I'm having an issue with all my calls going out my SIP provider. I'm using Jitsi as a softphone registering to a local Asterisk PBX.
I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider.
The call is correctly established and conversation is OK. IF the Jitsi user hangs up first, the remote end is also disconnected immediately.
However, if the remote party hangs up first, the jitsi user is not immediately disconnected.
That, of course, is undesirable.
I'd like to understand why the call isn't automatically hung up and fix it.
I'm supposing that Jitsi isn't receiving a BYE as expected in a correct SIP transaction.
I don't know why though.
Here's my network setup:
Jitsi asterisk extension 4053 at 10.215.144.48
Asterisk eth0: 10.215.147.111 but Jitsi registers to the alias/floating IP for failover setup 10.215.147.115
Asterisk eth1: 192.168.103.111
Asterisk default gateway: 192.168.103.1
-> Asterisk accesses Internet via eth1 (192.168.103.1 is a DSL modem/router)
I did a tcpdump on the asterisk server while calling from Jitsi as so:
tcpdump -s0 -X -n -w jitsi_via_asterisk.cap -i eth0 host 10.215.144.48
I also dumped the Jitsi log files here:
Jitsi Asterisk Sip
Check out Jitsi as a Service. Connect the users of your website or app. Get branding & tight access controls. Have notifications, transcriptions & recordings delivered straight to your backend Learn more Start & join meetings for free No account needed. Connect your calendar to view all your meetings in Jitsi. Jitsi is a simple to configure, simple to use, multi-platform softphone with many useful features. This softphone has been tested and shown to be stable in Windows, Linux and OSX. Although the software supports many other communication methods we will specifically be. All software has been installed in the last few days with the last versions (jitsi 2.4.4997 and 2.5.5065 on Ubuntu, jitsi 2.4.4997 on windows), Asterisk 11.12 compiled from source). I tried with and without firewall results are the same. Both Ubuntu and windoz computers are on the same network. Could it be a problem with an external library?
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Jitsi Asterisk Setup
- Setup and configuration of Jigasi - Jitsi’s SIP Gateway element for connecting to SIP telephony.
- Some SIP servers (Asterisk in particular) do not announce the MESSAGE support, despite supporting it. If you enable the account property FORCEMESSAGING, Jitsi will attempt to use MESSAGE for chats, despite your configured SIP server not explicitly announcing this support to connected clients.