search for: match_auth_usernam

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 be...

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...

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

..."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 w...

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 trans...

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

...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...

...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'. >...

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): >...

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 > tota...

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 r...

...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'. >>...

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

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

...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...

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,