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 using SIP-TLS I get on
Asterisk-CLI:
== Problem setting up ssl
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
I am using voip with Vodafone as SIP peer for outbound telephony and i
have a huge problem establishing calls to other people. It works like in
1 of 5 tries. The peer is sending SIP 480 temporarily not available.
It took a while to identify this, because on the phone you just hear
busy tone.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
-users] Problem with SIP trunk I've set up between two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk
] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4
2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box
@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
I am having a strange problem with an external SIP phone. It can
register and receive calls but it cannot initiate any calls. A
softphone on the same network works without problems.
As far as I can notice the difference is that the hard phone is not
sending the proper contact
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
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 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
Hello All,
I've been having some intermittent trouble with an Asterisk 1.2.10
installation that is supporting roughly 50 SIP clients on a LAN, mostly
soft phones and about 10 snom VoIP phones. We have a custom soft phone
client which displays presence information for various extensions.
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
Hi,
maybe someone can give me a hint to solve the following issue. I want to
limit the calls to a specific SIP-destination. Disabling callwaiting at
the phones is not an option, because it should be configured via the *
database.
My solution uses GROUP_COUNT, which works fine most of the time.
Hi all
I have setup sips accounts to an asterisk server from a
provider, I know that there are using asterisk real time for sip users
definitions.
Sometimes in a ramdom basis I receive:
chan_sip.c:9596 handle_response_register:
Forbidden - wrong password
Hi all
I have setup sips accounts to an asterisk server from a
provider, I know that there are using asterisk real time for sip users
definitions.
Sometimes in a ramdom basis I receive:
chan_sip.c:9596 handle_response_register:
Forbidden - wrong password
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 occur to the hard/soft SIP phones
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
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 whenthat
extensionisn't in use.
We have a workaround for this, as we can't restart
asterisk
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.
Hi
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
Hi
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
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
I am not able to get softphone registered (active) with * .
new installation , new user
Able to get server started , and phone appears to register
gets the SIP reponse 481 message
Register SIP 4009 at 192.168.200.10 port 2199
expires 120
Unregistered SIP 4009
Register SIP 4009 at
i am configure ser:
if (method==INVITE) {
if (uri=~sip:[EMAIL PROTECTED]) {
rewritehostport(192.168.0.183:5080);
};
};
an asterisk:
sip.conf
; config Xlite
[1234]
;context=sip
context=from-ser
type=friend
auth=md5
username=1234
secret=chooseapassword
;fromdomain=sorcier.com.pe ; para
Hi
I have configured sip accounts and they work some times. when i make a call
to another SIP account it works right
but some times i get the following error
Jul 29 07:17:00 WARNING[802]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
102 (Critical
Hello
My name is
Will.
I have a problem
with SIP on ASTERISK
How many ways it has
to register and to work in sip.conf?
Thanks
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Dear all,
just upgraded to cvs head june 6th, using 1.0.7 sip.conf but can't accept
any calls from SIP proxy. Anyone encountered the same problem?
[general]
context=sip-in
recordhistory=yes ; Record SIP history by default
port=5070 ; UDP Port to bind to
Hi, I have my asterisk server inside a NAT.
When i connect a softphone SIP inside my net, all go well. Ok.
But when I try connect a SIP softphone client outside my NAT, I get:
NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex
sip:[EMAIL PROTECTED]' failed for '83.41.119.25'
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
Hi
I am trying to get 2 incoming SIP accounts working from 2 different
providers. One is sipgate.co.uk and the other is voipuser.org. If I load
the Register command seperate they will both register phone and incoming
works. If I try to load them both only sipgate registers. Anybody got
any
I guess I found a bug in the register logic in chan_sip
I'm trying of registering two extensions from a SIP gateway into Asterisk.
I have defined two user entries in sip.conf as follows:
[0191]
type = friend
auth=md5
username=0191
secret=planet
disallow=all
allow=ulaw
dtmfmode=inband
host =
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!
Having a weird problem after I updated the other day.
On my SIP softphone, when I stop speaking the audio stops. So if im not
talking I cant hear the other person.
Kyle
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
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 outgoing calls from that gateway.
Thats what I have:
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
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 I try to call the initial demo from the
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 open files. The discussion was around the fact
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
etxusa.com/
- Original Message -
From:
John
Foster
To: [EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 12:49
AM
Subject: [Asterisk-Users] Problem with
SIP authentication
Hi List,
After going through mailing list and manual of asterisk, I s
Hi List,
After going through mailing list and manual of asterisk, I still could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params
Ok I have the following on the Asterisk every minutes.
Got SIP response 481 Call Leg/Transaction Does Not Exist back from 2XX.2XX.133.1XX.
The user of this Sip phone is able to make calls and get calls. A Windows 2000 pro
using MS Messenger! I loaded it on my PC as well and it does the same
Hi all,
this message occurs if i was connected or not:
WARNING[213006]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
102 (Response)
If i was connected, the call will be disconnected after a few seconds.
What does it means ? I don't see
My configuration is comprised of two Snom 200 phones, two FXO cards
connected to two PSTN lines, and one SIP account at iConnect. Snom1 has
a VPN connection to the remote Asterisk server. Snom2 is using UPnP
behind a Linksys WRT54G router/firewall to connect to the same server.
All outgoing calls
I have a X100P and am calling out from a desktop
within the same network. I connect to * then dialout a local phone number
to my cell phone. It rings 2 times then hangs up.
I'mtesting Sipps as the
softphone.
* is saying "retries exceeded".
Has anyone had this problem? It's probably
with
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