Archives : July-2017
Hi!I just tried setting up Asterisk realtime database following the wiki article https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a Debian 9 machine (which switched from MyQSL to MariaDB).One has to install mariadb-plugin-conne..
is it possible to increase talking volume for caller in ConfBridge as standard without need to press buttons after joining an conference room.best rega..
I am tryint to get to https://wiki.asterisk.org/wiki/display/AST/Function_REGEX both via V6 and v4 and it seems to be t..
Hi.Im trying to get a list of the channels currently in use on an Asterisk server (1.8.32.1 if it matters) over AMI.I send the command sip show channels, and I get back a response along the lines of (* used to protect the innocent…):Peer User/ANR C..
guys,Could you please let me know whether the latest Asterisk has a support for inbound UPDATE ?In my case, the carrier is sending an UPDATE to change the codec which is replied by 5xx from Asterisk 11.17..
Ok, so a few years ago, when 13 first came out, I was having a core dump (crash) issue with asterisk 13. I worked with Josh some and even used my Digium subscription for support. Never was able to get it fixed at that time so let it go. Well now I..
Currently the AMI Queues action outputs the same text that the CLIoutputs when running a queue show command, which does not conform with the AMI spec. It should follow the same format as other AMI actions, structured in a key value list. The QueueSta..
I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technol..
SGV5IGZyaWVuZCwgDQoNCg0KSnVzdCB3YW50ZWQgdG8gc2F5IGhlbGxvIGFuZCB3aXNoIHlvdSAg YSBnb29kICBkYXkhIEJ5IHRoZSAgd2F5LCAgaGF2ZSAgeW91IHNlZW4gdGhhdCBodHRwOi8vd3d3Lmphc21pbmVjaG9wcy5jb20vc2VwYXJhdGUucGhwPzMxMzANCg0KUGFyZG9uIG15IG1vbmtleSB0aHVtYnMsIFNhbHZhdG9yZSBDZW50b..
Does anyone know whether chan_sip in Asterisk supports DTMF in clock rates other than 8000? I looked for telephone-event/16000 in the changelogand in Jira but no luck.Any help would be ap..