Scenario:Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisks wiki, the transp..
I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transp..
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..
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?or is chan_pjsip better supported?or the recommended way for asterisk is use respoke.io?my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)chan_sip.c:10496 process_s..
Morning,We recently pushed our Asterisk video bridge into a DMZ and since then, local calls have been unreliable to say the least. While offsite calls work nicely, calls on our internal server usually fail to ring the far end. Two test calls that w..
,I have a fresh install of Asterisk 12.0.0 and Im going to use it only as a client. Im trying to SIP REGISTER with a remote SIP provider.The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of ..
I am using Asterisk 11.6.0 built by root @ linux-t784 on a x86_64The issue is a huge UDP handle leak, presumably coming froooh323With 45 calls open calls (ooh323 to SIP), I have netstat -anp | grep asterisk | wc -l6669lsof -p 6785 -i -n -P | grep U..
I saw this thread which is very similar to my issue, though I cannot solve mine with iptables.http://lists.digium.com/pipermail/asterisk-users/2013-September/280429.htmlUsing asterisk 11.5, IAX does not seem to be able to receive any packets. My IP tab..
I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017)and get a SIP 488 Not Acceptable Here response. I have no problems using the same Asterisk configuration and the same page to make a call from Chrome.I have seen other people p..
After struggling with one way audio issues as a result of STUN binding errors on both the Asterisk side and the Chrome side, weve decided to just simply go with a TURN relay for RTP packets until the issues are resolved.I configured rtp.conf so t..