Archives : May-2016
I finally secure SIP session between Asterisk server and a remote client. My questions is the following; do I need to open port 5061 UDP on my firewall or just port 5061 TCP for SIP sessions.? I am not interested in securing RTP only SIP sessions.Tha..
Dear List wanna configure click2call in such a manner that my asterisk box call two mobile numbers and connect both numbers to talk. I have configured voip gateway, my internal and external calls are working fine. please ..
I am having a strange problem with Asterisk 13 on a CentOS 7 plataform.I have several servers running on this configuration but a particular installation on a Dell PowerEdge 220 server is the one giving me the most problems.All installations are automa..
HI!Im trying to compile asterisk 13.8.2+ on openSUSE Linux but it fails. It seems file ./main/libasteriskssl.so.1 is present when it fails. Building 13.7.2 works without any problem. It fails since 13.8.0.$ ./bootstrap.sh$ ./configure$ make menuselect.makeopts;menuselect/menusel..
all,I have a intriguing issue that the RFC is not really clear about. Sometimes call hang-up on 45min mark because no-one refresh the call ( far-end hangup)On both good and bad calls:1)We initiate an invite2)200OK is answered as refresher=UAS3)Send AC..
I am trying to secure SIP session with TLS on Asterisk Server 1.8. Ikeep getter an error, == Problem setting up ssl connection: error:14094418:SSLroutines:SSL3_READ_BYTES:tlsv1 alert unknown ca[2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection:F..
Have a strange issue at a customer, they went and replaced all of their old PoE switches with brand new HPE 5130 EI Switch Series.Their PBX has been up and stable for several years with no recent changes, but since they change the switches they are hav..
It looks as though something might be going wrong in the AGI script itself.Did you use a proper AGI library, or a quick-and-dirty homebrew solution?(There is little virtue in walking all the way to the tool shed to fetch a chisel, if you know the screwdri..
I have an Asterisk 13.8.2, which is supposed to be only a client to an encrypted SIP service. All local phones are connected via UDP.Since I cant use PJSIP (see my mailing list post from yesterday), Itried configuring chan_sip to work that way. My setti..
2016-05-03 16:43 GMT+02:00 Matt Fredrickson :OKYes, I think issue must come from incorrect Audiocodes settings. Requiring T.38 settings within first INVITE seems very unusual.Thank you very much fo..