Archives : September-2019
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When using chan_sip with qualify=yes Asterisk seems to send the SIP OPTIONSdirectly and does not send them via OUTBOUNDPROXY. Has anyone else come across this and if yes what did you do to correct it..
Using Asteirsk 13.28.1:If I configure my pjsip transport to handle NAT from the Internet:[transport-tcp]type=transport protocol=tcp bind=10.75.22.8:5060local_net=10.75.22.0/24external_media_address=[external address redacted]external_signaling_address=[exter..
all,we have just upgraded from Asterisk 11 to Asterisk 16. After porting all the config to 16 we figured out some major load problems.the majority running of our Asterisk instances is still having Asterisk 11so we can compare load handling on both versio..
Hey everyone,I posted a topic on the community forum which can be found here:https://community.asterisk.org/t/request-for-feedback-queues-penalties-ringall-and-use-cases/80960It covers a scenario encountered while debugging an issue, and I would l..
I would like to know for App_Originate that how I can pass the argument to next dialplan as listed below partial dialplan;same => n,GotoIf($[${COUNT} > 0 ]?bridge_conference,1)same => n,Set(__TMPEXTEN=${EXTEN}1)same => n,ExecIf($[0${CONFBRIDGE_INFO(parties,${EXTEN..
Hello;I have 10 Caller IDs and I need each call (each time) to use one of these Caller IDs to be the caller id. I know that I can use this syntax as example:exten => _90ZXXXXXX,1,Set(CALLERID(num)747576)But how I can set the callerid each time f..
Im trying to track down a CPU spike we are seeing in a system.We have tracked down the spike to a single CPU and TID using that CPU.Indications are that its asterisk running this TID.Im trying to figure out what asterisk is doing on this thread aro..
How can I use an IF statement with a true value being a variable that has a colon in it?The colon in the true value variable is being taken as the delimiter for the false value.The only solution I came up with was some hackery to use STRREPLACE to repl..
all I maintain the above – it was set up by an external party with whom relations have now been severed by my employer. Quite early after the deployment it became evident that all .gsm audio files produced on this virtual instance at Azure via MixMoni..
Hallo, is there a Freepbx mailinglist? or can this be posted here?
Best Re..