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I am working with ubuntu and asterisk 1.8. I've set the configuration for conference calls:

in extensions.conf

exten => 2115,1,Answer()
exten => 2115,n,Set(CHANNEL(language)=gb-f)
exten => 2115,n,Set(CHANNEL(musicclass)=default)
exten => 2115,n,ConfBridge(1234,Mcs,123)

Is there an android application with whom to test this conference channel? I would like to test it between 3 sip clients. How to do this? Are my settings correct?

1234 is the conference room also set in Meetme.conf

THX appreciate

user1222905
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  • do you have 3 devices? than you can use SipDroid, CSIpSimple, IMSDroid. You can also use software clients - Ekiga etc. – hovanessyan Mar 12 '12 at 14:39
  • can i test with imsdroid on all 3 devices? how to do tthat? i've made a simple call to 2115 with all 3 devices using imsdroid. the video stops. audio works. when client 3 ends the call both audio and video works. – user1222905 Mar 12 '12 at 14:46
  • if the video and audio works in some cases, that probably means your Asterisk configuration is OK, and you may have problems in your SDP offers. Also research the capabilities for video conferencing on Asteriks. I know the audio mixer should work, but I am not sure how it handles video conferencing. Make note of the video codecs you use on all devices...this could also be an issue. – hovanessyan Mar 12 '12 at 14:55
  • the video codecs are ok. The sdp i guess it's correct bcause it's cone by ConfBridge. I don't understand why on ekiga there are no 2 other video windows with each device from the conference call. have any ideas? – user1222905 Mar 12 '12 at 15:20
  • nope sorry, my knowledge in conferencing extends to audio calls. – hovanessyan Mar 12 '12 at 15:30
  • as i've seens the communications are between the server and each client:) ...is it correct? – user1222905 Mar 12 '12 at 15:41
  • i didn't set 2115 in sip.conf. Do I have to set it there too? – user1222905 Mar 12 '12 at 15:46
  • yes - the communication is between the server and each client. In the case of conference-audio call, the audio mixer on the server mixes the rtp streams from all clients into 1 rtp stream and sends it to 1 client, and it does that for all clients. – hovanessyan Mar 12 '12 at 16:41
  • no, you don't need 2115 into sip.conf – hovanessyan Mar 12 '12 at 16:46

1 Answers1

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I already say you here

Asterisk support only ONE way video conference.

If you are expecting get mixed conference stream, see vmuktu or other solution.

arheops
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  • can you tell me how to install vmuktu on ubuntu and how to use it with asterisk? – user1222905 Mar 13 '12 at 08:20
  • I 've understood that asterisk is one way video conference, but when a 3 sip clients comes in the conference the video is freeze on all phones. No video at all. why?1 – user1222905 Mar 13 '12 at 09:42
  • No. i can't "tell" you how to install vmuktu.Please consult project page. Metmee video work only if 1) all codecs SAME 2) only one set to "speeking" mode. all other is muted off. – arheops Mar 13 '12 at 12:45