Puzzled by NAT

@Thucar wrote:

Hi all, I'm setting up a PBX for our company with the main purpose of learning more about SIP/PBX setups.

We use VoIP phones in our company. I took one of the numbers and used it as a trunk for my setup. After some fiddling I managed to get it to work with one small weird issue. I have the IP configuration set to "Static IP." I can make calls out the trunk with no problems but when trying to receive calls, I'm getting the Telco announcement "The number is not in use." My SIP trunk shows up as Registered at the same time.

Now when I switch the IP configuration to "Public IP" I can make and receive calls. BUT, on incoming calls I have no outgoing audio. i.e the person calling via telco, can not hear me.

This is where it gets interesting. When I switch back to "Static IP" after using "Static IP" then absolutely everything works (Incoming calls, outgoing calls, audio both ways) for about 5 minutes. After that I'm getting the "Number is not in use" announcement again - while "sip show peers", "sip show registry" both show the trunk as OK/Registered

My sip settings:

sip*CLI> sip show settings

Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: No
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-14.0.1.9(13.15.0)
SDP Session Name: Asterisk PBX 13.15.0
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: Disabled
Externhost:
Externaddr: (null)
Externrefresh: 10
Localnet: 192.168.1.0/255.255.255.0

Global Signalling Settings:

Codecs: (alaw|ulaw|g722|h263|h264|h263p|vp8|mpeg4)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
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: Yes
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:No
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: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: en
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
RTCP Multiplexing: No

Posts: 4

Participants: 1

Read full topic