Im trying to configure sip2sip, which says: http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it w..
Author : Jonathan H
Home » Articles posted by Jonathan H (paged 3)
Ive not needed to do a dialplan reload for a while, so I dont know exactly which version is stopped working, but on 15.5, Im not seeing ANY debug info at any debug level. So Im not really sure how to find mistakes in the dialplan.This is all I get..
Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.co..
Is there a bit more of a detailed explanation of TALK_DETECT anywhere?I googled and found nothing really beyond the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Function_TALK_DETECTI really only want it to listen for one side (the call..
Before I file a bug…. any ideas?Got valid certs, working fine for everything I need them from, but this seems to popup randomly in my logs.[Feb9 12:43:07] ERROR[14968]: iostream.c:507 ast_iostream_close:SSL_shutdown() failed: error:00000005:lib(0):func(0)..
So as yall know, with your help I managed to get Opus installed at last. Yay!With excitement, I wrote my dialplan, dialled in, and….[Jan 28 21:30:11] ERROR[29977][C-0000001d]: format_ogg_opus.c:95ogg_opus_rewrite: Cannot write OGG/Opus streams. So..
Before I got an log a ticket, can I just check Im not doing anything wrong?In 15.2, to install Opus:1) run `make menuselect`2) Highlight Codec Translators and press enter.3) Scroll down to codec_opus in the section labeled External4) Press enter to sel..
I want to start recording with a prompt of press or say 1 to 5. If no DMTF is pressed, I want to send the recording to Google Speech to get the number back (got that part working already).If any dtmf key is pressed while Application_Recordis runn..
I know that hangup handlers (https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers) have to finish quickly.So its no surprise that my speech to text agi which takes 8 seconds gets killed.However, can anyone think of a way round this? So, once ..
Briefly: I want to be able to have press or say (number), with Asterisk listening for a spoken number, but accepting a DTMF digit, too. Im posting everything I found so far, here, partly to show working, but also in case anyone else finds it usef..