Harlow Tech Posted May 9, 2015 Report Share Posted May 9, 2015 I am trying to figure out how to configure and add a trunk, dial plan, and contact in the global contacts that will allow me to direct call the Join.me SIP line so that we can avoid tying up our other trunks and take advantage of the HD audio. They have their information post at http://help.join.me/knowledgebase/articles/580113-join-me-sip-access I have already tried setting up a trunk as mentioned at outbound but can not seem to get it to work. Any help would be greatly appreciated. Thank you - Daniel Quote Link to comment Share on other sites More sharing options...
Vodia support Posted May 12, 2015 Report Share Posted May 12, 2015 If I understand correctly, you want to dedicate a particular trunk to use with join.me? and you want to transfer to outside user or do you want to keep it internally? Quote Link to comment Share on other sites More sharing options...
Harlow Tech Posted May 13, 2015 Author Report Share Posted May 13, 2015 I was trying to setup a trunk and dial plan to dial a SIP URI directly, similar to what was described at http://forum.vodia.com/index.php/topic/3215-sip-uri/ Also the join.me link might not be working, not sure what is wrong with their knowledge base tonight (as when I search for, it says 7 found but only shows one article) here is what it had said: join.me allows customers using an IP PBX, IP phones, or softphone to directly connect to join.me conferencing. The SIP connects directly through the internet and bypasses the traditional telephone network. HD Audio If your IP device supports High definition audio codec (G.722) you can take advantage of HD Audio quality. This codec is usually incorporated into newer IP phones and conference room equipment. SIP Addresses 12132261066@audio1.join.me (non-encrypted SIP access) No registration required Your nine digit conference ID and four digit PIN act as your authentication Embedding the Conference IDTo bypass the welcome greeting and prompts, use the format below. sip:logmein#XXXXXXXXX*YYYY@audio1.join.me (non-encrypted SIP access) XXXXXXXX= 9 digit conference ID YYYY= 4 digit pin code Example:sip:logmein#123456789*1234@audio1.join.me Other supported standardsjoin.me supports various standards: RFC 2833 standard for in-band DTMF NAT traversal, STUN (will automatically attempt to configure a quality connection with your device or client) Standard codecs G.711 and G.722 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 13, 2015 Report Share Posted May 13, 2015 That looks pretty cool. Did you already try to get a PCAP trace for this? Is the join.me server responding at all? Quote Link to comment Share on other sites More sharing options...
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