I am trying to get Asterisk 11 to co-exist with a CentOS 7 box that has pulse audio running as a l..
Author : Jerry Geis
Home » Articles posted by Jerry Geis (paged 3)
I use both confbridge to bring several devices into a receive only or listen mode, then allow the one person on the phone to speak live over those devices. Works great.However – now I would like to be able to play a toneinto the conference before ..
I see in my log file this:Jun 30 21:44:26] NOTICE[42192][C-000002f3] chan_sip.c: Call from (5.189.144.120:5076) to extension 011972592675431 rejected because extension not found in context default.which is great its rejected – however in my sip.c..
I am looking for some great instructions on using asterisk with pulse.Im using CentOS 7 and pulse as a user and not having much luck.I have changed all permissions for the asterisk directories. set asterisk.conf user and group to be my user that is runni..
all,Anyone have any great ideas on getting asterisk 11 to work with pulseaudio?I chown user:group /var/lib/asterisk/var/run/asterisk/var/spool/asterisk/var/log/asterisk/usr/lib/asterisk/etc/asteriskchange user and group in asterisk.confwhen is run..
I am using 11.17.0 -and MulticastRTP. Doesnt seem to work with polycom phones as other devices receive my multicast just fine.Is there something special to do to get multicast working with polycom phones?(other than enable multicast on the actual phone).T..
I have two machines on the internet. Box A and Box B.Box A has a SIP trunk to the world, Making calls box A works fine I have audio to my cell and all works.I defined a SIP trunk between box B and A. tried to make a call originating from box B – to ..
I have a machine with three IP addresses. NIC eth0NIC eth1and a virtual address on ETH1All my devices work normally communicating to the virtual address on eth1. My question is just for mulitcast.The end device has an option for allowed source so I ..
I am running a heartbeat… Asterisk 11.15.0 – same behaviour is noticed on1.4.43 alsoI issue a call through the API that does the below. just UserEvent and Hangup — Executing [s@heartbeat:1] UserEvent(Local/s@heartbeat-0000000f;2,HeartBeat, Noop)..
I have a PRI working just fine.There is an automated system that needs to hear at least one ring. Currently when I call into asterisk it answers right away and Ido not hear the ring at all. I am 1.4.43 for this systemis there a way to set answer on r..