Displaying 20 results from an estimated 8000 matches similar to: "SIP & IAX behind NAT"
2004 Dec 22
0
Ticket: 12775 Multiple IAX client behind a NAT
Hello!
I have a number of IAX clients behind a NAT (on the same LAN) and
asterisk server on the Internet. And that clients doesn't speak directly
to each other, traffic goes through the asterisk server.
What should I configure to make IAX clients on the same LAN to speak
directly, please?
notraster=no is set in iax.conf
The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1
2006 Oct 11
10
GPL Softphones
Hi,
I'm searching for GPLed softphones. I found WengoPhone but actually not
available for Asterisk PBX, only for Wengo network. I found Kiax but only
for IAX protocol.
Did you know a good GPLed softphones which works on Windows ?
Thanks
Greg
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2010 Jan 28
1
iax softphones - not reconnecting
Hi together,
I try to find a softphone (freeware) solution for Windows 32, that works without problems ...
Right now I use iaxcomm wich was best, of the ones I tried.
But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor.
But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got
2005 Aug 04
4
Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone
Hi!
Problem:
I can't hear what the people at Location B i saying, they hear me but I do
not hear them. They can call, I can call. Just no sound.
My current setup is:
Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet
<-> Firewall/Nat <-> Softphone/hardphone(Location B)
I am having problems with sound, I have opened the
2004 May 20
2
Softphone lag
Hi,
IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 second.
The two phones were on 2 different pc's near me. When I speak on one, i hear it on the other after about 1 second.
I tried using iaxComm, Xten Xlite, etc. Same.
FYI: The codec used was GSM.
Using the fxo and fxs interfaces on the digium cards with POTS have no such issues.
Any clue where the
2007 Jul 24
10
What is the best softphone work with Asterisk
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
Regards
Bilal
____________________________________________________________________________________
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.
2008 Feb 05
3
[Softphones] ZoIPer vs. XLite?
Hello
I need to hook up someone's remote PC onto our Asterisk server over
the Net. There are firewalls on each side, so I figured it's time to
give IAX a try, and see if it's less of a pain to use than SIP. And
since IAX hardphones are pretty are, I guess I'll go softphone.
Apparently, the two most well-known IAX and SIP clients for Windows
are ZoIPer and X-Lite, respectively.
2005 Aug 07
3
Can call from iax extn but cannot call it - unable to cteate channel iax
Hello
I have created an iax exten in my iax.conf file:
[300]
type=friend
username=300
secret=***
context=default
host=dynamic
callerid="some name" <300>
auth=md5
Then in my extensions.conf I have:
exten => 300,1,Dial(IAX/${EXTEN},20)
exten => 300,2,Hangup
I can dial from iaxComm (a soft IAX client) and that works fine. But when I
try to dial 300 get:
WARNING[22077]:
2003 Nov 13
3
iax configuration
Hi,
I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has excellent voice quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now.
I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as
2005 Jan 12
1
linphone -> NAT -> * -> NAT -> firefly woes.
Hi folks
an issue I don't understand. I'm running * stable 1.0.3 on public
internet, with following iax.conf / sip.conf entries:
iax.conf
[100]
type=friend
username=Foo
context=default
auth=md5,plaintext,rsa
secret=secret
host=dynamic
callerid="Foo" <100>
qualify=no
sip.conf
[10]
type=friend
username=Bar
context=default
callerid=Bar <10>
2003 Sep 19
7
IAX vs SIP
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
overseas IP connection, and somehow SIP seemed to work better.
Peter
2004 Jun 01
2
iax codec problem
Hi everybody
i have a problem trying to connect an incomming phone call from pstn to my
(soft phone) iaxcomm, the phone rings but when i try to answer the call,
asterisk sends a message like this.
Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping
incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since
our native format has changed to ALAW
i'm working
2004 May 04
3
Linux IAX client
Folks,
It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there
another IAX2 client that is usable under Linux (Debian preferred)?
Thanks,
Tim
--
2003 Dec 02
5
Iax Client Library Issues? (DIAX, iaxComm, etc.)
Hi,
I seem to be having problems with IAX clients based on the iaxClient
library. I have been working on my own client (an augmentation to the
Call Manager I released last week) and it seems to regularly miss
incoming calls entirely. It also occasionally misses the drop signal
when the remote end drops a call.
Has anybody else seen this kind of behavior? I have tested with my
client, with
2003 Oct 30
2
Fwd: Re: SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
--- Peter Zeltins <peter@fintrading.com> wrote:
>
>
> Well, I happen to be one of those very specific cases... ;) and looks
> like
> will have experiment with it myself. Although I'd hate to re-invent
> the
> wheel.
>
> Peter
Checking e-mail this morning it looks like we have two independent
"fixes" that both do what has been suggested in this
2011 Sep 21
1
RTP stream when * and Xlite are both behind Nat devices.
Hi,
I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone
behind a different NAT network.
Asterisk -> Nat -> Internet -> Nat -> Softphone.
I can register my softphone to the asterisk box ok via SIP but the RTP
stream from the asterisk box is addressed to the private non-routeable
address of the softphone when I turn on rtp debuging.
How can I configure the rtp
2005 Feb 22
1
Astersik CVS HEAD + T1 e&m wink + IAX client doesnt detect call answered on Zap channel
Hello,
I've got very annoying behaviour from our asterisk PBX.
We have 12 channels T1 e&m wink start for TDM and using iax softphones
internally (iaxcomm, but tried firefly-thirdparty and discarded for
bad sound quality).
Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card.
In some cases when call is placed from softphone to TDM, system does
not detect call answered on Zap channel and
2004 Aug 23
4
Asterisk WITH Swyx... Any Idea?
Hi,
I'm a student and my thesis work consist in testing
Asterisk with Swyx(SwyxWare).
My approach is to declare asterisk as h323 gateway for
the Swyxserver using oh323 Plugin.
Is there any possibility to connect Asterisk with
Swyx? how?
the outgoing call must pass from Swyxit->to
Swyxserver-> Asterisk->to PTSN
Thanks
2005 Jun 09
3
Softphone for Linux desktops
Hi all,
We are successfuly running an Asterisk server with standard SIP hard
phones and it is working well. We are looking to deploy some soft
phones on our Linux desktops. There seems to be several floating
about. Anyone out there with some good/bad experiences with particular
Linux softphones. We only need g711 and prefer IAX but a SIP one will
do....
Thanks....
2005 Jan 14
1
iaxComm 0.99pre11 binaries posted to Sourceforge
iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
It is distributed as part of Steve Kann's iaxclient library.
I've just posted new Windows, Linux and Mac OSX binaries to sourceforge.
The Windows binary was compiled on WinXP.
The Linux binary was compiled on RedHat 9.
The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4
(Tiger) beta.