Displaying 20 results from an estimated 8000 matches similar to: "Fwd: RE: Asterisk behind LinkSys NAT Routing"
2003 Nov 02
6
Asterisk behind LinkSys NAT Routing
Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside
2003 Mar 04
3
Fwd: Re: Fax support?
I can't seem to make the fax detection work. Here's
an excerpt from zapata.conf:
signalling=fxs_ks
group=0
context => guestaccess
channel => 47-48
and from extensions.conf:
[guestaccess]
include => incomingmain
[incomingmain]
exten => s,1,Dial,Zap/1&Zap/9&Zap/10&Zap/11|24
exten => s,2,Voicemail,u7000
exten =>
2003 Aug 07
1
Sip Trunk config
incominglimit is already implemented for SIP. Just specify under the
endpoint how many incoming connections are allowed.
For example,
[cisco]
type=friend
username=cisco
secret=blah
nat=yes ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms away
2003 Jul 21
0
RE: Asterisk-Users digest, Vol 1 #873 - 16 msgs
I don't know if 911 uses caller ID or BTN (Billing Telephone Number)
900 calls, operator calls, and 800 calls use the BTN not the Caller
ID...
Anyone????
3. Re: E911 and asterisk (Martin Pycko)
Message: 3
Date: Mon, 21 Jul 2003 12:05:38 -0500 (CDT)
From: Martin Pycko <martinp@digium.com>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] E911 and asterisk
2003 Sep 17
1
Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
----- Original Message -----
From: <asterisk-users-request@lists.digium.com>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, September 13, 2003 7:55 PM
Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
> Send Asterisk-Users mailing list submissions to
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2004 Jan 13
1
max queue time; newbie question (fwd)
Martin Pycko <martinp@digium.com> writes:
> sure, use the 'n' option of the queue and put voicemail app as the next
> priority
Will that work? From my read of the code, the timeout parameter is
only checked while the call is being sent to an agent's phone (inside
the try_calling function). The timeout doesn't seem to be checked
while the user is waiting to get to
2013 Jun 28
1
Asterisk behind NAT and Kamailio --> Internal IP in SDP and not "externip"
Hi,
We have some Asterisk servers that we are moving behind a NAT to
preserve public addresses and make room for growth. This is Asterisk 1.4
NAT works very good with the externip/localnet-setting when we are
connected directly to our teleco. But when I try to use NAT and put them
behind our Kamailio something interesting happens: The media-address in
the SDP is the internal ip and not the
2004 Dec 14
1
Asterisk to sip client behind Firewall/NAT-cancall but cannot receive calls ?
As far as I can remember I only opened sip and tftp ports for the phone.
For some reason (didn't look into it too much) the call stays with sip
and doesn't use RTP.
The problem you describe (the call doesn't even ring on the other side)
is something I had and was solved by upgrading the firmware.
Checkpoint's tracker explicitly said what connection attempts were
blocked and why.
2007 Oct 18
2
A linksys SPA921 behind NAT and firewall
I've got someone sat in a home-office with an SPA921 behind NAT, and
most probably a firewall. I've got a STUN-server running, and calling
in from the SPA921 to our Asterisk box works fine - though I had to
open the firewall for UDP traffic on port 10000-20000.
Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this
is due to the NAT/firewall on the other side,
2004 Jan 26
1
SIP behind NAT - use of "externip" option
I am having difficulty configuring SIP behind NAT (using latest CVS).
Using sip.conf:
[general]
port=5060 ; Port to bind to
externip=ww.xx.yy.zz
bindaddr=0.0.0.0
nat=yes
register=>[userid]:[password]@voiptalk.org/2000
[voiptalk.org]
nat=yes
externip=ww.xx.yy.zz
type=friend
secret=[password]
nat=yes
reinvite=no
canreinvite=no
I fail to register. SIP Debug gives:
SIP
2003 Dec 30
2
* crash when forward voicemail message [problem solved]
Thanks for all your help Martin,
Guys,
This is a good find and hopefully could help someone else.
