Displaying 20 results from an estimated 52 matches for "autocreatepeers".
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autocreatepeer
2004 Aug 21
0
autocreatepeer and sip peer options
Hi all,
quick question...i am using autocreatepeer to get asterisk to work with SER
without having to specify each UA in sip.conf and in ser separately.
2 questions:
1. obviously this is not very secure. assuming i block incoming requests on
the port asterisk is running SIP on (excluding requests from the SER, of
course) does this adequately protect the server from unauthorized users or
is there
2004 Sep 08
0
re: asterisk, SER and autocreatepeer
Hi all,
quick question...i am using autocreatepeer to get asterisk to work with SER
without having to specify each UA in sip.conf and in ser separately.
2 questions:
1. obviously this is not very secure because anyone can bypass the SER
and register themselves as a peer with the asterisk. assuming i block
incoming requests on the port asterisk is running SIP on (excluding
requests from the SER, of
2004 May 14
4
sip authentication
Good day all
How do I get my asterisk and sip to use the password.I'm using x-lite.If
I use just the username and no password it still logs on?
Here is my sip.conf entry?
[101]
type=friend
callerid="Test User" <101>
context = test_1 ; Default context for incoming calls
username=101
secret=123456
host=dynamic
dtmfmode=inband ; Choices are inband, rfc2833, or info
2010 Jan 12
2
SIP Security
Hey guys,
I've been running asterisk on my server for some time now (currently
running Asterisk 1.6.2.0). I am having security issues with my SIP
accounts. Unauthorized people have been able to access the server (bots)
and they have been able to make calls (in today's case to Cuba).
Here's a copy (slightly modified) of my sip.conf:
[general]
context=default ; Default
2005 Mar 08
4
force SIP authentication
Hello,
is it possible with Asterisk to force SIP authentication? Right now, it
seesm that just any SIP client can at least connect to my PBX, which I
don't want. I want users to authenticate with username and password and
otherwise deny them access.
Thanks
Florian
2005 Sep 05
2
Asterisk won't listen on another port
Hello,
Hope somebody can help me - Asterisk is behaving very oddly and I'm
totally stumped! I have SER and Asterisk running on the same box. I want
SER to listen on port 5060 (it is) and Asterisk to listen on port 5062.
I have configured my phones to register with x.x.x.x:5060 (SER) and
Asterisk will purely act as a voicemail server at the moment. However I
cannot get Asterisk to listen on a
2004 Sep 30
1
easy way of add 100 extensions
Hi,
Is there a "easy" way of adding 100 extensions?
I mean, I don't want to create 100 section in the sip.conf.
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2006 Jan 30
0
re: help with redirect from SER
hello all,
i have a problem, and i'm tearing my hair out...any assistance is
appreciated. I am trying to redirect from SER to Asterisk, both on the same
machine. In 1.09 I didnt need to set up a peer for SER, just
autocreatepeer=yes, and rewritehostport from SER as below, and asterisk
accepted the requests without a problem. When I updated to 1.23 requests
from SER to asterisk die quietly, no
2004 Jun 23
1
Asterisk user/host registration
Hi Folks,
I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server.
When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below.
*CLI> sip show peers
Name/username Host
2006 Feb 05
2
re: questions about sip requests to asterisk 1.2
hi all,
I keep asking the question and getting no replies, so i'll keep asking :-)
In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request
from SER, specifically
rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk
picks up the request and matches it to the dialplan, i.e. if in ser i was
sending to 151@myServer, it will make it
2007 Mar 28
3
Multi-line phones - Asterisk uses wrong callerid
I have some phones (and an ATA) that are shared between two users who
each have separate voicemail but they are not behaving as desired nor
expected.
Incoming calls show up on the correct lines.
Calls originating from the device are seen, at the terminating device,
as coming from the account listed last in sip.conf, regardless of the
line selected.
This creates three main issues I would like
2006 Nov 07
3
Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
Hi All,
I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:
asterisk sip >< sip TNT pri >< pri asterisk
The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other way. The call from asterisk to pri to tnt
is good, the TNT is passing SIP invite to the
2004 Sep 30
4
Caller ID Info from Cisco router to Asterisk
Dear Asterisk Gurus:
Our county is finally ready to begin implementing IP telephony. We intend
to use a Cisco router as our PSTN gateway and Asterisk as our soft switch.
The plan is to use SIP between the Cisco router and Asterisk. We will have
a single PRI T1 connected to the Cisco router for PSTN access. My question
is this:
Are Cisco routers able to pass caller ID information (from PRI
2010 Nov 06
2
One way voice with Asterisk
Let me explain:
When I dial into Asterisk ( I have a SIP trunk - which I need to make
sure is not faulty), I only get one-way voice communication.
The calling party, from the SIP trunk hears nothing - the extension
rings on the Asterisk server (you can see it in the CLI and hear it at
the computer), and the softphone rings
However, when you answer the SIP softphone , you can only hear the
2003 Nov 22
3
SIP channel improvements
I just discovered that the SIP channel has undergone some major improvements.
I'm now able to dial any SIP URL with dial, couldn't get it to work earlier,
all domains had to be defined in SIP.conf.
This, in addition to the SIPDOMAIN variable, makes the SIP channel even more
useful.
Thank you, Mark, for your additions!
Now, ENUM/E.164 will propably work even better. I'll give it a
2004 Jan 12
1
Cisco FXO as PSTN gateway
I have been compiling information on this configuration onto the Wiki:
http://voip-info.org/wiki-Asterisk+cisco+FXO
I can call out to the PSTN just fine, but inbound calls all appear in my
[bogon-calls] context.
Can anyone help me locate why?
(Config files are on the Wiki)
I have done a packet sniff & decoded using Ethereal-0.10.0, but this
doesn't tell me a great deal - I just see
2004 Jan 15
3
Cisco FXO as PSTN gateway (updated request for assistance)
I have been compiling information on this configuration onto the Wiki:
http://voip-info.org/wiki-Asterisk+cisco+FXO
I can call out to the PSTN just fine, but inbound calls all appear in my
default [bogon-calls] context, not in [pstn-incoming]
Can anyone help me locate why?
(Config files are on the Wiki)
I have done a packet sniff & decoded using Ethereal-0.10.0, but this
doesn't tell
2004 Aug 19
0
SIP reinvite code negotiation
Hi,
We're routing SIP calls through Asterisk and we want to
be able to reinvite calls without Asterisk performing
codec conversion.
We've performed the following test:
Asterisk has license for G.729 installed
sip.conf
[general]
context=default
autocreatepeer=yes
disallow=all
allow=alaw
allow=g729
canreinvite=yes
nat=no
We have configured two endpoints:
EP1, preferred codec order
2004 Dec 06
0
Phone Giptel G100 with Asterisk?
Hi there,
so far I've been unable to get a Giptel G100 phone to operate with
Asterisk - I tried both SIP and MGCP. Since I've been playing with quite
a number of different SIP (and also MGCP) user devices I am starting to
think that it might be the phone's fault and not mine... ;->
Still: Anyone out there has this working with Asterisk?
The symptoms are that the G100 won't
2004 Dec 09
0
Ser + Asterisk & DMZ
Hi all
I am in this strange situation: we had ser configured to relay calls to
numbers to asterisk extensions and all used to work nicely, with both ser and
asterisk running on the same machine with public ip (ser on port 5060 and *
on 5061). We had to move temporarily our server to another provider which put
our server on a dmz, so that now we have our server with private ip but
reachable from