Hi
Thanks in advance if anyone can point me in
the correct direction .
I have kamailio running in front of some
asterisk VM’s. SIP endpoint register to their asterisk PBX
via Kamailio dispatcher module. I’m running rtpengine with a
Wan and private interface to bridge audio stream between these
endpoints on the WAN to asterisk PBX running on LAN IP behind
Kamailio.
Calls from ext to ext work fine audio both
directions , calls outbound to PSTN via SIP trunk to SIP
provider via trunk on asterisk work fine audio both
directions. But incoming calls via SIP provider I only get
audio on stream from asterisk registered ext to external
caller , no audio from external caller to the asterisk ext.
I reckon I have something wrong in my
Kamailio.cfg . if I register an ext direct to asterisk I get
audio both ways on incoming calls. And rtp logs from
rtpenegine show it as trying to send the rtp to the private
address of the sip endpoint rather that its WAN address.
I think my mistake in somewhere in the cfg
below.
Do I need to handle invites from the
backend asterisk servers different that invites from sip
endpoints?
Gerry Kernan
Infinity
IT | 17 The Mall | Beacon Court | Sandyford
| Dublin D18 E3C8 | Ireland
Tel:
+353 - (0)1 - 293 0090 | E-Mail: gerry.kernan@infinityit.ie
Managed IT
Services Infinity IT -
www.infinityit.ie
IP
Telephony Asterisk Consulting
– www.asteriskconsulting.com
Contact
Centre Total Interact
– www.totalinteract.com
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