11-20-2019 08:17 AM
Hello,
I've tried to join a meeting which I've planned in a certain space on my free tier WebEx team.
The SIP URI showed up on this screen:
In order to join this meeting I've used MicroSIP as ultra-leightweight SIP softphone (with a configured SIP account of an external service provider which means: no CUCM). The problem in that case that I don't receive any A or SRV records if I query this subdomain and that's why my softphone cannot connect the destination SIP host.
Is it even possible to use generic SIP devices in a non-CUCM environment and what should I try next if it is possible in theory?
Best regards
Elysweyr
11-21-2019 07:22 AM
11-23-2019 09:09 AM
Hi Anthony,
thanks for your reply.
I asked my SIP provider and they said that they don't support SIP URI dialing for other SIP hosts since 2013 which means that I need to get my Asterisk server up and running now until I can test this setup so you'll get an update in a little while.
Best regards
Elysweyr
12-15-2019 03:27 PM - edited 12-15-2019 03:28 PM
I haven't forgot this post here.
I'm just having troubles switching the transport protocol to TLS on my asterisk pbx (asterisk <-> external SIP is using UDP; local extension <-> asterisk is fine).
Generally I'm working on this for some days now and have no clue whether I'll manage this or not.
Let's see how this ends.
12-19-2019 03:28 AM
Do you have an idea how I can set up a trunk to meetup.webex.com (authentication - user/password, ...)?
That's the only way how I can force this to TLS - do you have any clue about this topic?
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