QUESTION Caller Not Getting My 200 OK

EcstaticMark

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I'm losing sleep over this, so here goes. Short version: I can't answer inbound calls to my trunk's DIDs, but CAN answer calls on the same trunk to my ported DID.

Asterisk Ver. 13.6.0, Incredible PBX 12.0.74.
1 Digium Trunk with 2 lines. 234-521-xxx1 and 234-521-xxx2.
One 1 line, my home number was ported over so 330-208-xxxx points to 234-521-xxx1.
Calls to and from my ported home number (330-208-xxxx) work fine.
Calls to either of the original DIDs (234-521-xxxx) ring my phone but when answered the calling phone keeps ringing. There is no audio path since the calling phone is still ringing.

I'm not sure when the problem started. I've been using this as my home phone system and test lab (I know, I know...) for a few months.

I see the following in a SIP trace, followed by '[2016-09-29 20:38:52] WARNING[24815]: chan_sip.c:4009 retrans_pkt: Retransmission timeout reached on transmission 03c667a0-0149-1235-e592-002590e549c4 for seqno 97261883 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
Really destroying SIP dialog '03c667a0-0149-1235-e592-002590e549c4' Method: ACK
Really destroying SIP dialog '[email protected]:5060' Method: BYE

Retransmitting #10 (NAT) to 8.17.32.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 8.17.32.12:5060;branch=z9hG4bK85ff.3fe5abfb57124cd25ee5b5c47b560dde.0;received=8.17.32.12;rport=5060
Via: SIP/2.0/UDP 8.17.32.37;rport=5060;branch=z9hG4bK85ff.bd0cb542.0
Via: SIP/2.0/UDP 64.24.35.216;branch=z9hG4bK85ff.742f434872cf4f37c852d5ec15aadbfa.0
Via: SIP/2.0/UDP 64.24.35.79;rport=5060;branch=z9hG4bKQQ1H66gtr4ejQ
Record-Route: <sip:8.17.32.12;lr=on;ftag=XyUQ3Kv2tjFKD;did=d1d.2a02>
Record-Route: <sip:8.17.32.37;lr=on;did=3d.be2e2284>
Record-Route: <sip:64.24.35.216;lr=on;ftag=XyUQ3Kv2tjFKD>
From: "WIRELESS CALLER" <sip:[email protected]>;tag=XyUQ3Kv2tjFKD
To: <sip:[email protected]:5060>;tag=as1dcc0d36
Call-ID: 03c667a0-0149-1235-e592-002590e549c4
CSeq: 97261883 INVITE
Server: FPBX-12.0.74(13.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
P-Asserted-Identity: "Main Desk 1" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 477

v=0
o=root 1798961617 1798961617 IN IP4 69.81.140.7
s=Asterisk PBX 13.6.0
c=IN IP4 69.81.140.7
t=0 0
m=audio 10692 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:7bf183cf46c1fcdb0471d412461ace69
a=ice-pwd:142d2cf6583b102e1d83976850cf0ea2
a=candidate:Hc0a801e6 1 UDP 2130706431 192.168.1.230 10692 typ host
a=candidate:Hc0a801e6 2 UDP 2130706430 192.168.1.230 10693 typ host
a=sendrecv

I've tried changing various NAT-related settings in Asterisk and putting the Asterisk server in my router's DMZ. I don't see any packet drops from IPs in the SIP conversation in my firewall's logs.

Digium support is pretty sure the problem is on my end. I've tried Googling but I'm not even sure what to search for.

Is this a known problem? Can anyone point me in the general direction I should be focusing? Should I find another hobby? I can provide any logs or configuration files needed.

Thanks for your time and any advice.

Edit 9/30: Nevermind, I seem to have figured it out. I had ICE enabled in /etc/asterisk/sip_general_custom.conf. Had I ever looked in Settings, Asterisk SIP Settings, chan_sip I might have noticed the subtle BIG RED LETTERS saying not to put stuff in that file.

Shame on me!

Mods, if you want to delete this post, fine. Or you can leave it as a warning to those who follow.
 
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