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DDNS Troubles
chrischevy: Since there is no external phones, I don't think that it would affect the system. Without an external address, you won't be able to call into the system. If that's not a problem, you're...
View ArticleWMI on Grandstream HT702
i have a pots phone connected to FSX1 and have WMI & polling enabled, but I don't get any indicationI have a Grandstream 1625 IP phone and the WMI is workingI have tried both a Plantronics &...
View ArticleDDNS Troubles
Isn't that only an issue if you have external extensions ?I thought that these "public IP" settings were only useful for external extensions. I did the test and it seems to be working fine. I have a...
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Unable to sign in to community
sorvani: And that helps how? I clearly stated the problem and included an image. Sure. But how does it help to state what you stated. Either way it doesn't work for you sometimes but it works for me...
View ArticleFreePBX after upgrade admin dashboard not loading
rm -Rf /var/www/html/admin/modules/sysadmin
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Why I get "res_pjsip_header_funcs.c: This function requires a PJSIP channel"?
Hi all, Fresh install of STABLE SNG7-FPBX-64bit-1706-1 (Asterisk 13)I get loads of errors "res_pjsip_header_funcs.c: This function requires a PJSIP channel"I have tried to Google it but I have not...
View ArticleWhy I get "res_pjsip_header_funcs.c: This function requires a PJSIP channel"?
Are you having issues? These are harmless and you can ignore them. There are places where FreePBX call flows examine SIP headers Since there are separate dialplan functions for each channel driver,...
View ArticleHotdesk and Queue
Any way to have hotdesk log your extensions into any dynamic queue you are a member of without having to login via hotdesk, then login to the queues as seperate function? Thanks in advance.
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Grand Stream GXP2000 Phones
Like I have repeated here over and over again its not your server's fault. The sip debug is useless in this case, You will see the same output always. If you and cullen understand how SIP works you...
View ArticleExtension rings once then "No Answer"
We have a queue setup with 5 people in linear mode. All exts ring fine up till the last one (1332). This user starts to ring properly and then after one ring the asterisk log says "No answer". I have...
View ArticleSNG7 Admin password change
I have used the below before on SNG7. mysql -D asterisk -e "UPDATE ampusers SET password_sha1=SHA1('YourPassword') WHERE username='YourUsername'"
View ArticleCustom Contexts applied to a IAX2 trunk
Sorry but the PBX2's routes are managed by another VoIP administrator, so I can't trust him. That administrator is the owner of PBX2, and ?'m from PBX1. I need a way that let me filter the incoming...
View ArticleSeveral Numbers with one Sipgate Basic account
There is another problem:I just found out, when I call from another sipgate account, the "To: " field in the header is the sipgate account number of the receiving account. When I call from my mobile...
View ArticleMaximum PBX stack exceeded
We are having issues with this: [2017-05-16 13:30:31] WARNING[32615][C-00002516]: pbx.c:2461 pbx_find_extension: Maximum PBX stack exceeded We were having issues with it before but then we condensed 6...
View ArticleGrand Stream GXP2000 Phones
I'd be happy to troubleshoot this on IRC. Join us in Freenode at #freepbx. Not willing to fight anyone off in order to accurately work your problem. If you don't have an IRC client, or can't use the...
View ArticleSNG7 OpenSSL Error
Hello,i tested the new SNG7 Distro and i have some problems with openssl. I have installed the Nginx from repository and tried to connect a phone with a valid SSL Client cert with the server. Nginx...
View ArticleSangoma Phone - PJSIP Registration Issue
The entries you are seeing with the sip debug are the result of a bug in EPM that only shows itself when using PJSIP (they will not be related to any registration issues)....
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SNG7 OpenSSL Error
mitterhuemer: Does anyone know how to solve this? The distro uses Apache. We don't support nginx. Sorry.
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