Category : Asterisk Users
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Downloading asterisk-16-current.tar.gz is still showing Asterisk 16.19.0..
Is there a way to not compile/configure pjsip in 18 ?I am still using the older SIP channel driver and have not converted over just yet.Th..
The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and 16.8-cert10.These releases are availa..
I started noticing a few days ago that whenever I dial any number or extension there is a delay of 5 to 10 seconds before Asterisk reacts. I see nothing on the CLI for that time and then the call goes through. I have checked my network..
Hi. I have the following situation: An Asterisk 16 server on which I have complete control of the dialplan, and which has (a) a SIP trunk to a PSTN gateway provider, and (b) several SIP credentials for accounts (extensions) on another Asterisk server…
I need to check the return value of a sub, the sub may return empty so Ineed to check for that. If the return value isnt empty set another variable (ARG1) . This is the code Ive used in extension.conf, but didnt work (the CLI log is after the code).*Extension.conf:*[macro-dial]s..
All.We have a provider that requires us to SOURCE the SIP connection on TCP 5061.I honestly have no clue how to force Asterisk to always SOURCE the SIP connection on a certain port.Can anybody point me in the right direction?I am using PJSIP.Thank y..
We have a project where people will be making payments over the phone. Iwould like block Asterisk from logging any time the system is processing a card. So be it SayDigits(123456789), when the user enters DTMF or when Ipass a card number as a varia..
I have an asterisk setup using pjsip. Everything used to work correctly until one remote site changed internet provider and thier router does not support sip protocol algorithms. It works for some time, but then suddenly audio stops working both directio..
I have asterisk behind kamailio, and i want to enable acl for clients that are connecting through the proxy. Basically when i go to asterisk and type sip show peers i seem them connected using the proxy IP. I saw in the headers that in the contact hea..