23 Mar
2018
23 Mar
'18
4:33 a.m.
OMG, what are the odds, a client reported the same problem today! Edge
proxy running same 4.2.3, requests are forwarded to a farm of Asterisks v13
in a similar way based on $rd, far-end NAT traversal is handled by Kamailio.
I've had only an hour or so to debug today. Re-invites containing SDP are
handled the same way as invites in terms of SDP mangling, all looks good in
that sense. There's nothing special to be done about re-invites.
Preliminary clue is that this happens (or not) depending on the type of
firewall/NAT behind which the phone is located. In the case with the
trouble, it's a Sonicwall, probably a Symmetric NAT. Is doesn't happen to a
phone behind a Full/Restricted Cone NAT.
What nat= are you setting for Asterisk peers?
Do you engage rtpproxy/rtpengine?
Any far-end NAT traversal manipulations involved such as SIP ALG or STUN?
Cheers.
On Thu, Mar 22, 2018 at 3:55 PM, gerry kernan <gerry.kernan(a)infinityit.ie>
wrote:
Hi
Hoping someone can point me in the right direction.
I have a Kamailio Ver: 4.2.3-1.1 running in front of a few asterisk
servers Ver: 13.17.2 sip is routed to an asterisk server depending the
domain name in the sip request, all working as expected . but if a call is
put on hold after resuming the call the party that placed the call on hold
can’t hear any audio. The other party can hear . do I need to do anything
special to handle re-invites for calls put on hold?
*Gerry Kernan*
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