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
PJSIP trunking Asterisk 16 with OXE 12.2
- ReverseFlash
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- frank
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Re: PJSIP trunking Asterisk 16 with OXE 12.2
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.
I'll check it out and will update with authentication.
Code Free Or Die
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