Displaying 9 results from an estimated 9 matches for "localextensions".
2005 Jan 08
0
Any experience with Linksys WRT54GP2 as localextensions to Asterisk ?
...users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Robert Rozman
> Sent: Saturday, January 08, 2005 7:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Any experience with Linksys WRT54GP2 as
> localextensions to Asterisk ?
>
> Hi,
>
> I'd just like to confirm compatibility of Linksys router WRT54GP2 as
> local
> extensions to Asterisk.
>
> Can it register to local Asterisk behing him ? How stable/good is
analog
> interface ?
>
> Any experience would be more tha...
2005 Mar 10
0
Re: Polycom phones do not talk to each other
...efault
VAR: agi_extension: 51
VAR: agi_priority: 1
VAR: agi_enhanced: 0.0
VAR: agi_accountcode:
Detected protocol 'sip' ... 200 result=1
Detected caller '50' ... 200 result=1
Set limit - 24 200 result=1
Limit not exceeded (3 < 24) for localextensions 200 result=1
Set limit - 5 200 result=1
Limit not exceeded (3 < 5) for 50_out 200 result=1
Detecting destination for '51' ... 200 result=1
Found Destination localextensions (range 51 - 51) 200 result=1
Setting destination 'localextensions' ... 200 r...
2009 Jan 21
0
About Asterisk 1.6.0.1
...; Port to bind to (SIP is 5060)
bindaddr = 192.168.1.243 ; x = Asterisk server IP address
disallow=all
;allow = ulaw ; Allow all codecs
;allow = alaw
context = bogon-calls ; Send SIP callers that we don't know about here
canreinvite=no
directrtpsetup=yes
nat=no
;subscribecontext= localextensions ;default
allowsubscribe=yes ; Disable support for subscriptions.
(Default is yes)
[App]
type=friend
username=App
;regexten=1234 ; When they register, create extension
1234
;secret=password
host=dynamic
context=from-sip
mailbox=App
disallow=all
allow =...
2009 Jan 22
0
Query About Asterisk 1.6.0.1 Dialog Event Package.
...60 ; Port to bind to (SIP is 5060)
bindaddr = 192.168.1.243 ; x = Asterisk server IP address
disallow=all
;allow = ulaw ; Allow all codecs
;allow = alaw
context = from-sip ; Send SIP callers that we don't know about here
canreinvite=no
directrtpsetup=yes
nat=no
;subscribecontext= localextensions ;default
allowsubscribe=yes ; Disable support for subscriptions.
(Default is yes)
[App]
type=friend
username=App
;regexten=1234 ; When they register, create extension
1234
;secret=password
host=dynamic
context=from-sip
mailbox=App
disallow=all
allow =...
2005 Mar 11
1
Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI.
...fault
VAR: agi_extension: 56
VAR: agi_priority: 1
VAR: agi_enhanced: 0.0
VAR: agi_accountcode:
Detected protocol 'iax2' ... 200 result=1
Detected caller '101' ... 200 result=1
Set limit - 24 200 result=1
Limit not exceeded (1 < 24) for localextensions 200 result=1
Set limit - 2 200 result=1
Limit not exceeded (1 < 2) for 101_out 200 result=1
Detecting destination for '56' ... 200 result=1
Found Destination localextensions (range 56 - 56) 200 result=1
Setting destination 'localextensions' ... 200...
2003 Jun 15
2
a few questions about sip implementation
I'm looking at RFC 3261, I think the latest SIP standard and have a few
questions about the * sip implementation:
1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue
a provisional response to non-INVITE requests.
2003 Jun 17
3
sip.conf
HI,
can somebody tell me how and where must I put the SIP register line? I
think is in [general] section of the sip.conf and that I have to put:
register => user:password@host:port/localextension
but, user and password of the SIP gateway? Because I'm trying this and
doesn't work...
thanks a lot in advanced
michelle
-----
Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam
2003 Jun 15
1
SIP REGISTER behavior change: specific domains possible in REGISTER
Mark has fixed the REGISTER issues to be more RFC compliant. I've
created a new thread so that those of you who got bored with the old
thread might read this new one. The feature that has just been added
was added a while ago, but now it actually seems to _work_. :-)
If you have a SIP server to which you are trying to REGISTER, and
they demand valid domain (the part after the
2005 Mar 03
3
Problems dialing out - possible settings changes
First, I apologize if this info has been covered, I tried doing a search
on Google and the wiki and found nothing, but I could be searching for
the wrong info.
Second, I am new to asterisk and linux. My knowledge is Asterisk is
much better than linux, so excuse the ignorance in some areas.
Configuration:
Novell Linux Desktop
Asterisk 1.0.6
Intel CPU/MB
TDM12P digium card (I think that is it -