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Jitsi VideoBridg and Asterisk server SIP Registration failed

Hi all,

I have an issue with connecting the Jitsi VideoBridg to my private Asterisk server. The problem seems to be that the Jitsi VideoBridg wants to connect to the sip server at port 5070, however Asterisk is setup to use either port 5060 5061 for sip. This is resulting in a connection failed worng password error from Asterisk. If i use the Asterisk im plugin and use port 5070 it also can not connect however if i use port 5060 it connects.

How can I change the Jitsi VideoBridg SIP port from 5070 to 5060/5061.


change or add the openfire property


Hi Dele.

this hint was very helpful for me. Jitsi videobridge now connects to asterisk via new extension (200). When i call another SIP extension from ofmeet videobridge, it doesnt use the registered extension (200) to make a call to another audio-only participant (say ext 105). Instead, asterisk see that call as an external call (from external trunk) and not from internal extension.

Did i misunderstood something, i thought that the SIP registration feature is for calling audio-only participants from ofmeet?


That worked. Now how does the jitsi videobridge do phone mappings? I tried using Asterisk-IM Openfire Plugin however the users are not appearing online (in openfire under User Summary) when they connect.

How can i set up phone mappings for the jitsi videobridge and have users appear online when they are connected?

use tel:nnnnn instead of sip:nnnnn@domain.com

When you use tel:nnnnn, it will use the SIP registration as the outbound proxy. When you use sip:xxxxxx@domain.com, it will send a direct SIP invite to domain.com

There is no connection or relationship between jitsi videobridge and asterisk-im plugin. It is what it says it is. A videobridge for multi-user video/audio calls. Not a phone or PBX. It simply uses SIP to enable phone users join or be invited to existing video conferences.

thanks Dele, now it really uses the SIP registration.

unfortunately, after answering, the call is dropped after few seconds (5-10).

Im experiencing the same efect. When i use tel:number it dials via sip account configured in jitsi settings but when call is established, i can hear voice of phone user in conference but phone user has silence on his side. Call is dropped after 5-10 seconds.