Hey all,
Trying to figure out this issue with my Lync 2013 and Asterisk environment. I realize that this is not a support topology but this is what I have:
Lync clients (internal and external) can receive inbound calls correctly.
However, on outbound calls to PSTN, there is no audio from the callee until the Lync client makes a noise first. The callee does not hear anything either.
However, when the Lync user makes a small sound, RTP traffic starts to flow through Asterisk and the call proceeds normally. But until there is a sound from the Lync client, the audio from both sides is simply not there. No audio at all on either side. It is the Lync user that has to make a noise in order for RTP traffic to stop. The callee can make noise all they want but the RTP doesn’t start. Has to be from Lync.
This is causing us issues obviously.
When I do a Wireshark capture between my client and the front end server, you see the RTCP traffic traversing to the Asterisk box but then stops as RTCP Sender Report packets. This corresponds to the muted audio. WHen I make a sound on the Lync side, you immediately start seeing RTP flow to Asterisk, corresponding to the call completing as normal.
On the Asterisk monitor, I show the following in the call setup:
== Using SIP RTP ToS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/QuestBlue-SBC/12102263232
– SIP/QuestBlue-SBC-00000005d is making progress passing it to SIP/Lync-0000005c
– SIP/QuestBlue-SBC-00000005d answered SIP/Lync-0000005c
And then it halts until the said needed noise comes through… then immediately displays
[b] > 0x24eea90 -- Probation passed - setting RTP source address to 172.20.1.11:11724
> 0x7f10380a5c40 -- Probation passed - setting RTP source address to 4.55.42.130:9340[/b]
And the call proceeds as normal. The IP addresses above, the 172 is my internal front end server, and it looks like the 4.55 address is the address of the top level PSTN provider, Level 3, beyond the SBC IP that my Asterisk trunk connects to.
I opened up all firewall rules to these addresses. I messed with NAT settings, directmedia, prematuremedia, strictRTP options, but can’t seem to handle whats going on.
Lync Front End and Asterisk machine on same subnet with direct internal IP addressing. Asterisk connects to an SBC controller with my SIP provider behind a NAT.
Lync Trunk Settings:
host=172.20.1.11
port=5060
nat=no
canreinvite=no
fromdomain=172.20.1.90
qualify=no
type=peer
transport=tcp
dtmfmode=rfc2833
context=from-internal
disallow=all
allow=ulaw&alaw
directmedia=yes
prematuremedia=yes
USER Incoming settings blank.
Lync server updated to 5.0.8308.556.
Asterisk 11.10.2
FreePBX version 2.11.0.41
Any ideas?