vicidial.org

So got a small setup with ViciBox v.7.0.4-170113 , running VERSION: 2.14-620a BUILD: 170623-2142 on a single server. Two carriers setup one for outbound one for inbound. Worked great for a few weeks, and then starting yesterday the outbound carrier became unreachable. Checked with the carrier (SIP.US) and they say they see no traffic from our IP, and our ISP sees no issues. The inbound carrier is chugging along like normal no issues. I can ping the trunk IPs from the dialer no problem, and a traceroute returns everything okay. They can ping my router no problem also. The setup on the router is real simple, just port forwarding 5060, and 10000:20000 to the dialer, and it had been working fine up until yesterday. There has been no changes on the network or dialer for a week or two. I'm concerned it may be an issue on the router itself. (Cisco Meraki MX84) But it's brand new, and we have another location with one that doesn't have any similar issue, but I've opened a ticket with them regardless, haven't heard back yet though, wanted to check here and see if you guys had any thoughts of other things I can look into.

Here's a sip debug of the carrier when doing a sip reload:

Code: Select all

[Jul 21 16:21:03] Reliably Transmitting (NAT) to 65.254.44.194:5060:
[Jul 21 16:21:03] OPTIONS sip:gw1.sip.us SIP/2.0
[Jul 21 16:21:03] Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK67415a7b;rport
[Jul 21 16:21:03] Max-Forwards: 70
[Jul 21 16:21:03] From: "asterisk" <sip:asterisk@66.xxx.xxx.xxx>;tag=as3c904525
[Jul 21 16:21:03] To: <sip:gw1.sip.us>
[Jul 21 16:21:03] Contact: <sip:asterisk@66.xxx.xxx.xxx:5060>
[Jul 21 16:21:03] Call-ID: 3dd65f156bf4c31841446799119a12b4@66.xxx.xxx.xxx:5060
[Jul 21 16:21:03] CSeq: 102 OPTIONS
[Jul 21 16:21:03] User-Agent: Asterisk PBX 11.25.1-vici
[Jul 21 16:21:03] Date: Fri, 21 Jul 2017 20:21:03 GMT
[Jul 21 16:21:03] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 21 16:21:03] Supported: replaces
[Jul 21 16:21:03] Content-Length: 0
[Jul 21 16:21:03]
[Jul 21 16:21:03]
[Jul 21 16:21:03] ---
[Jul 21 16:21:04] Retransmitting #1 (NAT) to 65.254.44.194:5060:
[Jul 21 16:21:04] OPTIONS sip:gw1.sip.us SIP/2.0
[Jul 21 16:21:04] Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK67415a7b;rport
[Jul 21 16:21:04] Max-Forwards: 70
[Jul 21 16:21:04] From: "asterisk" <sip:asterisk@66.xxx.xxx.xxx>;tag=as3c904525
[Jul 21 16:21:04] To: <sip:gw1.sip.us>
[Jul 21 16:21:04] Contact: <sip:asterisk@66.xxx.xxx.xxx:5060>
[Jul 21 16:21:04] Call-ID: 3dd65f156bf4c31841446799119a12b4@66.xxx.xxx.xxx:5060
[Jul 21 16:21:04] CSeq: 102 OPTIONS
[Jul 21 16:21:04] User-Agent: Asterisk PBX 11.25.1-vici
[Jul 21 16:21:04] Date: Fri, 21 Jul 2017 20:21:03 GMT
[Jul 21 16:21:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 21 16:21:04] Supported: replaces
[Jul 21 16:21:04] Content-Length: 0
[Jul 21 16:21:04]
[Jul 21 16:21:04]
[Jul 21 16:21:04] ---
[Jul 21 16:21:05] Retransmitting #2 (NAT) to 65.254.44.194:5060:
[Jul 21 16:21:05] OPTIONS sip:gw1.sip.us SIP/2.0
[Jul 21 16:21:05] Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK67415a7b;rport
[Jul 21 16:21:05] Max-Forwards: 70
[Jul 21 16:21:05] From: "asterisk" <sip:asterisk@66.xxx.xxx.xxx>;tag=as3c904525
[Jul 21 16:21:05] To: <sip:gw1.sip.us>
[Jul 21 16:21:05] Contact: <sip:asterisk@66.xxx.xxx.xxx:5060>
[Jul 21 16:21:05] Call-ID: 3dd65f156bf4c31841446799119a12b4@66.xxx.xxx.xxx:5060
[Jul 21 16:21:05] CSeq: 102 OPTIONS
[Jul 21 16:21:05] User-Agent: Asterisk PBX 11.25.1-vici
