Yealink T58V and Video Calling
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@bigbear said in Yealink T58V and Video Calling:
I had some very old yealink video phones that have never worked on FreePBX. I am not doing any provisioning though, just entering sip server ip, user, auth information.
Weird, any idea why they did not work?
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I just assumed it wasnt supported by default. Gonna spin up a fresh install of FreePBX and try from scratch.
So you think they should just work by default?
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@bigbear said in Yealink T58V and Video Calling:
I just assumed it wasnt supported by default. Gonna spin up a fresh install of FreePBX and try from scratch.
So you think they should just work by default?
I would expect so.
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@bigbear I would expect it to, yes.
I have not had any clients ask for video phones, so I have not tested this.
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@JaredBusch said in Yealink T58V and Video Calling:
@bigbear I would expect it to, yes.
I have not had any clients ask for video phones, so I have not tested this.
I figured out I could pay $23 more on bulk orders and get the T58V over the 58A. The unfortunate side effect of this deal is that those customers are now asking about it. I told them its not supported but then thought I would give it a try.
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So the issue appears to be with FreePBX's GUI not offering video codec settings in the web interface for PJSIP.
I was able to set it on each extension as per the comments in this bug report...
https://issues.freepbx.org/browse/FREEPBX-11577
And you have to set it as: h264,ulaw,alaw under allowed codes under Advanced in each setting.
Now I am trying to figure out if I can get phones on the same remote LAN to send RTP streams direct (phone to phone) so that the video doesn't freeze up. It's really more of a gimick I think (the video calling). Seems like it would be annoying to have people peeking in at you over intercom all day.
But still, I would like to figure out how to get the direct RTP stream going. This can be done in Freeswitch so I imagine Asterisk will do it. Not sure the GUI would have a setting for it though.
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@bigbear said in Yealink T58V and Video Calling:
But still, I would like to figure out how to get the direct RTP stream going. This can be done in Freeswitch so I imagine Asterisk will do it. Not sure the GUI would have a setting for it though.
Direct meaning peer to peer rather than passing through Asterisk?
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Yes, directmedia I suppose.
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Will give it a try tomorrow. With it being hosted I wonder if I need to do the directmedia=yes entry.
Will report back whatever configs work...
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http://doxygen.asterisk.org/trunk/sip.conf.html
Skip down to "Media Handling"
