Archives : August-2019
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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..
I am running several Asterisk boxes with realtime around the world. Does anyone have a recommendation for a light db that would work with Asterisk that would also allow replication between all sites (so if I add an entry to one box it will work w..
all – has anyone used gstreamer to have and audio source (some music playing on a PC for example) and then use Mp3Player() to play that source? what is the gstreamer command to do that ?T..
Ok, so this might seem weird, but hang with me on this. I have two sites, Indy and Lafayette that each have their own Asterisk server. They each have their own outside PRI line. They are also trunked internally via and IAX tunnel over a private fi..
–cBz73V41rX4G2GXAJWymOOACf5HfIvaNf Content-Type: text/plain; charset=utf-8Content-Language: en-USContent-Transfer-Encoding: quoted-printable ,Im having a strange problem when using pjsip wizard and reloading(pjsip reload on CLI): some data (specifica..
Does anyone out there know of male french talent for Asterisk sound files where the talent already recorded the bulk of the Asterisk sound files?TIA.Reg..
list, Im looking for a solution that can be applied to a stock asterisk 16 (pjsip if it matter) running Debian 9 (php7.0). Statistics should be available for normal calls and queues using a WEB interface. Open source better but not necessary, Any feedb..