Archives : November-2021
to the whole list.i need help i cant solve the problem.I need to be able to track in a log file when a user answers the phone.I want to get a log composed as follows:calling number:number called:extension that answered the call.now i can log in w..
on an asterisk16/Buster compiled by Debian team we suddenly face errors like channel.c: Exceptionally long voice queue length queuing to IAX2/-7768 aso followed by chan_iax2.c: Max retries exceeded to hoston IAX2/-7768 (type = 6, subclass = 11, ts=600..
Also a addition:If I call from the 8961 TO the ACR pho..
Here is my problems (sent a copy to the developer of ACR Phone aswell): I have a mobile phone with ACR Phone client, and a fixed Cisco 8961connected to a PBX with DID. Incoming calls answered in the 8961 – audio works both ways fineOutgoing calls cal..
– Any one using SIPML5 ? How many video connections can a normalasterisk serverbox (2.2G 8GIG ram) handle in a SINGLE video session ?Th..
Since Macro is deprecated I am trying to eliminate it from my diaplan. Ibelieve I have successfully done so in the example below.; dial an internal extension exten => 101,1Macro(ext,100,Dahdi/15)TO:exten => 101,1,Dial(Dahdi/15,30)So far it seems to wo..
We currently use the Queue.Under app_queue, it uses module res_monitor (which is on the to be deprecated list). Is it safe to continue using Queue (app_queue)?DanThis email is intended only for the use of the party to which it is addressed and may cont..
Justrecently moved over from chan_sip to PJSIP and am slowly cleaning up whatever needs to be. I cant seem to make sollicitated MWI work, but unsollicitated works fine. I got my phones subscribing to mailbox@context (i.e. 100@whatever) I have my rela..
I am trying to use the SIPML5 at https://www.doubango.org/sipml5/call.htm?svn%2and when I hit the login button – and asterisk says wrong password and the web page says Forbidden.I have triple checked that I entered the correct password on the websi..
Hi. I have a setup which comprises some front-end Asterisk servers which have SIP trunks to external providers, and very simple dial plans, and some back- end servers which only talk to the front-end machines, and have the majority of my dialplan lo..