Incoming Caller ID manipulation

Asterisk
1

Hi all,

I have a 3CX server and it has a SIP trunk to an Asterisk. Asterisk is setup to register to the 3CX as a peer. my question is;

when a call comes into asterisk, the caller ID shows as 11000(the peer extension) and the caller name is the extnesion that is calling on the 3CX side. See below for the invite. Is it possible to get asterisk to pull the caller ID from the Remote-party-id?

Request-Line: INVITE sip:2270@192.168.10.14:5060 SIP/2.0
    Method: INVITE
    Request-URI: sip:2270@192.168.10.14:5060
    [Resent Packet: False]
Message Header
    Via: SIP/2.0/UDP 192.168.10.12:5060;branch=z9hG4bK-d8754z-84470a753548743f-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:11000@192.168.10.12:5060>
    To: <sip:2270@192.168.10.14:5060>
    From: "nelson"<sip:11000@192.168.10.14:5060>;tag=0457d826
    Call-ID: ZDc2NTBmMDdiZDMwNDNmNjIwMTlhZDEyZWQwMjBlNmQ.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 14.0.49169.513 (48654)
    Content-Length: 209
    Remote-Party-ID: "nelson"<sip:1001@192.168.10.12:5060>;party=calling

My 3cx peer in sip.conf is setup as:

[3cx]
type=peer
insecure=invite,port
username=11000
fromuser=11000
secret=123456
host=192.168.10.12
context=from-3cx
port=5060
dmfmode=rfc2833
canreinvite=no
nat=never
disallow=all
allow=ulaw
qualify=yes

2

trustrpid=yes

However, I think this was introduced after canreinvite was deprecated.

3

Is this to be put in the sip.conf under that peer?

4

Yes. Please obtain a copy of sip.conf.sample for your version, which will explain the options allowed, and the correct way of controlling directmedia use.

5

Im running AsteriskNow 1.8 (just installed it fresh).

I dont have a sip.conf.sample, only the below:

[root@localhost asterisk]# ls *.sample
extensions_custom.conf.sample

6

You need to go to http://svn.digium.com/svn/asterisk/tags/xxxxxxx

where xxxxxxxx is your version of Asterisk then navigate down. In early versions, it in configs I think it is in configs/samples in later ones.

I would note that if the FreePBX peole had wanted you to hand configure sip.conf, they should have provided you with this file.