Both sites can receive calls from analog and LOCAL sip trunks without any issue.
Users from site A can contact phones in site B. The opposite is also true.
If I dial SITE A using it's local FXO loop, I can contact AA and transfert to a phone in site B. The opposite is also true.
What doesn't work:
If I dial SITE A using it's local SIP TRUNK , I can contact AA and transfert to a phone in site B. When I do this, The call rings at site B, I see the caller ID, I can pick up the call, but I hear no Audio. The call disconnects after 15 seconds. If I try the opposite (dialing in from Site B's SIP trunk and transfering to a phone in site A), it doesnt work neither
From my understanding, when someone dials into Site A and tries to call someone in Site B, there is two channels involved. One from the sip trunk provider and one from the intersite trunk. So basically, it's like if the Site A was conferencing both channels. Why does it work with analog lines, why does intersite calling works, but not when it's coming from a SIP Trunk ???
I don't understand how this is possible... does anyone have a clue ? I would appreciate any help.
Re: No Audio on intersite transfer but only when using sip trunk
I spoke with TAC today and they found out how to fix the problem.
I honestly cannot explain what the problem was but the tech support guy told me it depends on how the sip trunk provider is configured. He mentionned something about Session border control and nat traversal.
The only lines I had to change were in my dial peers for the intersite trunk. My numbers were 2100 and 2101, yours might be different...
I removed the voice-class h323 on my intersite and it worked fine.
Here is what I typed:
dial-peer voice 2100 voip
no voice-class h323 1
dial-peer voice 2101 voip
no voice-class h323 1
This worked for me, but take care, I don't know if it will work for you (Take a backup before).
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