need help setting up a Yealink with FreePBX without End Point Manager
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Which phone do you have. I have the SIP-T20P.
You should be able to use the basic settings and get signed it.
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Sorry about that, yeah I guess that would help.
I have the SIP-T22P.
what is a basic setup? A Generic PJSIP Device? Do I need to do more than just put and Extension and name on it when provisioning it?
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Under one of the extensions I created (PJSIP) it says
This device uses PJSIP technology listening on udp:0.0.0.0:5060 -
Here's the /var/log/asterisk/freepbx.log file
[root@c1115759-27921 asterisk]# tail freepbx.log [2015-Apr-10 14:48:52] [INFO] (libraries/modulefunctions.class.php:1864) - Generating CSS...Done [2015-Apr-10 14:48:52] [PHP-NOTICE] (/var/www/html/admin/page.modules.php:367) - ob_flush(): failed to flush buffer. No buffer to flush. [2015-Apr-10 14:48:52] [PHP-NOTICE] (/var/www/html/admin/config.php:534) - ob_end_flush(): failed to delete and flush buffer. No buffer to delete or flush. [2015-Apr-10 14:49:05] [PHP-WARNING] (/var/www/html/admin/modules/cxpanel/brand.php:3) - file_get_contents(/etc/schmooze/operator-panel-brand): failed to open stream: No such file or directory [2015-Apr-10 14:49:05] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 48 [2015-Apr-10 14:49:06] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/general.php:41) - Undefined index: deploymentid [2015-Apr-10 14:49:17] [PHP-WARNING] (/var/www/html/admin/modules/cxpanel/brand.php:3) - file_get_contents(/etc/schmooze/operator-panel-brand): failed to open stream: No such file or directory [2015-Apr-10 14:49:17] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 48 [2015-Apr-10 14:49:17] [PHP-DEPRECATION_WARNING] (/var/www/html/admin/libraries/BMO/GPG.class.php:510) - Comments starting with '#' are deprecated in Unknown on line 20 [2015-Apr-10 14:49:17] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/general.php:41) - Undefined index: deploymentid -
OK found the other log
[2015-04-10 15:06:31] NOTICE[16181] res_pjsip/pjsip_distributor.c: Request from '"1280" <sip:[email protected].>' failed for '69.:44868' (callid: 309444733@) - No matching endpoint foundOK a path to head in.
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Ok switched over to CHAN_SIP
[2015-04-10 15:11:01] NOTICE[14872] chan_sip.c: Registration from '"1290" <sip:[email protected]:5061>' failed for '69.63.:17005' - Wrong password -
I setup ~75 Yealink T3XG phones manually. It is an extremely straight forward process. Sorry I realize that isn't very helpful. What problems are you running in to? It just won't register?
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Here is a screen shot from my phone:
Label and display name: username
Register name / user name: valid extension
Password: passwordServer Host: IP or some.domain.com
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I'm made some progress.
When creating my user in FreePBX under Applications > Extensions
I didn't realize I had to click the RED Apply Changes button at the top to make them active.
So now my phone tries to connect, and the log if crammed with all kinda of update files, etc trying to be pushed to the phone.. I think I'm hanging up on that now.
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@Dashrender said:
I didn't realize I had to click the RED Apply Changes button at the top to make them active.
You will need to do this for almost every action you do on FreePBX. This runs the Asterisk "reload" command and some other things in the background to apply config settings.
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Yes! I am golden with my first extension!
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Does it matter if I use PJSIP or CHAN SIP?
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@Dashrender said:
Does it matter if I use PJSIP or CHAN SIP?
I believe that Asterisk 12 is going to be pushing PJSIP by default soon. I remember reading that somewhere. It apparently is a lot more lightweight alternative to CHAN SIP. So I don't think it matters.
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Great
I now have to phones setup, a T22P and a Zioper softphone.
when I dial one extension from the other the one I'm dialing from rings, but the one I'm calling does not.
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OK tried calling from the Zioper and received an error messagae
It is also seen when the contact on the other side uses an unsupported by his PBX server audio codec (which causes him to seem to be offline).
OK might have a codec conflict.
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I really really have no idea how you have went so far south on this process.
It is literally a snap to do from a full ISO install point of view. I realize that you had problems because you installed into an existing CentOS box. and that is outside the normal flow (even if it is supported).
But once you had it up and running, you should drop into the GUI, add the ext, drop to the phone add the credentials, done.
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@JaredBusch said:
I really really have no idea how you have went so far south on this process.
It is literally a snap to do from a full ISO install point of view. I realize that you had problems because you installed into an existing CentOS box. and that is outside the normal flow (even if it is supported).
But once you had it up and running, you should drop into the GUI, add the ext, drop to the phone add the credentials, done.
He is doing this on Cloud@Cost, which doesn't allow custom ISOs to be uploaded.
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@coliver said:
@JaredBusch said:
I really really have no idea how you have went so far south on this process.
It is literally a snap to do from a full ISO install point of view. I realize that you had problems because you installed into an existing CentOS box. and that is outside the normal flow (even if it is supported).
But once you had it up and running, you should drop into the GUI, add the ext, drop to the phone add the credentials, done.
He is doing this on Cloud@Cost, which doesn't allow custom ISOs to be uploaded.
Exactly.
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OK I now have two T22P, they are both registered, yet I can't call between them.
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@Dashrender said:
OK I now have two T22P, they are both registered, yet I can't call between them.
Have you looked at the asterisk console when trying to make the call? It generally has some verbose logging.
