Bookmark the permalink. These both take port numbers and it isn't clear to me if they should be the same port. I am now running FreePBX 13 (10.13.66, Asterisk 13). I am trying to do the same thing and I can't seem to get it to work on incoming calls. have a peek at this web-site
In general, however, it's totally insecure because it bypasses any further authentication (including my issue with the username mismatch). On my other machine, same version of Asterisk, I have added SIP Trunk registered to that extension (200). Anyone know how to tell asterisk to accept this format of username in the digest authentication? Go to INBOUND ROUTES and select add incoming route3.
The add new extension screen says that SIP uses port 5061. Doesn't the Toshiba support a trunk? I can call out on the side that has the registered extension from the other switch. Encryption in the 19th century Best way to change site IP address - from the end user perspective?
Make sure you have authuser=XXXXXXX set in your peer details and the authuser included in your register string like this username:secret:[email protected]/username. Note that although the add extnesion screen said SIP uses 5060, the advanced tab screen for extension 100 had the port set to 5060. Likewise with the toshiba. Freepbx Failed To Authenticate Device I think it doesn't even reach routing contexts, here is how it looks like in logs:
[2013-12-13 11:24:31] WARNING: chan_sip.c:14558 check_auth: username mismatch, have <130>, digest has
[2013-12-13 11:24:31] NOTICE: chan_sip.c:22796 handle_request_invite:
The calls were coming from Sipura 3102 and no settings were changed other than upgrading the framework. Tell the world! killpl 2013-12-13 10:40:33 UTC #3 I've checked again with from-internet and dynamic host, still doesn't work. Investigation I Tailed the log in /var/log/asterisk/full I found the following errors when an incoming call was placed: WARNING chan_sip.c: username mismatch, have , digest has <> NOTICE chan_sip.c: Failed to
It is just called inbound as a convenience in the trunk page. No Matching Endpoint Found Freepbx If its only the IVR that cannot be reached then it would still appear to me that the inbound route to the IVR is not set up correctly irishAndy_24 2012-05-16 08:04:45 Construct a Gaussian Matrix Is using Basic Authentication in an iOS App safe? That's the least of your trouble.
or delete the one, recreate a new one and have a try? Set the DID number to the extension number you created for the IVR4. Freepbx Check_auth Username Mismatch Have Digest Has The biggest clue was pstn between the brackets because the name matched the inbound route I had setup for the landline. Handle_request_invite: Failed To Authenticate Device Leave a Reply Cancel reply Your email address will not be published.
This must match exactly the name of the inbound route name. Check This Out share|improve this answer answered Apr 8 '10 at 20:48 Matt 161114 Not sure if this is ok, but accepting my own answer with nothing better posted. –Matt Apr 12 However, the big discovery was that Simple Setup in Polycom's VVX310 was not sufficient to perform registration because another setting under Settings->Lines[Authentication] had a radio button setting 'Use Login Credentials' which gardnerale (Gardnerale) 2015-11-17 13:09:56 UTC #3 I have deleted the user and re-created the user under User Management module and used a different secret. Insecure-invite
Connected to Asterisk 13.5.0 currently running on localhost (pid = 8928)[2015-11-16 15:09:34] WARNING: chan_sip.c:16653 check_auth: username mismatch, have <100>, digest has <>[2015-11-16 15:09:34] NOTICE: chan_sip.c:27498 handle_request_subscribe: Failed to authenticate device "Thorpy Why do you need to register 16 extensions with the toshiba? Why not one and just send the extension you want to call down the trunk. Source Required fields are marked *.Comment All comments must go through an approval and anti-spam process before appearing on the website.
DID работает правильно. Натолкните на путь истинный... Is there an error in my configuration? Not the answer you're looking for? Where does the digest get its user name from?
I was guessing maybe registration is done through one port and the rest of it is through another port. The digits your carrier is sending must not match what you have in the route. Danny also has participates in a part time project called [email protected] [http://code.google.com/p/energyathome/] for monitoring energy usage on a premise. http://jscience.net/failed-to/failed-to-load-module-evdev-module-requirement-mismatch-0.html I had previously used 5060 and so ignored the 5061 and went with 5060 and the phones could not register.
Still the same result. Lithium Battery Protection Circuit - Why are there two MOSFETs in series, reversed? Cheers! In my example above this would be "pstn".
Any pointers here? The 3com phones are communicating SIP with the Asterisk, but are unable to register because they present a digest username value that doesn't match what Asterisk thinks it should. Here is the sip.conf info for that extension:  deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=1234 qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes [email protected] host=dynamic dtmfmode=rfc2833 dial=SIP/2321 context=from-internal canreinvite=no callerid=device <2321> allow=ulaw, alaw call-limit=50 ... The key is the from-internal context.
Scroll down the page to SET DESTINATION and select IVR from the drop down menu and then select the recording you want to use from the drop down menu next to When I added in the two extensions, using the same phones with the new secrets, I could not get them to register with the PBX server. Dedicated to I.T since studying pure Information Technology since the age of 16, Danny Tsang working in the field that he has aimed for since leaving school. I can make outbound calls but no incoming.
I thought insecure=port,invite should fix this but it didn't. What's the English word for something that given attention too much to Can a 50 Hz, 220 VAC transformer work on 40 Hz, 180VAC? Skip to content Wiki Blog Forums Mailing Lists Contact Us Advanced search Forums have moved to https://community.asterisk.org Board index RSS RSS Change font size FAQ Information The requested topic does not The extensions are setup as SIP (not PJSIP).
Check out the FAQ! Пожалуйста, войдите здесь. Часто задаваемые вопросы О нас вопросы тэги пользователи награды старый форум задать вопрос ВСЕ НЕОТВЕЧЕННЫЕ Задайте Ваш вопрос did inbound - username mismatch, have
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