanks,
Dave
2006 Apr 02
0
no audio between sip channels * 1.2.6
Hello all,
I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until
recently all was good. on Friday I was running 1.2.5 when I added the fourth
phone. I have to admit to initially wiring the rj11(crossed wires) wrong the
first time but other than that nothing I can think of. Added the appropriate
entries in sip.con and on the PAP2. I then tried to call from one line to
the
2007 Jul 12
0
No subject
<asterisk-users at lists.digium.com>
Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys
firewall
Date : Sat, 2 Feb 2008 18:25:16 -0700
> And the firewall is in between the phones and both servers or are you
registering the phones on a local server and trunking to the other server
through the firewall?
>
> In terms of nat and Cisco 7960s I've never had a
2010 Dec 06
1
[3102] How to rewrite CID name + number?
Hello
I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:
http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png
The problem is that by default, Asterisk doesn't rewrite the CID name
+ number in incoming calls, so that XLite displays whatever name I
used in the 3102 and the extension the 3102 uses to register with
Asterisk.
How can I tell
2010 Jan 28
2
rtp.c:883 ast_rtcp_read: RTCP Read too short
Hi:
I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?:
?
-- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack
??? --
2005 Jul 05
0
Re: [Serusers] NAT considerations...
You will also need your SIP clients that are behind the same NAT to
support ICE (Interactive Connectivty Establishment) if you want calls
between them. Xten Eyebeam and Snom phones are the only ones I'm
aware of that support it.
On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote:
> And even worst.
> There are some kind of NAT that STUN does not work.
> You can check
2004 Apr 18
2
grandstream and stun
Hi,
I noticed some issues with how grandstream handles
stun test. GS is running version 1.0.4.50. First I
reset the NAT router. Then reboot GS, get results of
"restricted cone". Immediately reboot GS, get results
"full cone". I tried quite a few public and commercial
stun servers. Also tried different model/version of
linksys routers. I always got the same issue. Winstun
on
2005 May 24
3
Budgetone and NAT not working
I have a couple of Budgetones that I am playing with trying to get them
to work with * from a remote network over the Internet (yes NAT joy!).
My * server is in my DMZ and I have 5060 and my RTP range forwarded
(UDP) to my public address (through a Cisco PIX). Internally, I can
setup my budgetone, it registers and works great. I then have a Linksys
router connected to another Internet
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
Asterisk is on public IP (as described in the first email)
i have 10 years experience in voip, 4 years webrtc in production. i know
about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism
but i confess. i dont understand WHY Asterisk SOMETIMES switches
destination IP in RTP. this is not only about ICE. its about RTP engine
too which is Asterisk specific
and Asterisk DEBUG is
2006 Feb 25
1
Asterisk as a dedicated Analog PSTN gateway
Hi there,
I was wondering if anyone has successfully used Asterisk as a dedicated
Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or
an 8 port model?
Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need
an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the
Mediatrix 1204 does a very good job and I figure I can build a much more
2008 Nov 21
0
PSTN Gateway setup
Hello list,
I recently bought a Linksys SPA400 as a PSTN gateway. The gateway is
connected to
an * server and i have 10 users using this setup. I do have some
problems in establishing
a call to an outside location (call that goes through the SPA400). The
first attempt doesn't
get through.
I suspect the spa400 being the source of the problem. The Linksys
SPA400 has a lot of
params on the