FreePBX not replying to OPTIONS ping

FreePBX
1

We use an ISP called Bright House Networks here in central florida to provide us with SIP trunks for our PBX. Outgoing calls are working like a champ, but incoming calls ring twice and then end with an “all circuits are busy” message.

I attempted to get support from Bright House and apparently they continously send our pbx an “OPTIONS” ping to which our pbx needs to reply to in order to kep the connection alive. Our pbx is not responding to this ping according to them. I have tried to google my way to success and look through every option that freepbx has in the gui, but to no avail. I turned off all firewalls, and the PBX is connected directly to a SIP switch (Audiocodes Mediant 800) which they provided us with which connects to a dedicated fiber line.

Please help.

2

Here is some debug output.

<— Transmitting (NAT) to 97.69.197.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 97.69.197.193:5060;branch=z9hG4bK00olaq1080sgih0f65f0;received=97.69.197.193;rport=5060
From: sip:[email protected];tag=vck3800sr3200-unh1000
To: sip:[email protected];tag=as1c63f82f
Call-ID: [email protected]97.193
CSeq: 50942 OPTIONS
Server: FPBX-AsteriskNOW-12.0.37(11.14.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:192.168.0.72:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]97.193’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘[email protected]97.193’ Method: OPTIONS
[2015-02-13 10:23:13] NOTICE[23701]: chan_sip.c:16428 check_auth: Correct auth, but based on stale nonce received from ‘"Brian"sip:[email protected]:5060;tag=2c19ee5f’
[2015-02-13 10:23:25] WARNING[23701]: chan_sip.c:4019 retrans_pkt: Retransmission timeout reached on transmission YzgwZDcyZDhhMjMzMWZlZWE2NjEwOWNlZDFjOTIyNmU. for seqno 134 (Non-critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10688ms with no response

<— SIP read from UDP:97.69.197.193:5060 —>
OPTIONS sip:192.168.0.72:5060 SIP/2.0
Via: SIP/2.0/UDP 97.69.197.193:5060;branch=z9hG4bK00olao208o01bhc705f0
To: sip:[email protected]
From: sip:[email protected];tag=vck3800sr3200-vnh1000
Call-ID: [email protected]97.193
CSeq: 50943 OPTIONS
Max-Forwards: 1
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Sending to 97.69.197.193:5060 (NAT)
Looking for s in from-sip-external (domain 192.168.0.72)

<— Transmitting (NAT) to 97.69.197.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 97.69.197.193:5060;branch=z9hG4bK00olao208o01bhc705f0;received=97.69.197.193;rport=5060
From: sip:[email protected];tag=vck3800sr3200-vnh1000
To: sip:[email protected];tag=as2ee83488
Call-ID: [email protected]97.193
CSeq: 50943 OPTIONS
Server: FPBX-AsteriskNOW-12.0.37(11.14.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:192.168.0.72:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]97.193’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘[email protected]97.193’ Method: OPTIONS

<— SIP read from UDP:97.69.197.193:5060 —>
OPTIONS sip:192.168.0.72:5060 SIP/2.0
Via: SIP/2.0/UDP 97.69.197.193:5060;branch=z9hG4bK00olam308o01bismu190
To: sip:[email protected]
From: sip:[email protected];tag=vck3800sr3200-0oh1000
Call-ID: [email protected]97.193
CSeq: 50944 OPTIONS
Max-Forwards: 1
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Sending to 97.69.197.193:5060 (NAT)
Looking for s in from-sip-external (domain 192.168.0.72)

<— Transmitting (NAT) to 97.69.197.193:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 97.69.197.193:5060;branch=z9hG4bK00olam308o01bismu190;received=97.69.197.193;rport=5060
From: sip:[email protected];tag=vck3800sr3200-0oh1000
To: sip:[email protected];tag=as0110b4f3
Call-ID: [email protected]97.193
CSeq: 50944 OPTIONS
Server: FPBX-AsteriskNOW-12.0.37(11.14.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:192.168.0.72:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]97.193’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘[email protected]97.193’ Method: OPTIONS
[2015-02-13 10:25:02] NOTICE[23701]: chan_sip.c:16428 check_auth: Correct auth, but based on stale nonce received from ‘"Brian"sip:[email protected]:5060;tag=2c19ee5f’
[2015-02-13 10:25:11] WARNING[23701]: chan_sip.c:4019 retrans_pkt: Retransmission timeout reached on transmission YzgwZDcyZDhhMjMzMWZlZWE2NjEwOWNlZDFjOTIyNmU. for seqno 136 (Non-critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 7936ms with no response

3

Did you ever figure this out?

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