[Jul 21 16:21:05] Date: Fri, 21 Jul 2017 20:21:03 GMT
[Jul 21 16:21:05] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 21 16:21:05] Supported: replaces
[Jul 21 16:21:05] Content-Length: 0
[Jul 21 16:21:05]
[Jul 21 16:21:05]
[Jul 21 16:21:05] ---
[Jul 21 16:21:06] Retransmitting #3 (NAT) to 65.254.44.194:5060:
[Jul 21 16:21:06] OPTIONS sip:gw1.sip.us SIP/2.0
[Jul 21 16:21:06] Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK67415a7b;rport
[Jul 21 16:21:06] Max-Forwards: 70
[Jul 21 16:21:06] From: "asterisk" <sip:asterisk@66.xxx.xxx.xxx>;tag=as3c904525
[Jul 21 16:21:06] To: <sip:gw1.sip.us>
[Jul 21 16:21:06] Contact: <sip:asterisk@66.xxx.xxx.xxx:5060>
[Jul 21 16:21:06] Call-ID: 3dd65f156bf4c31841446799119a12b4@66.xxx.xxx.xxx:5060
[Jul 21 16:21:06] CSeq: 102 OPTIONS
[Jul 21 16:21:06] User-Agent: Asterisk PBX 11.25.1-vici
[Jul 21 16:21:06] Date: Fri, 21 Jul 2017 20:21:03 GMT
[Jul 21 16:21:06] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 21 16:21:06] Supported: replaces
[Jul 21 16:21:06] Content-Length: 0
[Jul 21 16:21:06]
[Jul 21 16:21:06]
[Jul 21 16:21:06] ---
[Jul 21 16:21:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul 21 16:21:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul 21 16:21:07] Retransmitting #4 (NAT) to 65.254.44.194:5060:
[Jul 21 16:21:07] OPTIONS sip:gw1.sip.us SIP/2.0
[Jul 21 16:21:07] Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK67415a7b;rport
[Jul 21 16:21:07] Max-Forwards: 70
[Jul 21 16:21:07] From: "asterisk" <sip:asterisk@66.xxx.xxx.xxx>;tag=as3c904525
[Jul 21 16:21:07] To: <sip:gw1.sip.us>
[Jul 21 16:21:07] Contact: <sip:asterisk@66.xxx.xxx.xxx:5060>
[Jul 21 16:21:07] Call-ID: 3dd65f156bf4c31841446799119a12b4@66.xxx.xxx.xxx:5060
[Jul 21 16:21:07] CSeq: 102 OPTIONS
[Jul 21 16:21:07] User-Agent: Asterisk PBX 11.25.1-vici
[Jul 21 16:21:07] Date: Fri, 21 Jul 2017 20:21:03 GMT
[Jul 21 16:21:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 21 16:21:07] Supported: replaces
[Jul 21 16:21:07] Content-Length: 0
[Jul 21 16:21:07]
[Jul 21 16:21:07]
[Jul 21 16:21:07] ---
[Jul 21 16:21:07] Really destroying SIP dialog '3dd65f156bf4c31841446799119a12b4@66.xxx.xxx.xxx:5060' Method: OPTIONS

And here is the carrier config:

Code: Select all

Registration String:   52xxxxxx:xxxxxxx@gw1.sip.us

Account Entry:
[52xxxxxxGW1]
type=peer
insecure=port,invite
host=gw1.sip.us
port=5060
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
qualify=yes
qualifyfreq=30
nat=force_rport,comedia
trustrpid=yes
fromdomain=gw1.sip.us
username=52xxxxxx
secret=xxxxxxxx
context=default
rfc2833compensate=yes
session-timers=refuse

Global String: SIPUS = SIP/52xxxxxxxGW1

Dial Plan:
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${SIPUS}/1${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup


Yep looks like it was due to a firmware upgrade on the router. Will have to wait until after-hours tonight to test it after the roll back but looks like there was an upgrade early this morning. Dang these fancy cloud enabled routers, doing stuff all by themselves. To be fair they sent me an email a 10 days ago saying it was going to happen. I guess it is on me for not checking it after the maintenance window.

Looks like the roll back didn't fix it. Anyone have any other ideas of what I can look into?

IP address change? Have you tried pinging your carrier from your server? perhaps your carrier changed whitelisted IPs? also get rid of your call id in your code... that looks like a registration string