Dear,
I have an IP Office 500 V2 connected to an Asterisk via SIP Trunk. Calls from Asterisk to IPO it's OK.
Calls from the IP Office to the Asteisk fail. In the SIP traceo I am see the following error:
Thank you very much for any help.
Regards
773110mS SIP Call Tx: 17
INVITE sip:252@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.50.2:5060;rport;branch=z9hG4bK2946989dfc50d18ddb6ee8625f68b404
From: "IP1" <sip:3200@172.16.20.5>;tag=6172094185efa652
To: <sip:252@172.16.20.5>
Call-ID: 8628aea8468e32c6b1d1bfa56ec0422a
CSeq: 1492371744 INVITE
Contact: "IP1" <sip:3200@172.16.50.2:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 8.1 (73)
Content-Length: 260
v=0
o=UserA 1073276292 127401303 IN IP4 172.16.50.2
s=Session SDP
c=IN IP4 172.16.50.2
t=0 0
m=audio 49156 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
773353mS SIP Call Rx: 17
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.50.2:5060;branch=z9hG4bK2946989dfc50d18ddb6ee8625f68b404;received=172.16.50.2;rport=5060
From: "IP1" <sip:3200@172.16.20.5>;tag=6172094185efa652
To: <sip:252@172.16.20.5>;tag=as47ef845c
Call-ID: 8628aea8468e32c6b1d1bfa56ec0422a
CSeq: 1492371744 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Digest algorithm=MD5, realm="asterisk", nonce="133ea6b6"
Content-Length: 0
773356mS SIP Call Tx: 17
ACK sip:252@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.50.2:5060;rport;branch=z9hG4bK2946989dfc50d18ddb6ee8625f68b404
From: "IP1" <sip:3200@172.16.20.5>;tag=6172094185efa652
To: <sip:252@172.16.20.5>;tag=as47ef845c
Call-ID: 8628aea8468e32c6b1d1bfa56ec0422a
CSeq: 1492371744 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
User-Agent: IP Office 8.1 (73)
Content-Length: 0
I have an IP Office 500 V2 connected to an Asterisk via SIP Trunk. Calls from Asterisk to IPO it's OK.
Calls from the IP Office to the Asteisk fail. In the SIP traceo I am see the following error:
Thank you very much for any help.
Regards
773110mS SIP Call Tx: 17
INVITE sip:252@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.50.2:5060;rport;branch=z9hG4bK2946989dfc50d18ddb6ee8625f68b404
From: "IP1" <sip:3200@172.16.20.5>;tag=6172094185efa652
To: <sip:252@172.16.20.5>
Call-ID: 8628aea8468e32c6b1d1bfa56ec0422a
CSeq: 1492371744 INVITE
Contact: "IP1" <sip:3200@172.16.50.2:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
User-Agent: IP Office 8.1 (73)
Content-Length: 260
v=0
o=UserA 1073276292 127401303 IN IP4 172.16.50.2
s=Session SDP
c=IN IP4 172.16.50.2
t=0 0
m=audio 49156 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
773353mS SIP Call Rx: 17
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.50.2:5060;branch=z9hG4bK2946989dfc50d18ddb6ee8625f68b404;received=172.16.50.2;rport=5060
From: "IP1" <sip:3200@172.16.20.5>;tag=6172094185efa652
To: <sip:252@172.16.20.5>;tag=as47ef845c
Call-ID: 8628aea8468e32c6b1d1bfa56ec0422a
CSeq: 1492371744 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Digest algorithm=MD5, realm="asterisk", nonce="133ea6b6"
Content-Length: 0
773356mS SIP Call Tx: 17
ACK sip:252@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.50.2:5060;rport;branch=z9hG4bK2946989dfc50d18ddb6ee8625f68b404
From: "IP1" <sip:3200@172.16.20.5>;tag=6172094185efa652
To: <sip:252@172.16.20.5>;tag=as47ef845c
Call-ID: 8628aea8468e32c6b1d1bfa56ec0422a
CSeq: 1492371744 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
User-Agent: IP Office 8.1 (73)
Content-Length: 0