I've been having a problem with forwarding voicemail from one mailbox to
another. I ran down the sendmail and soundcard path and came up goose eggs.
With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9
Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it
2011 Mar 02
2
asterisk behind nat
I'm running asterisk on a Freebsd with 2 Nic's.
Inside NIC is 192.168.5.x where the phones are.
Outside NIC used to be a public IP with the ISP's device set to
bridging, but the new WiMAX router only offers me the public ip
94.18.x.x on the outside,
and forwarding everything to 192.168.1.50 on the "Outside NIC"
Some of the phones are being disconnected with Asterisk
2003 Apr 04
0
non-telephony use of T400P?
Another issue to consider is T1 framing. If your application is putting
bits onto the T1 at the rate of 1.544 Mbit/s then the T1 would need to
be unframed. I don't believe this is an option in zaptel! If however,
it is putting bits on at a rate of 1.536 Mbit/s and adding 8000 bit/s
for framing then you may be able use the suggestion below.
Don Pobanz
On Thursday, April 03, 2003 3:28 PM,
2009 Jan 29
2
Don't get asterisk to run behind NAT router
Hi people!
I am not getting smart getting asterisk 1.6 behind a NAT to run.
1. I enabled IP forwarding on debian linux
2. told asterisk in "general" that he is behind NAT and mentioned him
his external static IP Adress as well his domain in the outside world.
If a client who is connected with a DSL modem calls me, a grandstream
module in the LAN behind the router, in the same network
2005 Jun 02
3
asterisk on internet sip phone behind nat - doessomeone even have this working
Lance,
Have you configured your sip.conf to use these aprameters under General?
;externip=66.213.227.66
;localnet=192.168.1.0
;localmask=255.255.255.0
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance
Grover
Sent: Thursday, June 02, 2005 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial
2004 Dec 20
0
Calling SIP Address From Behind NAT
My asterisk box is behind a NAT firewall. I have friends that are on
Earthlink, Vonage, etc.
I'd like to make VOIP calls directly to them rather than going through the PSTN.
With Earthlink, I can make this work through FWD peeting numbers, but
that's sort of a waste of FWD bandwidth.
WIth Vonage, it doesn't work. I suspect this is because of the
breakage between FWD and Vonage that
2005 Mar 15
0
trying to get trunk to register with * behind NAT
i've got * and phones in small home network all behind NAT. Outbound to
iconnect proxy works great. Now to get in/out working with another carrier.
Carrier2, Commpartners, i have working with one of the phones and a soft
phone without * just fine.
Next I register the phone with * fine. Create a trunk, but it the trunk
fails to register... help
I'm getting the following msg during
2007 Feb 08
0
SIP Re-Invite behind a NAT
SetUp:
- Asterisk behind a NAT,
- Red Hat 9.0
- Asterisk 1.2.14
My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have
my dial plan set up so that when outside callers dial the DiD, the
call is answered by my auto-attendant. The caller can then select who
they'd like to speak to and the call is transferred to the external
line associated with that person (usually a mobile
2006 May 17
0
Asterisk SIP Gateway behind NATS - SIP/2.0 404 Not Found
Hi all,
I am running an Asterisk server behind a NAT.
I want to forward the calls from PSTN to a SIP phone (no nat and also an
asterisk).
I set the externip and localnet in sip.conf already. I opened the ports
in my firewall. (I changed SIP port from 5060 to 5065 and limited the
rtp port to 12000-13000)
However, I just can't call out. I've always received SIP/2.0 404 Not Found.
My
2003 Nov 04
2
Does externalip= do anything to help with SIP behind a Linux based NAT router?
I'm just curious if I was to place my * box behind a a FW/NAT box
running linux, if my SIP calls will still work. Box right now is a RH9
computer using iptables as the FW. I wouldn't mind placing my * box
behind it, but I'm wondering if anyone has actually gotten NAT working
with *?
Thanks,
--
+------------------------------------------+
|Leif Madsen -