Archives : August-2019
We have a system where two calls are in a ConfBridge with recording.This is Asterisk 16.3.0Channel A seems to work perfectly.Wireshark is showing the RTP to/from working fine and having no jitter/lag issues.This call hears everything from channel B.Chan..
The aco_option_register function is used by modules to register some configuration handling logic. The bridge feature API is defined in bridge_features.h[1] but Im not sure functionality you require (arbitrary adding of features) was ever added to ..
I am using these variables in my callfiles:
CallerID: My Fax-ID
setvar:FAXOPT(headerinfo)=My Fax-ID
setvar:FAXOPT(localstationid)=001234567890123
regards, andre
Am 03.08.19 um 19:00 schrieb asterisk-users-request@lists.digium..
HI I used asterisk 13s app_fax to send and recv fax. On the received fax/pdf, I got a fax with header like this:26-Ju-2019 | 06:03| my-test | unknown | p.1The my-test is set by LOCALHEADERINFO before I call send_fax. What I am not clear..
Anyone have any decent ways to handle post dial delay on asterisk? Doesnt seem like theres a timeout I can set for it. Id love a PROGRESSTIMEOUT=field in PJSIP. Basically something to bring down a call attempt if the delay between 100 and 18X is >30..