Archives : June-2019
! I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche Telekom. Asterisk works well, but I have really often an high delay (I understand it since the other party speak some seconds before he hears my question and answer) and someti..
Dear List Its probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multi..
This is using Asterisk certified/13.21-cert2, FWIW. I have a hangup handler on an outgoing SIP channel that grabs the SIP status like this: NoOp(keys=${HANGUPCAUSE_KEYS()} sipmsg=${HANGUPCAUSE(${CHANNEL},tech)}) This works fine if the call connects..
We have a need to record audio and allow the user to press any DTMF key to end the recording. Currently were using the AGI command record filewhich does allow us to specify which DTMF keys can end the recording.However we also need to know *which* ..
–_000_EFCDF2C6785A7B478B3A77A6E7C36369022F38C31Fmailxaccelnet_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableHelloAnyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSI..
Seems like I post about this about once a year, when its time to upgrade Fedora.I first got this error trying to compile a patched version of dahdi-linux-2.11.1; I noticed that there is now a dahdi-linux-complete-3.0.0+3.0.0, so I tried that one w..
Im trying to use linphone-android with asterisk but there is an aspect of the way asterisk and linphone-android interact with MESSAGEtransactions that is causing problems.The linphone-android folks consider both the To: and From: address in MESSAGE transacti..
We have a customer using ConfBridges. Party A is connected, audio is fine. We originate a call to party B through an Avaya switch.It forwards the call to IVR. The two channels are added to the same ConfBridge.Using a wireshark capture, I can listen..
If you are using JavaScript for *AGI/ARI/AMI we made small library for asterisk dialplan pattern matching and number manipulation https://www.npmjs.com/package/asterisk-pattern-matchingexamplesconst { validateNumber } = require(asterisk-pattern-matching..
I am receiving the following errors on any hangup handler subroutines. [2019-05-31 18:22:13.958] VERBOSE[23943][C-00000009] app_stack.c: PJSIP/104090401-0000000a Internal Gosub(PreventForwardingLoop,s,1)) start [2019-05-31 18:22:13.958] NOTICE[23943][C-000000..