On 4 July 2017 at 23:55, Mark Melling <mark.melling@savageminds.com> wrote: Hi,
I have a dialplan where a user calls a number which results in them being placed in a conference room, and an outbound called made to a verto client when the conference is created, by defining in the dialplan:
<action application="conference_set_auto_outcall" data="user/1000@ $${domain}"/>
The verto client (which is basically taken from the verto demo) rings and is answered.
The original inbound caller can hear the verto client speaking, but the verto client can't hear the the inbound caller (although the audio from the inbound caller is getting to the conference room).
If I change the dialling order - so the verto client makes the inbound call and the SIP client is dialled from the conference using the same technique as before (conference_set_auto_outcall). Then everything appears to work fine, i.e. each caller can here the other.
Any suggestions as to what might be the issue or steps I could take to help identify the problem.
Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc Try this: instead of generating autocall from inside conference (eg instead of using autocall),originate call to user, wait for her to answer, then (after she answer) sleep for 2 seconds, then transfer her to the conf -giovanni
Thanks
Mark
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