Displaying 20 results from an estimated 52 matches for "autocreatep".
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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 pro...
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 i...
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
...uthorized 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 context for incoming calls
videosupport=yes
rtcachefriends=yes
autocreatepeer=no
t38pt_udptl=yes
allowoverlap=no
udpbindaddr=0.0.0.0
srvlookup=yes
;pedantic=yes
disallow=all
allow=alaw
allow=ulaw
allow=speex
[1001]
type=friend
username=1001
secret=blah
subscribecontext=default
regexten=1001
callerid="blah" <XXXXXXXXXX>...
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
....0.0.0 :2000 LISTEN
asterisk
.
.
.
0.0.0.0 :2727
asterisk
0.0.0.0:4520
asterisk
0.0.00:5060
asterisk
x.x.x.x:5060
ser
127.0.0.1:5060
ser
My config is like follows
;sip.conf
[general]
context=default
port=5062
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
autocreatepeer=yes
[2092]
type=friend
username=2092
canreinvite=no
context=default
mailbox=2092
host=dynamic
nat=no dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
;extensions.conf
;leave voice messages
exten => 2092, 1, Voicemail(u2092)
exten => 2092, 2, Hangup
;play voice messages
exten =>...
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 matter how verbose my asterisk is. It's
as if the requests dont exist at all.
My setup is as follows: asterisk and SER on t...
2004 Jun 23
1
Asterisk user/host registration
...Mask Port Status
2001/2001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
2000/2000 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
I am pasting sip.conf & extension.conf
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = INVALID
;Autocreatepeer= yes
[2000]
type=friend
username=2000
secret=2000
host=dynamic
context=from-sip
mailbox=100
canreinvite=no
qualify=300
nat=no
[2001]
type=friend
username=2001
secret=2001
host=dynamic
context=from-sip
mailbox=100
canreinvite=no
qualify=300
nat=no
extensions.conf
[globals]
CONSOLE=Console/d...
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 151@myIP:5070, and asterisk will
matc...
2007 Mar 28
3
Multi-line phones - Asterisk uses wrong callerid
...Phone 202 shows an INVITE from 103 to 202.
103 happens to be the last listed in sip.conf and the first listed in
'sip show peers' (I have confirmed that this is dependent on the order
in the conf file, not numeric order)
sip.conf :-
[general]
port = 5060
bindaddr = 0.0.0.0
pedantic = no
autocreatepeer = no
context = sip
registertimeout=20
localnet = 10.10.10.0/255.255.255.0
srvlookup = yes
tos=0xb8
rtptimeout=300
rtpholdtimeout=1800
maxexpirey=3600
defaultexpirey=1200
[sip-101]
; Aastra 480i phones for general office
type=peer
insecure=very
disallow=all
allow=ulaw
allow=alaw
host=dynamic
dtm...
2006 Nov 07
3
Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
...d 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 SIP Asterisk server. I
have tried many variations of using sip options insecure,
autocreatepeer, permit/deny, host, user, etc.... but can't seem to get
asterisk to accept an unauthenticated call from the TNT using SIP. I
keep getting SIP/2.0 407 Proxy Authentication Required. I know others
have done this, but with older Asterisk versions, I'm wondering what
versions of Asterisk a...
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
...le clueless!
Thanks for all of your help.
Asterisk 1.4.31 built by root @ some_server.foo.net on a x86_64 running
Linux on 2010-06-10 14:32:34 UTC
Sip Settings:
Global Settings:
----------------
SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
A...
2003 Nov 22
3
SIP channel improvements
...d 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 try.
Now, to be the documentation-pain-in-the-*** I would like to get an explanation
of the autocreatepeer SIP.conf setting and functionality?
It's not in sip.conf.sample yet.
/Olle
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)
...esn't tell me a great deal - I just see the rejection message:
y.y.y.y x.x.x.x INVITE sip:1234@x.x.x.x:5060
x.x.x.x y.y.y.y Status: 100 Trying
x.x.x.x y.y.y.y Status: 503 Service Unavailable
y.y.y.y x.x.x.x Request: ACK sip:1234@x.x.x.x:5060
(resent as retested with 0.7.1 & the addition of autocreatepeer=yes)
Thanks a lot,
Fran.
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 aLaw, G.729
EP2, preferred codec order G.729
EP1 places call to EP2, we see two call legs:
EP1 to * is aLaw
* to EP2 is G.729
Is there a sip.conf parameter to disable cod...
2004 Dec 06
0
Phone Giptel G100 with Asterisk?
...rting 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 register, i.e. it doesn't respond to
the SIP nonce that Asterisk sends out. A static IP won't help, and if I
use autocreatepeer=yes then ringing the phone works, but upon picking up
the handset the phone keeps ringing...
However, since providers like Nikotel are selling the G100 there must be
a way to get it working; my current guess is that with SER or some other
SIP proxy it might come to life. Comments?
Cheers &a...
2004 Dec 09
0
Ser + Asterisk & DMZ
...; Port to bind to
;bindaddr = 10.0.0.229 ; Address to bind SIP channel to
bindaddr = 0.0.0.0
context = 82.184.xx.xx ; Default context for incoming calls
srvlookup = no ; Enable DNS SRV lookups on outbound calls
;;;;;;; tried with or without following lines, still mute :-(
autocreatepeer=yes
externip=82.184.xx.xx
register => asterisk:xxxxx@10.0.0.229/100 ;asterisk actually registers on
ser! realm=82.184.xx.xx
;;;;;;; tried also with public ip host, nat=no, canreinvite=yes, type=peer
[asterisk]
type=friend
secret=xxxxx
username=asterisk
host=10.0.0.229
nat=yes
canreinvite=no...