Displaying 9 results from an estimated 9 matches for "dissallow".
Did you mean:
disallow
2003 Nov 11
2
sip: 401 unauthorized with xlite
...isk, but until now i didn't get sucess. When i start the asterisk in debug mode, i see the message: sip/2.0 401 unauthorized. I know that this problem with authentication. I put in my sip.conf as below.
[2203]
type=friend
username=2203
auth=md5
secret=1234
reinvite=no
canreinvite=no
dissallow=all
allow=gsm
context= sip
host= 192.168.10.149 -> my machine that have xlite
extension.conf
[sip]
exten => 2203,1,Dial(${Phone1})
I have read and read many message in list but i could found anyone that explain in details how to setup this correct. My sip.conf i got from a example in...
2005 Aug 08
6
IAX TO IAX call between two registered servers
...ead: Call rejected by
69.xxx.xxx.xxx: No authority found
and get this message on away
Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read: Rejected connect
attempt from 165.xxx.xxx.xxx
home iax.conf
[away]
type=peer
username=away
auth=plaintext
secret=xxxxxxxxx
host=dynamic
context=pap2
dissallow=all
allow=ulaw
[away-in]
type=peer
auth=plaintext
secret=xxxxxxxxx
host=dynamic
context=pap2
dissallow=all
allow=ulaw
[away-out]
type=peer
secret=xxxxxxxxx
username=away
host=dynamic
disallow=all
allow=ulaw
away iax.conf
[home]
type=peer
user=home
secret=xxxxxxxxx
host=dynamic
context=default...
2005 Mar 18
1
Broadvoice hangs-up / disconnects after about 30 deconds
...main=sip.broadvoice.com
fromuser=425XXXXXXX
insecure=very
;context=from-broadvoice
context=from-pstn
dtmfmode=inband
canreinvite=no
qualify=yes
user=phone
[200]
type=friend
secret=010101
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
dtmfmode=inband
callerid="Fred F"<200>
dissallow=all
Extensions.conf
[default]
exten => 1000,1,Dial,Zap/1|20
exten => 1000,2,Voicemail,u1000
exten => 1000,3,Hangup
exten => 1000,102,Voicemail,b1000
exten => 1000,103,Hangup
exten => 2000,1,Dial,Zap/2|20
exten => 2000,2,Voicemail,u2000
exten => 2000,3,Hangup
exten =>...
2004 Jan 19
3
Security suggestion concering SSH and port forwarding.
...ere comes my suggestion:
I recently stumbled upon the scponly shell which in it's chroot:ed form is
an ideal solution when you want to share some files with people you trust
more or less.
The problem is, if you use the scponlyc as shell, port forwarding is still
allowed. This can of course be dissallowed in sshd_config, but not only
for certian users and/or groups.
Example scenario:
You're on a privat network, behind a firewall. You're letting port 22 in
to your linux machine. A few trusted people have normal accounts on this
machine allowing them to use -L to forward ports to other mac...
2004 Dec 16
0
SPA-3000 - Stop Message Waiting Indication
Hi,
I have my Sipura SPA-3000 setup with Asterisk as follows:
[spa3k_line1]
type=friend
context=home
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
dissallow=all
allow=ulaw
When an incoming call comes in, I have a Zap interface in Asterisk which
just does a Wait,15 then answers with voicemail.
The SPA-3000 detects the PSTN call and makes Line 1 ring - so I can
answer the phone if i'm around - if not, Asterisk gets to the end of the
15 seconds an...
2005 Mar 12
1
X-Lite and * SIP Problem
...st: Registration
from 'richard <sip:richard@sip>' failed for '192.168.0.100'"
My sip.conf is as follows:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[200]
type=friend
username=richard
secret=password
host=dynamic
reinvite=no
canreinvite=no
dissallow=all
context=sip
allow=gsm
And my X-lite Default SIP Proxy config is as follows:
Enabled: Yes
Display name: richard
Username: richard
Authorisation User: richard
Password: password
Domain/Realm: 192.168.0.102 (my asterisk server's IP)
Sip Proxy: 192.168.0.102
rest left as default
Can anyone t...
2007 Jan 26
0
from attribute in actionmailer not working w/ gmail
I assign an email to the from attribute in my actionmailer, but I think
because I use the TLS plugin to get actionmailer to work with gmail, it
dissallows me to use change the from address. Instead it uses my gmail
account. Has anyone experienced this before?
--
Posted via http://www.ruby-forum.com/.
--~--~---------~--~----~------------~-------~--~----~
You received this message because you are subscribed to the Google Groups "Ruby on Rails:...
2005 Mar 18
2
No sound when calling in from pstn
I am just starting out with * so bear with me please.
I have tdm400p with 4 fxo modules on it. When I call into the asterisk
box from my mobile, I can see the asterisk console picks the call up
and routes it to my computer with x-lite. There was no sound coming
from either - just silence. I then decided to route it directly to
voice mail to see if that would narrow the problem down, but it
2004 Jul 13
1
codec issues between linphone and *
...pirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
dissallow=all ; Disallow all codecs
;allow=all
;allow=speex
;allow=ulaw
;allow=gsm ; Allow codecs in order of preference
:allow=ilbc
*****************.linphonec config file ****************************
[net]
if_name=rhine
con_type=1
use_nat=0
[sip]
usern...