Displaying 16 results from an estimated 16 matches for "match_auth_username".
2010 Jun 03
0
SIP: match_auth_username=yes doesn't seem to work
Hi,
I'm trying to get the match_auth_username=yes sip configuration working.
It's mentioned as an experimental new feature of 1.6.2.x. (I'm using 1.6.2.8)
The sip.conf example states:
; if available, match user entry using the
; 'username' field from the authentication line
; instead of the From: field.
But still I've bee...
2015 Apr 07
5
Asterisk Inbound calls, multiple SIP accounts, calledID
Hi Dmitriy and others and thanks for your help so far.
The option "match_auth_username=yes" seems to have had no effect. From my
reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying to present the receptionist with a nice display of
which line the call came i...
2018 Feb 16
2
incoming call label
On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, thelma at sys-concept.com wrote:
>
> <snip>
>
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-4444
>>
>> But asterisk display:
>> Found peer 'pstn-9998' for
2015 Apr 08
0
Asterisk Inbound calls, multiple SIP accounts, calledID
..."setvar=fromCompany=Company1" to Company1 friend section..
b) In dialplan add
Set(CALLERID(name)=${fromCompany} ${CALLERID(name)})
Maybe this will help?
Dmitiy.
08.04.2015 2:48, Andrew Galdes ?????:
> Hi Dmitriy and others and thanks for your help so far.
>
> The option "match_auth_username=yes" seems to have had no effect. From
> my reading, this option will try to match the username of the incoming
> SIP account to a section heading. If that is how it must work then i
> can see a big problem. I'm trying to present the receptionist with a
> nice display of wh...
2015 Mar 06
2
res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
Hello.
Asterisk 13.2.
I transfer configs from chan_sip to res_pjsip.
In chan_sip i have "match_auth_username=yes" and have nothing in pjsip.
I have a lot of endpoints and registrations on same SIP server. And it's
problem in pjsip now. Is not it?
I requesting to add new value for endpoint option identify_by. The value
'uri'.
Simple config (cutted):
[siptrunk]
type=registration
transp...
2015 Apr 01
4
Asterisk Inbound calls, multiple SIP accounts, calledID
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
with the same service provides. We have 8 phone numbers in total.
Incoming calls from the public are all correctly directed to appropriate
office handsets. However, the display on the reception phone (the only one
i care about) is always showing the same "SIP/Account1_0843214321" rather
than the account
2015 Apr 08
0
Asterisk Inbound calls, multiple SIP accounts, calledID
...rewgaldes> | Google+
<http://google.com/+AndrewGaldes>
*Platform Architects for High Demand Web Applications.*
On Wed, Apr 8, 2015 at 9:18 AM, Andrew Galdes <andrew.galdes at agix.com.au>
wrote:
> Hi Dmitriy and others and thanks for your help so far.
>
> The option "match_auth_username=yes" seems to have had no effect. From my
> reading, this option will try to match the username of the incoming SIP
> account to a section heading. If that is how it must work then i can see a
> big problem. I'm trying to present the receptionist with a nice display of
> which...
2015 Mar 06
2
res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
...0:24, Kevin Harwell ?????:
> On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com
> <mailto:serov.d.p at gmail.com>> wrote:
>
> Hello.
>
> Asterisk 13.2.
> I transfer configs from chan_sip to res_pjsip.
> In chan_sip i have "match_auth_username=yes" and have nothing in
> pjsip.
>
> I have a lot of endpoints and registrations on same SIP server.
> And it's problem in pjsip now. Is not it?
>
> I requesting to add new value for endpoint option identify_by. The
> value 'uri'.
>...
2015 Mar 06
0
res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote:
> Hello.
>
> Asterisk 13.2.
> I transfer configs from chan_sip to res_pjsip.
> In chan_sip i have "match_auth_username=yes" and have nothing in pjsip.
>
> I have a lot of endpoints and registrations on same SIP server. And it's
> problem in pjsip now. Is not it?
>
> I requesting to add new value for endpoint option identify_by. The value
> 'uri'.
> Simple config (cutted):
>...
2015 Apr 02
0
Asterisk Inbound calls, multiple SIP accounts, calledID
This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes
Carefully read the description, it is better to test in "after hours".
02.04.2015 2:50, Andrew Galdes ?????:
> Hello all,
>
> I have an Asterisk server (Asterisk 10.12.4) with multiple sip
> accounts with the same service provides. We have 8 phone numbers in
> total...
2011 Jul 29
0
Asterisk SIP authentication against [section] instead of username
Hello,
Asterisk seems to try to authenticate incoming INVITE based on the [section]
in sip.conf and not the username specified.
I just removed the "insecure" option from my sip.conf requesting every
connection to be authenticated. I added the match_auth_username=yes in the
[general] section for extra security. To make it work, I have to use the
same [section] identifier as username. This is really bad because if
multiple provider are giving me the same username, it doesn't work.
If I put the following data in sip.conf, it doesn't work. Asterisk re...
2015 Mar 06
0
res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
...t; wrote:
> 07.03.2015 0:24, Kevin Harwell ?????:
>
> On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com>
> wrote:
>
>> Hello.
>>
>> Asterisk 13.2.
>> I transfer configs from chan_sip to res_pjsip.
>> In chan_sip i have "match_auth_username=yes" and have nothing in pjsip.
>>
>> I have a lot of endpoints and registrations on same SIP server. And it's
>> problem in pjsip now. Is not it?
>>
>> I requesting to add new value for endpoint option identify_by. The value
>> 'uri'.
>> S...
2010 Jun 04
1
originating a sip call from the CLI
Hello again!
I just got a SIP account and it seems - from a config on the net -, that
I've configured it correctly. But I get no call to the outside. Registration
was OK.
I tried:
channel originate sip/1/echo at iptel.org Application ...
I see the channel active for a while, but no call gets established.
In my config I have defined the section [iptel] for the outgoing call and I
2017 Feb 09
3
Disallow CALLS without registry
HI ALL
got small question
i use call-limit=1 on peers
but call limit is not working if user is not registered on PBX and
making calls
so the main question is -- how to Disallow CALLS without registering on PBX
--
Best regards
Antony
tel. +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy
2011 Jan 28
3
Disabling Music On Hold
...ured default language
dtmfmode=rfc2833 ; default dtmfmode for sending DTMF
(Dual-tone multi-frequency)
directrtpsetup=no ; Disable the new experimental direct
RTP setup
allowtransfer=yes ; enable all transfers for peers and
users
match_auth_username=yes ; use 'authentication username'
instead of 'username for authentication' (if available)
session-timers=originate ; Request and run session-timers
always
session-expires=3600 ; maximum session refresh interval
session-minse=600...
2008 Oct 19
6
adding a second extension
I'm trying to add a second extension to my setup. The second device is
able to successfully connect to the Asterisk server. I am unable to
contact extension 101 from 102 and vise-versa. Also are my context
setup logically or is there a better fashion to organize them? My
error is at the bottom.
Here is the extension.conf
[default]
;
; By default we include the demo. In a production system,