We have setup Freepbx but we we are calling to any mobile number its showing pilot number not the caller number. the setup are following.
Provider given us WAN IP POOL----- 100.64.xx.208/29 (Customer End IP: 100.64.xx.212 255.255.255.248 with Default Gateway— 100.64.xx.211) Voip Server IP 100.64.xxx.4
Pilot Number : 34823xx000 Start Number : 34823xx001 End Number : 34823xx149
The server has 2 interface one for internet another for VOIP .A static route create to on the server to route 100.64.xxx.0/24 traffic to 100.64.xx.211
we have setup a trunk “trunkout”
SIP Server 100.64.xxx.4
From Domain 100.64.xxx.4
From User +913482xx3000
Dial Number Manipulation Rules
match pattern XXXXXXXXXX
Outbound Dial Prefix +91
Outbound CID +913482xx3001
username and password
All other Extension with different number
Route CID +913482xx3000
Trunk Sequence for Matched Routes : trunkout
Dial Patterns : XXXXXXXXXX
Inbound Routes 1
DID : +913482xx3001
CID : Any
Destination Extensions: 001 +9134823xx001
Inbound Routes 2
DID : any
CID : Any
Destination Trunks: trunkout (pjsip)
All other Extensions which created with diffrent numbers
We are able making outgoing and income call from mobile but when we are calling to any number its giving us pilot on mobile. Please tell us where the mistake I have done.
Thanks in advance.
If your provider allows you to send caller ID in the From header, remove the From User setting. Or, if they accept the calling number in P-Asserted-Identity or Remote-Party-ID, set Send RPID/PAI accordingly. Otherwise, consult their documentation for how to send the calling number.
changing the P-Asserted-Identity or Remote-Party-ID not working. When removing From user number in trunk outgoing call not working. Whatever number I am giving from the range its showing that number on mobile. Is there anyway to solve this?
Do you mean in From User? If so, try leaving From User blank, but set Outbound CID for the extension to (for example)
Confirm that Route CID for the Outbound route (if set) is in the same format and Override Extension is not set. Also, Confirm that Outbound CallerID for the trunk is in the proper format and Allow Any CID is set.
If you still have trouble, at the Asterisk command prompt type
pjsip set logger on
and make a test call. The From header should contain the caller ID for the extension.
Thank you for your replay
when From User is set to blank, out going call stopped. Outbound CID is set in extension +913482xx3001 same format. Route CID for the Outbound route is set in the same format and Override Extension is not set. Outbound CallerID is trunk is set and Any CID enabled.
getting the following in the test call log
[2022-03-20 06:33:59] VERBOSE res_pjsip_logger.c: <--- Transmitting SIP request (453 bytes) to UDP:100.64.xxx.4:5060 --->
5646 PRACK sip:100.64.xxx.4:5060 SIP/2.0
5647 Via: SIP/2.0/UDP 100.64.74.212:5060;rport;branch=z9hG4bKPja85adba9-30f1-4693-8f7b-15d44900cffd
5648 From: <sip:[email protected]>;tag=cb04c33d-0a92-4b55-b6c3-ecceeb808a1b
5649 To: <sip:[email protected]>;tag=16-3717793-746065-140364115527424
5650 Call-ID: 7b7e5a53-ed1f-4f23-a4df-9d8f7ca712bc
5651 CSeq: 28189 PRACK
5652 RAck: 4 28185 INVITE
5653 Max-Forwards: 70
5654 User-Agent: FPBX-16.0.17(18.10.0)
5655 Content-Length: 0
I think some setting is to set which i have missed. Please help me.
Why do you believe the PRACK is in error? Without the complete context of the call it isn’t really possible to know whether anything is wrong, but, if PRACK gets sent at, the initial assumption would be that it is completely correctly sent.
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