ricky gutierrez xserverli...@gmail.com schrieb:
compilation problems with the module srtp , check the module
module show like srtp
Now available on OpenWRT... :(
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
--
2015-06-05 14:29 GMT-06:00 Luca Bertoncello lucab...@lucabert.de:
I think it is a problem on Asterisk for OpenWRT... :(
Regards
Luca Bertoncello
(lucab...@lucabert.de)
compilation problems with the module srtp , check the module
module show like srtp
--
rickygm
ricky gutierrez xserverli...@gmail.com schrieb:
Hi lucas , dou you try this:
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
Tested right now.
Same problem...
I think it is a problem on Asterisk for OpenWRT... :(
Regards
Luca Bertoncello
(lucab...@lucabert.de)
--
2015-06-05 12:21 GMT-06:00 Luca Bertoncello lucab...@lucabert.de:
Hi list!
I'm trying to configure my Asterisk to accept SIP-TLS connections, too.
I followed this HowTo:
http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/
But as soon I try to connect to my Asterisk
Does each box show up in the others SIP SHOW PEERS?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject:
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Looks like a connectivity issue, doesn't it?
IP of box2, 172.17.145.145, doesn't show up even once in the SIP dialogues.
What happens on box2 (asterisk -vvvr and tcpdump port 5060) in the
moment that you place a call through box1 to box2?
Also what's strange is that you are trying to call
@lists.digium.com
Sent: Tuesday, December 11, 2012 3:53 AM
Subject: Re: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention
Hi, thanks a lot by the answers. But without the application Answer() the
problem remains.
Realized over a battery of tests and refined the problem. Follows:
A = External link that came with my Voip number.
B = Operator.
C = The extent to which A want to speak.
A called my number and B answer.
Hi!
I'm experiencing a problem with my SIP channel's. When I have an
external connection for one of my SIP carrier's, I can listen to the
client and the client listens to me normally. The problem is when I will
transfer this connection, the call is mute for the extension I have
Hi!
client listens to me normally. The problem is when I will transfer this
connection, the call is mute for the extension I have transfered. Only the
client hears normally.
I *think* there is/was an entry in the bug tracker on this. You might
want to search https://issues.asterisk.org (also
This is the exit of core show version:
Asterisk 1.6.0.28 built by root @ AST on a i686 running Linux on 2010-06-28
12:21:24 UTC
Obg,
Rodrigo Lang.
2010/7/20 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Hi!
client listens to me normally. The problem is when I will transfer
Rodrigo Lang schrieb:
Good afternoon list.
I'm experiencing a problem with my SIP channel's. When I have an
external connection for one of my SIP carrier's, I can listen to the
client and the client listens to me normally. The problem is when I
will transfer this connection, the call is
Hach Segal a écrit :
Hello All,
I've been having some intermittent trouble with an Asterisk 1.2.10
Before anything else did you tried an updated asterisk 1.2
The last one is 1.2.28 or something like that, and there has been
a lot of security patches, and fixes since your version.
Did you
Diego Andrés Asenjo González wrote:
Hi!
I'm registering an asterisk server in a Sysmaster with a SIP account.
The registration succeeds and I can establish a call that come from the
Sysmaster.
After around 80 seconds the Sysmaster sends a BYE SIP message and the
call hang up. This does not
Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas wrote:
Hi all,
i've a problem in my Asterisk system. We have around 30 SIP phones
connected to an asterisk system, and sometimes some SIP channel
(associated to an extension) gets busy all the time, even when that
extension isn't in use.
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my
Are you doing port forwarding on your firewall?
Just make sure your asterisk port is open...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Piqueras
Sent: 30 May 2005 10:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problem
Has you redirected all the RTP ports? You must redirect the SIP and the
RTP streams. Take a look to the rtp.conf file of your asterisk
installation to configure the RTP ports that you want to use.
Best regards.
