Archives : January-2019
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Joshua That is exactly what is happening. I get a separate CDR for the originate of the call to the agents extension, and then a separate CDR for the call that then goes out to the client. The separate CDR for an originate event DOES appear to be corre..
In article , Stefan Viljoenwrote: Tony Yes, that is exactly what Im doing… no Answer() calls anywhere in the dialout parts of my dialplan, as detailed in my previous posts. Yes, that is correct, I get a separate CDR for the originate of the call..
./configure LDFLAGS=-z muldefs –libdir=/usr/lib64–with-unixodbc=$(odbc_config –include-prefix)/ –disable-asteriskssl-enable-xmldoc NOISY_BUILD=yesres_pjsip/config_transport.c: In function ‘cipher_name_to_id’:res_pjsip/config_transport.c:982:..
All;I have an AudioCodes MP-114 four FXS ATA that recently stopped registering to my PBX. Im pulling my hair out here trying to figure out the root cause without much success. Does anyone have a sample config file that I could use as a sample? Any insi..
on an Asterisk 16 with PJSIP I want to know the state of a device (idle, busy, unavailable, …) in the dialplan. I tried with ChanIsAvail() but this one doesnt return the real state (eg a device calling an extension which is running ChanIsAvail()..
Regarding this Ive read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be eq..
Asterisk 16.1.0 Im using hagi and SRV records for a high availability configuration of AGI servers.When the first AGI server in the list is completely down, asterisk quickly moves on to the next one. That is all good. My concern is what will happen..
Hiya, I would have expected this to show the channels in the bridge inside the anonymous function – it shows the bridge is empty though? var bridge = ari.Bridge(); bridge.create({ type: holding, name: event.application+ bridge }, function(err, brid..
Asterisk 11.25.0 user here. Im trying to set up failing over to a second SIP peer if the first SIP peer doesnt answer on our SIP INVITE within 2 seconds. In sip.conf I set timerb=2000 for this peer, but it doesnt seem to have any effect. The time..
guys A few months ago I upgraded most of my Asterisk servers to 13 from 1.8. Ive still got about 25% of my servers on 1.8. Ive since noticed that ringtime on Asterisk 13 – the time difference between start and answer in the CDR record for any call, ..