Archives : July-2015
I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success.I have an application that sends an action Originate to AMI for calling, its working well, but when i see to Asterisks CLI, i see 2calls for just one originate:pftestes40copiabh*C..
I am trying to get Asterisk 11 to co-exist with a CentOS 7 box that has pulse audio running as a l..
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 ..
CAN U HELP MEIf there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidi..
GuysGiven these occassional errors on my Asterisk CLI:[Jul2 10:23:36] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission17bb3a993ad10f8818970ae952b81e73@192.168.11.31:5060 for seqno 102 (Critical Request) — ..
Howdy,I built an LXC container with an image of asterisk 11.18 precompiled and installed.It runs fine on the dev platform, which is a Dell R320 running Ubuntu 14.04LTS.I shutdown the container, tarred it up, and untarred on a Dell PE1850, also runn..
everyoneI want to know if it is somehow possible for asterisk to consider new registration attempts instead of matching them with old nonce Correct auth, but based on stale nonce received from test
allIs there someway ability to insert custom Header to SIP 486 message, when HANGUP application is invoked?Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HAN..