How about if you set; exten => _sexxxx,1,Dial(IAX2/cloud/1000,30,r)Mc GRA..
Has there been any discussion about the the effect of the changes in net neutrality to VoIP service quality.It seems to me that prioritizing streaming traffic from certain content delivery companies could have an impact on the latency for VoIP wh..
2017-12-14 16:28 GMT+01:00 Tzafrir Cohen :Im a bit ashamed of this but I must have forgotten an apt-get update, before trying to install asterisk-dbgsym. I did it and it worked perfectly.Sorry for the noise !How would you roughly evaluate this performa..
all,Im trying to resolve a weird issue with SIP routing.I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf:[gener..
Im used to install Asterisk on Debian stable platforms.A customer is asking how I would proceed on a CentOS platform.After a short research (see  as an example), Im wondering what are general kernel practices on CentOS regarding Asterisk and w..
I am new on asterisk and do some tests on freepbx.I have 2 SIP provider:Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbersOn Asterisk site i have 3 phones(branch ??, dont know ..
Im giving HangupCauseClear() a try on a Debian Stretch / Asterisk 13.18.3stack.My dialplan is:exten = 1234,1,Set(CHANNEL(hangup-handler-push)=myhandler,s,1)same = n,Dial(SIP/foo/1234)same = n,Gosub(myhandler,s,1)same = n,HangupCauseClear()same = n,Dial(SIP/bar/1234)[myhandler]ex..
Currently using PJSIP.First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing similar work.For PJSIP… I currently have an endpoint configured to a system using IP based authentication..
The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15.The available security releases are released as versions 13.13-cert9, 13.18.4,14.7.4 and 15.1.4.These releases are available for immedi..
I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled.When I receive a SIP INFO, the logs tell me that a DTMF begin is generated, but no related DTMF end is generated, unless the call is ended. Here is an excerpt..