Dear Sirs,
PBX A - extensions with 1XX
10.7.208.245 freepbx 6 with Asterisk 11.20.0
|
|OpenVPN tunnel 10.7.218.3 gw
|
PBX B - extensions with 2XX
10.7.218.245 Freepbx 10.13.66-11 Asterisk 13.7.1
the pbx are configured as trunk:
[toPBX_B]
host=10.7.218.245
type=peer
insecure=port,invite
context=from-internal
[toPBX_A]
disallow=all
host=10.7.208.245
type=peer
allow=ulaw,alaw
nat=no
insecure=port,invite
context=from-internal
I can call from extension 250 of PBX_B to extension 150 of PBX_A, but not from A to B.
I’ve enabled sip debug and I get a SIP message of 401 Unauthorized.
PBX_SL*CLI> sip set debug on
SIP Debugging enabled
<— SIP read from UDP:10.7.218.3:5060 —>
jaK
<------------->
<— SIP read from UDP:10.7.218.3:1032 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.7.208.245:5060;branch=z9hG4bK67dbeed4
Max-Forwards: 70
From: “Chiara” sip:[email protected];tag=as133484b1
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76.2(11.20.0)
Date: Fri, 18 Nov 2016 14:18:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 405
v=0
o=root 2054851614 2054851614 IN IP4 10.7.208.245
s=Asterisk PBX 11.20.0
c=IN IP4 10.7.208.245
t=0 0
m=audio 15042 RTP/AVP 18 0 8 111 3 4 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (14 headers 18 lines) —
Sending to 10.7.218.3:1032 (NAT)
Sending to 10.7.218.3:1032 (NAT)
Using INVITE request as basis request - [email protected]:5060
No matching peer for ‘105’ from ‘10.7.218.3:1032’
<— Reliably Transmitting (NAT) to 10.7.218.3:1032 —>
SIP/2.0 401 Unauthorized – Remote UNIX connection disconnected
PBX_SL*CLI> sip set debug on
SIP Debugging enabled
<— SIP read from UDP:10.7.218.3:5060 —>
jaK
<------------->
<— SIP read from UDP:10.7.218.3:1032 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.7.208.245:5060;branch=z9hG4bK67dbeed4
Max-Forwards: 70
From: “Chiara” sip:[email protected];tag=as133484b1
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76.2(11.20.0)
Date: Fri, 18 Nov 2016 14:18:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 405
v=0
o=root 2054851614 2054851614 IN IP4 10.7.208.245
s=Asterisk PBX 11.20.0
c=IN IP4 10.7.208.245
t=0 0
m=audio 15042 RTP/AVP 18 0 8 111 3 4 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (14 headers 18 lines) —
Sending to 10.7.218.3:1032 (NAT)
Sending to 10.7.218.3:1032 (NAT)
Using INVITE request as basis request - [email protected]:5060
No matching peer for ‘105’ from ‘10.7.218.3:1032’
<— Reliably Transmitting (NAT) to 10.7.218.3:1032 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.7.208.245:5060;branch=z9hG4bK67dbeed4;received=10.7.218.3;rport=1032
From: “Chiara” sip:[email protected];tag=as133484b1
To: sip:[email protected];tag=as62357378
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: FPBX-13.0.190.2(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“614cf9be”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:10.7.218.3:1032 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.7.208.245:5060;branch=z9hG4bK67dbeed4
Max-Forwards: 70
From: “Chiara” sip:[email protected];tag=as133484b1
To: sip:[email protected];tag=as62357378
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.2(11.20.0)
Content-Length: 0
<------------->
— (10 headers 0 lines) —
<— SIP read from UDP:10.7.218.3:5060 —>
jaK
Via: SIP/2.0/UDP 10.7.208.245:5060;branch=z9hG4bK67dbeed4;received=10.7.218.3;rport=1032
From: “Chiara” sip:[email protected];tag=as133484b1
To: sip:[email protected];tag=as62357378
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: FPBX-13.0.190.2(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“614cf9be”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:10.7.218.3:1032 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.7.208.245:5060;branch=z9hG4bK67dbeed4
Max-Forwards: 70
From: “Chiara” sip:[email protected];tag=as133484b1
To: sip:[email protected];tag=as62357378
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.2(11.20.0)
Content-Length: 0
<------------->
— (10 headers 0 lines) —
<— SIP read from UDP:10.7.218.3:5060 —>
jaK
How can I allow the PBX B to call his extensions from PBX_A extensions?