Tag : sip
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CentOS-6.5 (FreePBX-2.6)Asterisk-11.14.2 (FreePBX)snom870-SIP 8.7.3.25.5I am having a very difficult time attempting to get TLS and SRTPworking with Asterisk and anything else.At the moment I am trying to get TLS functioning with our Snom870 desk-sets…
Hay guys, got trouble with registration with cisco 7975Here is the debug :REGISTER sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381From: ;tag=0c8525a68961001f44d503e2-d9359bd3To: Call-ID: 0c8525a6-89610004-b972d038-5864c98e@192.168.1.61Max-Forwar..
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 ..
Im reading the OReilly Asterisk the definitive guide, 4th ed, with a starfish on it.In some ways, astonishing that its not really that definitive, its more general — and it only clocks in at one ream of paper!In any event, Im having some port probl..
list. We have a problem with loss peers after sip reload, our configuration:Asterisk 11.6-cert1, SIP realtime peers, sip.conf: – rtcachefriends=yes- rtsavesysname=yes- rtupdate=yes- rtautoclear=yes When we do sip reload , peers are removing from availab..
Im investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. Im investigating, but can anyone provide some feedb..
Im currently evaluating asterisk 13 (Currently on 11).Were testing the migration from SIP to PJSIP.Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect816.749.2815 (Internal: e..
Two years ago we added logic to parse the isup-oli parameters, that arrive as part of the FROM Sip header. We need to finish the job and allow setting of this parameter for outbound calling, both in traditional SIP channel and PJSIP, which I beli..
all,Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111.222@mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is
all Over the last couple of..