is there an option to log calldate / start in GMT / UTC?
CSV has an option usegmtime=yes.
Best R..
Author : basti
Home » Articles posted by basti
i try to connect my SIP Client (linphone) via VPN to FreePBX. The routing looks OK. I can ping the Endpoints and traffic is routing. I can also Register my Sip Client. debpbx*CLI> pjsip list contacts Contact: =======================================================================================..
i try to use mail2fax with asterisk. all i have found is old stuff like – https://sourceforge.net/projects/asterfax/files/production/NoojeeFax or – https://github.com/siddolo/sidfax so I try sidfax (aka email2fax). Now asterisk show in sip debug m..
I use Asterisk 13 with FreePBX. When I try to connect my Softphone via VPN to Asterisk Im registered and Its show via pjsip list contacts Then I try to call an internal number / other extension I get the following: SIP/2.0 401 Unauthorized. The VPN ..
Hallo, is there a Freepbx mailinglist? or can this be posted here?
Best Re..
is there a way to view the call log (call history) from extern via browser, XML or whatever? At the moment I see no numbers in call-log. Asterisk do the CIDLokkup at the moment, and the Ring Groups has an CID Name Prefix. So an a Phone you can see: cal..
when I set qualify = yes on trunk I cant do outgoing call. Incoming is always working. [Feb 15 23:01:41] WARNING[12909][C-00000012]: app_dial.c:2525 dial_exec_full: Unable to create channel of type SIP (cause 20 – Subscriber absent) but my linphone..
again, my Grandstream GXP1610 is show the call on display but no ringing is hearing. The caller hear a ring tone but the phone is like muted. With an other phone all is fine so i think thats a phone setting. Has anyone a solution for this? I have fi..
I have some user that had have a hardwarephone and an softphone. I use pjsip driver and set Max Contacts = 2 to have register both at the same time.
But Only the softphone is ring. the hardware phone is mute.
How can i fix..
my Asterisk is installed on my router. From my ISP I only get an dynamic IP. In sip.conf I have try: externhost=host1.mydns.unix-solution.de externrefresh=300 but after reconnect I cant call from outside. asterisk*CLI> sip show registry Hostdnsmgr Usern..