Weve recently hit an issue with Asterisk 18.8.0 where a call comes in via SIP (using pjsip) but it can take 5 seconds before starting to execute the dialplan. This was intermittent, but frequent (eg approx half of the calls). We have verbose logg..
Author : Kingsley Tart
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We have instances where we dial multiple destinations simultaneously and then an answer macro prompts the callee to press 1 to accept the call or 3 to reject. Previously if they pressed 3 (or just hung up) the other destinations would continue to ri..
I see I can set qualify_frequency (for UDP) on an AOR to keep open holes through firewalls etc, and in [global] I can set keep_alive_interval for TCP based transports. However, is it possible to configure it so that these OPTIONS keepalives only ..
Im using Asterisk 18.8.0 with pjsip version 2.10. With a database defined endpoint, I cant find a way to define outbound_proxy with ;lr (without the quotes) on the end. It works fine if I configure an endpoint in pjsip.conf, eg: — 8< --------------------------------------------------..
I cant get Asterisk to send a SIP call to Twilio over TLS because it complains about Twilios wildcard certificate. This is with Asterisk 18.8.0 and PJSIP 2.10 pjsip show transport shows me this: allow_reload : false async_operations : 1 bind : 0.0.0.0:5..
my last few emails to this list havent appeared so Im just te..
I realise that this is not really specific to Asterisk, but this seems as sensible a place to ask as any. If I want to create a script to automate the build of my chosen Asterisk setup, whats the best way to automate my selections that I did interactiv..
When dialling a remote SIP host with PJSIP, is it possible either within the dialplan or via the AMI to find out the IP address of the remote host? If for example a remote host has multiple A records, I would like to know which one Asterisk has connec..
Im using Asterisk 18.7.1. I cant get res_fax to load. I built it accidentally with app_fax enabled, and was getting this in the log on startup: [Nov3 11:52:31] ERROR[10886] loader.c: Error loading module res_fax_spandsp.so, missing dependency: res_..
Im using Asterisk 18 to receive a call via SIP, dial a different SIP destination and bridge them together. However, even if the destination indicates that it doesnt support telephone-event, Asterisk is still sending DTMF as events, not transcoding..