Rpr
Alex Piqueras escribió:
Hi, I have my asterisk server inside a NAT.
When i
Luis,
I tried to simulate your situation using a sip agent (Xten X-Pro) and
having it register to Asterisk with two user ids simultaneously all on
the same LAN.
I cannot replicate your problem. Both id's registered immediately.
Can you test this in your environment replacing the gateway with
Hi!
On my SIP softphone, when I stop speaking the audio stops. So if im not
talking I cant hear the other person.
FAQ!
X-Lite: Menu -- Advanced settings -- Audio -- Silence
Set Transmit Silence to YES
P.
___
Asterisk-Users mailing list
[EMAIL
Hi!
X-Lite: Menu -- Advanced settings -- Audio -- Silence
set keep transmitting after silence to 1 or something like that
Cf
- Original Message -
From: Philipp von Klitzing [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 21, 2004 11:24 AM
Subject: Re: [Asterisk-Users
Ok that fixed it. But why all of a sudden did it start doing this after
I updated? Anyidea? It had been working fine for a few months.
Kyle
Philipp von Klitzing wrote:
Hi!
On my SIP softphone, when I stop speaking the audio stops. So if im not
talking I cant hear the other person.
FAQ!
On Thu, 2004-05-20 at 18:47, Kyle Hagan wrote:
On my SIP softphone, when I stop speaking the audio stops. So if im not
talking I cant hear the other person.
http://lists.digium.com/pipermail/asterisk-users/2003-November/027732.html
Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration from
'sip:[EMAIL PROTECTED]' failed for '192.168.0.6'
Are you sure your phone isn't registering? These errors aren't related to your
grandstream. Do a sip show peers at the Asterisk CLI and see if it shows your phone
G'day Marc,
On Wed, 25 Feb 2004, Marc Fargas wrote:
Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info
(SP5002/S) and traed to register to asterisk, It seems to autentícate but
sniffing the net it shows a 407 proxy authen required error message and I
cannot make any
Vic Cross wrote:
G'day Marc,
On Wed, 25 Feb 2004, Marc Fargas wrote:
Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info
(SP5002/S) and traed to register to asterisk, It seems to autentcate but
sniffing the net it shows a 407 proxy authen required error message and I
On this cuts note that the gateway has username 'Republica', you could see
some reference to Republica2 which corresponds to a second line on the
gateway that I have disabled.
Thanks for your help!
That's SIP debug when dialling '9' (9 would do Goto(s,1))
===
*CLI
*CLI
11
Seems like republica registers ok, but not republica2. Republica2 failes to authenticate.
You have a normal registration sequense here:
-Client sends a REGISTER without authentication
-Server sends trying...
-Server sends 407 Proxy auth (should be WWW auth) with challenge
-Clients ACK
-Client
I have commented de Republica2 in sip.conf 'cause if I uncomment it neither
Republica and Republica2 register (maybe because they're on the same
gateway?)
Well, inspite it register well when I try tocall any extension It plays
'busy' tone immediately after Asterisk takes the calls I thought it
On Fri, 2003-11-28 at 15:26, Ernst Lehmann wrote:
Hi All,
I am a newbie to asterisk, and here is my first problem, where I do not
know any further.
I have to grandstream BT100 connected to asterisk. Working fine, for
calling to each other, and to call via a IAX-Link to the outside.
If
On Wed, 2003-10-15 at 15:22, Alex Lopez wrote:
There was a tread that I googled for and could not find about Asterisk
being open to SIP DOS Attacks. I have a customer whose machine was
hammered last light by traffic on its SIP port causing the OS to use
up its resources. Namely number of
It looks like you are registering fine. If
you dial 12321 from another phone, does it not ring?
This is the transaction as I see it in the log that
you attached:
Phone: REGISTER
Asterisk: Proxy Authentication Required (Send me
your credentials)
Phone: REGISTER with CREDENTIALS
Asterisk:
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