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PJSIP trunking Asterisk 16 with OXE 12.2

Posted: 05 Feb 2021 14:46
by ReverseFlash
Hi guys!

I was checking a configuration to SIP Trunking between Asterisk 16 and OXE 12.2 through PJSIP.
I found this parameters and i share with you, my little bit contribution to this pages.

I notice this configuration is WITHOUT REGISTRATION (user,password) in Asterisk and OXE

If someone else has the configuration witht Registration from the OXE share please.

Prompt: ASTERISK:

[root@str asterisk]# cat pjsip.conf

[transport-udp]
type=transport
protocol=udp
bind=10.247.0.100:5060

;PJSIP Trunking

[oxe]
type=endpoint
transport=transport-udp
context=from-external
disallow=all
allow=ulaw,alaw,g729
aors=oxe
rtp_symmetric=yes

[oxe]
type=auth
auth_type=userpass
password=
username=oxe

[oxe]
type=aor
qualify_frequency=60
contact=sip:192.168.5.80:5060

[oxe]
type=identify
endpoint=oxe
match=192.168.5.80

Prompt OXE:

1. SIP Gateway:

lqReview/Modify: SIP Gatewayqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqk
x x
x Node Number (reserved) : 101 x
x Instance (reserved) : 1 x
x Instance (reserved) : 1 x
x x
x SIP Subnetwork : 10 x
x SIP Trunk Group : 101 x
x IP Address : 192.168.5.80 x
x Machine name - Host : oxebsiteb x
x SIP Proxy Port Number : 5060 x
x SIP Subscribe Min Duration : 1800 x
x SIP Subscribe Max Duration : 86400 x
x Session Timer : 1800 x
x Min Session Timer : 900 x
x Session Timer Method + RE_INVITE x
x DNS local domain name : -------------------------------------------------- x
x DNS type + DNS A x
x SIP DNS1 IP Address : ----------------------------------------------- x
x SIP DNS2 IP Address : ----------------------------------------------- x
x SDP in 18x + True x
x CAC SIP-SIP + False x
x INFO method for remote extension + False x
x Dynamic Payload type for DTMF : 101 x
x Overflow Licenses Threshold : 80 x
x x
mqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqj

Trunk group:


lqReview/Modify: Trunk Groupsqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqk
x x
x Node Number (reserved) : 101 x
x Trunk Group ID : 101 x
x x
x Trunk Group Type + T2 x
x Trunk Group Name : SIP OXE x
x UTF-8 Trunk Group Name : ------------------------------------------- x
x Number Compatible With : -1 x
x Remote Network : 10 x
x Shared Trunk Group + False x
x Special Services + Nothing x
x Node number : 1 x
x Transcom Trunk Group + False x
x Auto.reserv.by Attendant + False x
x Overflow trunk group No. : -1 x
x Tone on seizure + False x
x Private Trunk Group + False x
x Q931 Signal variant + ABC-F x
x SS7 Signal variant + No variant x
x Number Of Digits To Send : 0 x
x Channel selection type + Quantified x
x Auto.DTMF dialing on outgoing call + NO x
x T2 Specification + SIP x
x Homogenous network for direct RTP + NO x
x Public Network COS : 31 x
x DID transcoding + False x
x Can support UUS in SETUP + True x
x Associated Ext SIP gateway : -1 x
x x
x Implicit Priority x
x x
x Activation mode : 0 x
x Priority Level : 0 x
x x
x Preempter + NO x
x Incoming calls Restriction COS : 10 x
x Outgoing calls Restriction COS : 10 x
x Callee number mpt1343 + NO x
x Overlap dialing + YES x
x Call diversion in ISDN + NO x
x x
mqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqj

SIP External Gateway:

x x
x Node Number (reserved) : 101 x
x Instance (reserved) : 1 x
x SIP External Gateway ID : 1 x
x x
x Gateway Name : Asterisk x
x SIP Remote domain : 10.247.0.100 x
x PCS IP Address : ----------------------------------------------- x
x SIP Port Number : 5060 x
x Transport type + UDP x
x Belonging Domain : -------------------------------------------------- x
x Registration ID : -------------------------------------------------- x
x Registration ID P_Asserted + False x
x Registration timer : 0 x
x SIP Outbound Proxy : -------------------------------------------------- x
x Supervision timer : 0 x
x Trunk group number : 1 x
x Pool Number : -1 x
x Outgoing realm : -------------------------------------------------- x
x Outgoing username : -------------------------------------------------- x
x x
x Outgoing Password : -------------------- x
x Confirm : -------------------- x
x x
x Incoming username : -------------------------------------------------- x
x x
x Incoming Password : -------------------- x
x Confirm : -------------------- x
x x
x RFC 3325 supported by the distant + True x
x DNS type + DNS A x
x SIP DNS1 IP Address : ----------------------------------------------- x
x SIP DNS2 IP Address : ----------------------------------------------- x
x SDP in 18x + False x
x Minimal authentication method + SIP None x
x INFO method for remote extension + False x
x To EMS + False x
x SRTP + RTP only x
x Ignore inactive/black hole + True x
x Contact with IP address + False x
x Dynamic Payload type for DTMF : 101 x
x Outbound Calls 100 REL + Supported x
x Incoming Calls 100 REL + Not Requested x
x Gateway type + Standard type x
x Re-Trans No. for REGISTER/OPTIONS : 2 x
x P-Asserted-ID in Calling Number + False x
x Trusted P-Asserted-ID header + True x
x Diversion Info to provide via + History Info x
x Proxy identification on IP address + False x
x Outbound calls only + False x
x SDP relay on Ext. Call Fwd + Default x
x SDP Transparency Override + False x
x RFC 5009 supported / Outbound call + Not Supported x
x Nonce caching activation + NO x
x FAX Procedure Type + T38 only x
x DNS SRV/Call retry on busy server : 0 x
x Unattended Transfer for RSI + NO x
x Redirection functionality + NO x
x Attended Transfer + NO x
x Send BYE on REFER + YES x
x Support Redirection response + NO x
x OPTIONS required + YES x
x Support UTF8 characters set + NO x
x Support CSTA User-to-User + NO x
x DDI destination number + ReqURI x
x Video Support Profile + Not Supported x
x UPDATE in Allow header/INVITE + Optional x
x RFC 4904 supported + NO x
x Bulk registration (RFC 6140) + NO x
x RFC3264 m-line + True x
x Sendonly for hold + True x
x In Band DTMF + NO x
x Trusted From header + False x
x Support Re-invite without SDP + True x
x Type of codec negotiation + Single codec G729 x
x x
mqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqj

bbc

Re: PJSIP trunking Asterisk 16 with OXE 12.2

Posted: 18 Feb 2021 15:49
by frank
ha - I have used this years ago, and need to setup asterisk in my lab.
I'll check it out and will update with authentication.