SIP Trunk Inbound getting "The number you have call is out of Service"

Hello new here I have Inbound Route/ and SIP Trunk that receives several DID and they are routing to our main IVR of the company but some call are getting “The number you have call is out of Service” ( Intermittent ) below is the configuration of the Inbound route and SIP Trunk

Inbound Route:
Description: Main Corp IVR
DID Number: 1XXXXXXXXXX
CallerID Number:
CID Priority Route:

Alert Info:
CID name prefix:
Music On Hold: PM-BILAT
Signal RINGING:
Pause Before Answer:

Privacy Manager: No
Source: None

Language: None
Set Destination
IVR : Main I VR

SIP Trunk
Trunk Name: NAME
Outbound CallerID: blank
CID Options: Allow my CID
Maximum Channels: Blank
Disable Trunk: Uncheck
Monitor Trunk Failures: Blank

Dialed Number Manipulation Digits BLANK
Trunk Name NAME

Peer-Details
type=peer
nat=no
insecure=very
host=X.X.X.X
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
canredirect=no
allow=ulaw

Any help will be appreciated

Your trunking provider may be sending calls from more than one IP address. On a failing call, your log likely contains “Rejecting unknown SIP connection from …” or "Received incoming SIP connection from unknown peer”. If so, find out what addresses your provider can send calls from. If there are only two or three, define a trunk for each additional address. If there are many, see https://community.asterisk.org/t/received-incoming-sip-connection-from-unknown-peer-from-registered-sip-provider/69721 for one approach, or consider using a pjsip trunk, where you can simply list the addresses or networks in the Match field.

If the above is not your issue or you still have trouble, post a log of a failing call.

Stewart thank you the response I already did work session with DID provider and they sent PCAP trace showing FreePBX answering the call and IP that is coming from is allowed. Like I said I am new at FreePbx I tried to capture the call on the Logs that portal shows me but was not able to there is a section where I can send command can i use this to capture a failing call

Reports -> Asterisk Logfiles by default will show you 500 lines, which should be plenty. Make a call to the PBX that fails, then click Show. If you have trouble with that, just search the file /var/log/asterisk/full for the relevant info.

Please confirm that the call was sent from the exact address specified by the host=x.x.x.x for your trunk. If not, what do you mean by ‘allowed’? (With default settings) if Asterisk gets a call from an IP address that does not match a trunk, it will route to a ‘not in service’ message.

Yes the Call was from the specific IP that is configured in my SIP trunk, so the originating IP inbound to my Freepbx is allowed.

Did the number in the SIP URI (what’s after ‘INVITE sip:’ in the pcap they sent) match an Incoming Route? Try defining a catch-all route (DID number left blank) and see whether the problem calls are caught by that.

Otherwise, post a log (if it’s big, paste it into http://pastebin.freepbx.org/ and post a link).

If needed, type
sip set debug on
at the Asterisk command prompt, which will cause SIP traces to also appear in the log.

Do not know if this will help but was able to get SIP Trace: IP from provider is where you see “Supplier IP”

<------------>
[Jun 14 19:33:20] VERBOSE[19735] pbx.c: == Spawn extension (from-sip-external, s, 8) exited non-zero on ‘SIP/Supplier IP-08a4aec0’
[Jun 14 19:33:20] VERBOSE[19735] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/Supplier IP-08a4aec0”, “”) in new stack
[Jun 14 19:33:20] VERBOSE[19735] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/Supplier IP-08a4aec0’
[Jun 14 19:33:20] VERBOSE[25699] chan_sip.c: Really destroying SIP dialog ‘[email protected]’ Method: BYE
[Jun 14 19:33:21] VERBOSE[19739] manager.c: == Manager ‘admin’ logged off from 127.0.0.1

SIP/2.0 200 OK
Via: SIP/2.0/UDP Supplier IP:5060;branch=z9hG4bK2sansay139592514rdb2506;received=Supplier IP
From: "John Doe sip:[email protected] IP>; tag=sansay139592514rdb2506
To: < sip:[email protected] >;tag=as4bc75dae
Call-ID: 52327530-0-680484268 @ 209.166.154.67
CSeq: 2 BYE
Server: FPBX-2.9.0(1.6.1.6)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Stewart the Incoming Route matches call fails intermittent other wise connects to the right IVR.

Sorry, I’m traveling today and don’t have time to answer. I hope that someone else will help.

I don’t understand, does a catch-all route go the IVR (or wherever you send it) or not?

You didn’t post the relevant part of the log. Use the pastebin mentioned earlier and post everything for a failed call. In particular, showing the incoming INVITE and Asterisk’s responses to it.

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