Are you sure about
?
That’s from the freepbx gui after detecting external address. ifconfig does not show this.
Done. System is responding the same way as before. Asterisk never sees the inbound trunk. It is not until if fails over to the 2nd trunk that the system responds at all.
I don’t believe it will detect your localnet(s) just your “externaladdress” you need to do that yourself if you are NATting generally.
grep ip addr|grep inet
will get you on the right track.
Maybe you just need to define a public IP and no NAT if you are indeed such a beast.
Hi Mike,
Send again a log of an inbound call. Did you change the NAT settings?
Thank you,
Daniel Friedman
Trixton LTD.
I think I NAT okay. Here are the SIP settings.
accounting-services*CLI> sip show settings
Global Settings:
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-12.0.74(13.5.0)
SDP Session Name: Asterisk PBX 13.5.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
SIP address remapping: Enabled using externaddr
Externhost:
Externaddr: 88.214.255.66:0
Externrefresh: 10
Localnet: 88.214.255.64/255.255.255.224
Global Signalling Settings:
Codecs: (ulaw|alaw|gsm|g726)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: No
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language:
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
I’ll email the capture.
Thanks!
Hi,
- The NAT is still on static. It should be public. Fix it.
- you do not have an inbound route to your DID and that is why it goes to the Any route.
- There is a cli source lookup on your inbound trunks (cid superfecta). Remove it because it causes a delay.
Please report back.
Thank you,
Daniel Friedman
Trixton LTD.
Hi
try to set qualify=yes to qualify=no
@DanielF, I’m having this same problem with my freepbx hosted at cyberlink, would you mind going through some of this with me later today?
Greetings if you managed to configure the incoming calls correctly you could send me the configuration by mail thanks
[email protected]