Pjsip trunk to a static ip gateway

I have this scenario.

      asterisk     ---------sip trunk-----> fxs/fxo gateway (QUINTUM)

192.168.133.100 192.168.133.110

with a 37400 user defined at gateway.

at pjsip.conf:

[gateway]
type=endpoint
context=from-pstn
disallow=all
allow=ulaw,alaw
aors=gateway
direct_media=no

[gateway]
;;type=identify
endpoint=gateway
match=192.168.133.110 ; ip of the gateway

[gateway]
type=aor
contact=sip:192.168.133.110:5060

37400 registers to my asterisk.

but if i uncomment type=identify

then 37400 can no more register to my asterisk

these are the logs of the operation:

<— Received SIP request (414 bytes) from UDP:192.168.133.110:5060 —>
REGISTER sip:192.168.133.100 SIP/2.0
Call-ID: call-F1901175-0BA2-2110-060F-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 9 REGISTER
Expires: 300
From: sip:37400@192.168.133.100;tag=c0a8856e-6
Max-Forwards: 70
To: sip:37400@192.168.133.100
User-Agent: Quintum/1.0.0 SN/0030E1205D7B SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0013

<— Transmitting SIP response (376 bytes) to UDP:192.168.133.110:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0013
Call-ID: call-F1901175-0BA2-2110-060F-0@192.168.133.110
From: sip:37400@192.168.133.100;tag=c0a8856e-6
To: sip:37400@192.168.133.100;tag=z9hG4bK-tenor-c0a8-856e-0013
CSeq: 9 REGISTER
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

[2021-01-11 14:33:29] WARNING[2357]: res_pjsip_registrar.c:1080 find_registrar_aor: AOR ‘’ not found for endpoint ‘gateway’

<— Received SIP request (424 bytes) from UDP:192.168.133.110:5061 —>
REGISTER sip:192.168.133.100 SIP/2.0
Call-ID: call-F148764E-0BA2-2110-0611-2@192.168.133.110
Contact: sip:37400@192.168.133.110:5061
Content-Length: 0
CSeq: 9 REGISTER
Expires: 300
From: sip:37400@192.168.133.100;tag=c0a8856e-8
Max-Forwards: 70
To: sip:37400@192.168.133.100
User-Agent: Quintum/1.0.0 SN/0030E1205D7B SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110:5061;branch=z9hG4bK-tenor-c0a8-856e-0014

<— Transmitting SIP response (381 bytes) to UDP:192.168.133.110:5061 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.133.110:5061;rport=5061;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0014
Call-ID: call-F148764E-0BA2-2110-0611-2@192.168.133.110
From: sip:37400@192.168.133.100;tag=c0a8856e-8
To: sip:37400@192.168.133.100;tag=z9hG4bK-tenor-c0a8-856e-0014
CSeq: 9 REGISTER
Server: FPBX-15.0.16.81(16.13.0)
Content-Length: 0

same scenario but with a different model gateway:

     **asterisk    -----siptrunk-----> fxs/fxo gateway (grandstream for example)**

192.168.133.100 192.168.133.110

[gateway]
type=endpoint
context=from-pstn
disallow=all
allow=ulaw,alaw,gsm,g726,g722
aors=gateway

[gateway]
type=identify
endpoint=gateway
match=192.168.133.110

[gateway]
type=aor
contact=sip:192.168.133.110:5060

type= identify is not commented and 37400 registers normally.

so why with different gateways the same scenario is not working ?
and what is the role of this type=identify parameter !?
what are the standard configurations needed to be done at pjsip.conf for a static sip trunk to a static gaetway?
and what are the standard configurations of a non static siptrunk to a non static gateway?

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