Hello, Good afternoon.
I'm making a connection with a asterisk and IPO8.0 and when I try to call astreisk introduced SIP/2.0 503 Service Unavailable error.
Could someone help me?
Trace:
88043976mS SIP Tx: UDP 10.8.2.5:5060 -> 10.1.13.2:5060
INVITE sip:1144@10.1.13.2 SIP/2.0
Via: SIP/2.0/UDP 10.8.2.5:5060;rport;branch=z9hG4bK29b3852f5f331450cfa6bacf83692219
From: "Client" <sip:8567@10.1.13.2>;tag=9c5b47e75af39aac
To: <sip:1144@10.1.13.2>
Call-ID: d2aa7379faa7d1fa015625b3656fafb7
CSeq: 1888333747 INVITE
Contact: "Client" <sip:8567@10.8.2.5:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer,100rel
User-Agent: IP Office 8.1 (69)
Content-Length: 291
v=0
o=UserA 1720802966 4284920706 IN IP4 10.8.2.5
s=Session SDP
c=IN IP4 10.8.2.5
t=0 0
m=audio 49152 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
88044214mS SIP Rx: UDP 10.1.13.2:5060 -> 10.8.2.5:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.2.5:5060;branch=z9hG4bK29b3852f5f331450cfa6bacf83692219;received=10.8.2.5;rport=5060
From: "Client" <sip:8567@10.1.13.2>;tag=9c5b47e75af39aac
To: <sip:1144@10.1.13.2>
Call-ID: d2aa7379faa7d1fa015625b3656fafb7
CSeq: 1888333747 INVITE
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:1144@10.1.13.2>
Content-Length: 0
88044245mS SIP Rx: UDP 10.1.13.2:5060 -> 10.8.2.5:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.8.2.5:5060;branch=z9hG4bK29b3852f5f331450cfa6bacf83692219;received=10.8.2.5;rport=5060
From: "Client" <sip:8567@10.1.13.2>;tag=9c5b47e75af39aac
To: <sip:1144@10.1.13.2>;tag=as6c86d201
Call-ID: d2aa7379faa7d1fa015625b3656fafb7
CSeq: 1888333747 INVITE
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:1144@10.1.13.2>
Content-Length: 0
88044249mS SIP Tx: UDP 10.8.2.5:5060 -> 10.1.13.2:5060
ACK sip:1144@10.1.13.2 SIP/2.0
Via: SIP/2.0/UDP 10.8.2.5:5060;rport;branch=z9hG4bK29b3852f5f331450cfa6bacf83692219
From: "Client" <sip:8567@10.1.13.2>;tag=9c5b47e75af39aac
To: <sip:1144@10.1.13.2>;tag=as6c86d201
Call-ID: d2aa7379faa7d1fa015625b3656fafb7
CSeq: 1888333747 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
User-Agent: IP Office 8.1 (69)
Content-Length: 0
88044250mS Sip: Receive Temporary Failure on SIPTrunkEndpoint
88044253mS Sip: SIPDialog f50ba378 destroyed, size
I'm making a connection with a asterisk and IPO8.0 and when I try to call astreisk introduced SIP/2.0 503 Service Unavailable error.
Could someone help me?
Trace:
88043976mS SIP Tx: UDP 10.8.2.5:5060 -> 10.1.13.2:5060
INVITE sip:1144@10.1.13.2 SIP/2.0
Via: SIP/2.0/UDP 10.8.2.5:5060;rport;branch=z9hG4bK29b3852f5f331450cfa6bacf83692219
From: "Client" <sip:8567@10.1.13.2>;tag=9c5b47e75af39aac
To: <sip:1144@10.1.13.2>
Call-ID: d2aa7379faa7d1fa015625b3656fafb7
CSeq: 1888333747 INVITE
Contact: "Client" <sip:8567@10.8.2.5:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer,100rel
User-Agent: IP Office 8.1 (69)
Content-Length: 291
v=0
o=UserA 1720802966 4284920706 IN IP4 10.8.2.5
s=Session SDP
c=IN IP4 10.8.2.5
t=0 0
m=audio 49152 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
88044214mS SIP Rx: UDP 10.1.13.2:5060 -> 10.8.2.5:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.2.5:5060;branch=z9hG4bK29b3852f5f331450cfa6bacf83692219;received=10.8.2.5;rport=5060
From: "Client" <sip:8567@10.1.13.2>;tag=9c5b47e75af39aac
To: <sip:1144@10.1.13.2>
Call-ID: d2aa7379faa7d1fa015625b3656fafb7
CSeq: 1888333747 INVITE
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:1144@10.1.13.2>
Content-Length: 0
88044245mS SIP Rx: UDP 10.1.13.2:5060 -> 10.8.2.5:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.8.2.5:5060;branch=z9hG4bK29b3852f5f331450cfa6bacf83692219;received=10.8.2.5;rport=5060
From: "Client" <sip:8567@10.1.13.2>;tag=9c5b47e75af39aac
To: <sip:1144@10.1.13.2>;tag=as6c86d201
Call-ID: d2aa7379faa7d1fa015625b3656fafb7
CSeq: 1888333747 INVITE
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:1144@10.1.13.2>
Content-Length: 0
88044249mS SIP Tx: UDP 10.8.2.5:5060 -> 10.1.13.2:5060
ACK sip:1144@10.1.13.2 SIP/2.0
Via: SIP/2.0/UDP 10.8.2.5:5060;rport;branch=z9hG4bK29b3852f5f331450cfa6bacf83692219
From: "Client" <sip:8567@10.1.13.2>;tag=9c5b47e75af39aac
To: <sip:1144@10.1.13.2>;tag=as6c86d201
Call-ID: d2aa7379faa7d1fa015625b3656fafb7
CSeq: 1888333747 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
User-Agent: IP Office 8.1 (69)
Content-Length: 0
88044250mS Sip: Receive Temporary Failure on SIPTrunkEndpoint
88044253mS Sip: SIPDialog f50ba378 destroyed, size