Displaying 20 results from an estimated 10000 matches similar to: "Asterisk + Sip phones on Nat"
2003 Nov 28
2
Deltathree icomming problem
Hi,
I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :((
I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :(
This is my configurations files:
- sip.conf -
[general]
port = 5060
2004 Dec 06
2
Budgetone 101 phones ? SIP through NAT ?
I'm new to VOIP. We are thinking of setting up a VOIP system between a
couple remote offices. I've been lurking on this group for a while.
What is the consensus on these phones:
http://www.netvoice.ca/grandstream/budgetone101.htm
I'm confused about the SIP protocol... can a SIP phone be located behind
a NATing firewall ?
When people use asterisk on a broadband connection used
2003 Oct 19
1
Music on hold...
No, you don't need a sound card.
Do you have ztdummy loaded or zaptel device in your system?
Regards,
Gus
----- Original Message -----
From: "Chris Hariga" <contact@techselesta.com>
To: <asterisk-users@lists.digium.com>
Sent: Sunday, October 19, 2003 8:19 PM
Subject: [Asterisk-Users] Music on hold...
> Hi,
>
> I need a sound card and mpg123 for music on
2003 Dec 01
8
VoiceGlo
Hi,
VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll
Take a loock on http://www.voiceglo.com/
The softphone is IAX :)
Best regards,
Chris HARIGA
Techselesta Inc.
http://www.techselesta.com/
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2003 Oct 22
6
Running Asterisk and NAT on the same box?
Has anyone tried installing * on a box with two eth interfaces which is
acting as a NAT box? I have only one IP at this point and I would like
to get * working without all of the NAT issues. My idea is to run * on
my gateway (which is also running the firewall and masquerade services).
All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the
NAT screen, and will connect to the *
2006 Nov 06
7
several behind NAT
I've got my asterisk server in the DMZ of my local LAN - I've used my
Budgetone and GXP2000's from the Internet- on direct IP connections
with no problems. However, I'm about to deploy about 5 phones
(either budgetone or GXP2000's) all on a LAN behind a NAT- on a
different network than the Asterisk server. Should I look into using
STUN servers? Will this setup be a
2003 Oct 29
3
Am I missing somthing?
Should the following setup work?
SIP UA---NAT---Internet---NAT---SIP UA
If both UA's support STUN and report the external IP address in the SIP
packet..
I am trying to get away from using canreinvite=no so that traffic can go
directly between the UA's and not via the central server but I can't
seem to get it to work..
Has anyone set this up and can give me some pointers??
2008 Oct 16
2
SIP: difference between Grandstream and Cisco when behind NAT
I have used Grandstream phones for years, and have just started testing
a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling
and don't know whether it's just a limitation or something I haven't
done correctly.
The Asterisk server is directly on the Internet with a public IP.
The phones are on a private LAN with a NAT router to the Internet.
The sip.conf entries for
2003 Oct 12
4
No sound with SIP Phones on the Internet
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2005 Feb 23
2
multiple sip phones behind firewall
Hello List,
Can you please point me to the right resources on making multiple sip
phones behind a firewall w/ private address work with asterisk w/c is on
a public network.
I have seen STUN on the grandstream and Xtunnels on X-lite. What is most
deployed by members here with similar setups?
Thanks.
--
Cheers,
Paul P. Pongco
2005 Mar 03
2
Asterisk + SIP + NAT - seriously, what's the secret?
I'm at my wit's end!
I've spent 2 days now trying to get what I thought was a very simply SIP
+ NAT arrangement working. I've trawled the web and picked brains, but
nothing anyone suggests work.
My setup is very simple. I have a * server in a datacentre, with a
public IP address. There is no firewall in place, it's completely open
(at least, as far as I'm concerned). I
2004 Jun 17
3
Cheap (US$120 or less) SIP Phones
These are the three cheap SIP phones that I've used.
Grandstream BT10x $65/street
Number only LCD
Zultys ZIP 2 $100/retail
No LCD
Uniden UIP 200 $120/retail
PoE, built-in switch
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
2004 Jan 14
3
grandstream asterisk configuration
hi,
I have the following configuration:
Grandstream --> NAT (Netgear RP614)-->Internet-->Asterisk(public IP)
i can register fine and call ringing is working as good. The problem is =
i cant hear audio both ways and i get this error:
WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
my sip.conf file is as follows:
2003 Dec 05
3
GrandStream Budgetone Phone & DHCP & General Observations
Symptom: Phone after about 15mins will stop functioning
Problem: DHCP lease renewed but default route dropped
Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released
It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia
2005 Jan 27
3
SIP + NAT = horrible mess
Hi Guys,
After days of fiddling, I can't really get my SIP device to work
communicate with Asterisk behind NAT. Sometimes the STUN server is
flaky, sometimes the device isn't reachable if the connection is dropped
and then put back on, sometimes it registers OK, sometimes it doesn't, etc.
I've come to the same conclusion as the wiki: it's probably better to
avoid this
2004 Jan 10
5
Asterisk + BudgeTone (behind NAT)
I'm using Asterisk on a open server (no firewall or NAT) and trying to
communicate with a Grandstream BudgeTone 102 SIP phone which is behind
NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS
about a week ago. My problem is that I'm only getting half-duplex
communication -- I can hear voice from the Asterisk server but the server
does not understand any voice from
2003 Oct 30
1
SIP NAT
Should it work to have a multi-homed asterisk server with grandstream
phones on the internal network and another grandstream phone on the
internet and be able to call between them? I set the bindaddr to the
external IP and pointed the internal and external grandstream phones to
that address. The signalling works fine to call between phones, but when
you pick up the ringing phone you get a
2006 May 03
2
SIP Phones behind dynamic IPs
Greetings list,
I'm coming across an issue with some of the GXP-2000 phones we have out in
the wild at clients' employees' homes. In most cases they're behind consumer
ADSL NAT routers on a dynamic IP from their ISP.
In a nutshell, the phone is unable to be called unless it's restarted first,
after which it's fine for a good few hours, then it stops working until
2004 Dec 14
1
Asterisk to sip client behind Firewall/NAT - cancall but cannot receive calls ?
Check your FW-1 tracker and see if any sip packets are dropped during
call initiation.
I had this problem and it went away when I upgraded the BT's firmware to
the latest (16).
Beware, though, that people on the list claim that this firmware breaks
functionality of the message button and autoanswer.
I haven't checked this yet, cause I can't afford to go back a version.
I prefer a
2006 Jan 23
2
Polycom phones and dynamic IP for NAT
I know the Polycoms work with NAT, but you have to specify the public
IP.
Is there anyway for it to discover the external IP automatically?
I like the phones (been playing with a 301) but for some of our clients
who have a dynamic IP (and no hope of getting a static ie cable or
residential DSL) I'd be afraid to use them since you have to specify the
IP.
What about the Cisco phones?