If I understand correctly, settingencryption=nomeans that Asterisk will make outgoing calls without encryption, but will be happy to accept incoming calls regardless of whether the caller wants encryption or notIf encryption=yes, then Asterisk not o..
The sample config files in the Asterisk distribution and packages are really good for getting the demo up and running quickly, for example, to extend the demo to run behind a WebRTC proxy only required about 6 lines of extra code to define a peer..
Is the template capability in sip.conf compatible with realtime sip.conf entries such as users in a database?I notice that contrib/realtime/mysql/sippeers.sql and the wiki page dont mention a template column:https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structurewh..
Kamailio has both a ha1 and ha1b column in its user schema:ha1 = H(A1) = MD5(user:realm:password)ha1b = H(A1b) = MD5(user@realm:realm:password)This is intended to support some devices that append @realm to the user and/or to allow users to put eit..
Ive now prepared a blog about my experience setting up Asterisk 11 with repro as a SIP proxy for WebSocket clients:http://danielpocock.com/using-resiprocate-to-connect-asterisk-webrtcIn particular, the focus is on the use of packages because that ma..
Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk?In particular, I notice many of the references to XMPP on the wiki link tohttps://wiki.asterisk.org/wiki/display/AST/Calling+using+Googlewh..
Ive just done a test with a WebRTC client connecting to the repro proxy with the SIP messages relayed over TCP to AsteriskAsterisk successfully answers the call using SAVPF, SRTP and ICE.The client is greeted by the demoThis was tested in the Aster..
Building from the source RPM I get an error mISDNuser-devel is neededI was able to obtain all the other build dependencies from EPEL 6, but that one doesnt appear to existing in EPEL or in packages.asterisk.orgI then tried adding –nodeps to the rpmbu..
I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.orgI followed this guide:https://wiki.asterisk.org/wiki/display/AST/Asterisk+PackagesThe SRTP support appears to be missing though.I notice libsrtp was not automatically instal..
Ive set up a peer to use G.722 only and tried to make it talk to an Asterisk boxAsterisk always rejects the call with the following error:[Jan 14 22:20:16] WARNING: chan_gtalk.c:1343 gtalk_newcall:Capabilities dont match : us – 0x4 (ulaw), p..