Inbound route to 2 PABX

I have 1 GSM gateway integrate to 1 PBX( location A ) and another Pabx at (location B) with different public IP address. GSM gateway will pass the inbound call to PABX A and forward to sip extension 101. At PABX B use this sip extension 101(created at PABX A) register as sip trunk. at this flow, I unable to capture the called ID(GSM) at PABX B. any expert can help?

You need to create a real trunk between the two PBXs, either SIP or IAX, and define an intra-company route in order to preserve caller id.

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Due to PABX B not allow to open those port required, we can’t configure it as trunk to trunk.any way to use sip extension and manage to receive incoming call?

Setting up a trunk does not necessarily require modifying any firewall configuration.

On PBX B, at the Asterisk command prompt, type
pjsip set logger on
or
sip set debug on
according to the type of the extension or trunk on PBX B. Make a test call, go to Reports -> Asterisk Logfiles and view the incoming INVITE. If the caller ID is present somewhere it should be possible to adjust settings to capture it. Paste the log at pastebin.freepbx.org and post the link here.

Otherwise, you will need admin access to PBX A to modify its config. Please post details about how the extension is presently set up, including the configuration at both ends, which side is registering, etc.

I got the below invite event from sip debug. try to set the s as DID number on inbound route manage to receive call. By right we should get the correct DID to distribute the inbound call. Any idea?

<β€” SIP read from UDP:PABXA_IP:5060 β€”>

INVITE sip:[email protected]_IP:5060 SIP/2.0

Via: SIP/2.0/UDP PABXA_IP:5060;branch=z9hG4bK4250db02;rport

Max-Forwards: 70

From: β€œ0164219123” <sip:[email protected]_IP>;tag=as0bd319b9

To: <sip:[email protected]_IP:5060>

Contact: <sip:[email protected]_IP:5060>

Call-ID: [email protected]_IP:5060

CSeq: 102 INVITE

User-Agent: IPBX-2.11.0(11.25.3)

Date: Mon, 28 Sep 2020 11:34:18 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 289

v=0

o=root 1469381116 1469381116 IN IP4 PABXA_IP

s=Asterisk PBX 11.25.3

c=IN IP4 PABXA_IP

t=0 0

m=audio 17328 RTP/AVP 0 3 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

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