I say its a stange case of Normal Clearing but.....
So we have a cloud hosted Multi Tennat Fusion PBX and yesterday we had an issue with one of our clients. They have been experiencing power cuts over the last couple of days. When they restart they had issues with outbond calls on the phones. Intially we sorted this it was down to their external IP address changing.
~However today we had the same issue but now cat resolve it. Calls can be made in but not out. We have an external phone connected to the same domain and that one is fine. They can call internally and I can call in from the external extention. they cant call external phone (obviously as its external...)
I cant see any traffic going out so im now a bit lost. I have tried to to a pcap in a call and thats below : Any Ideas?
´Ä_^ñ
> > $Ä : E (üö@ 4ëªR!¤Ÿ%;B3/ÄiÒ_*ç–’ŒP(—£ ¶Ä_^—Ï ë ë $Ä E ÛÖh@ 6ØBT\Ö§%;B3/Äh],m7 PÚ] INVITE sip:115@LH.voip.synthesis-it.co.uk:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.8:12159;branch=z9hG4bK3823221383
From: "Julie" <sip:102@LH.voip.synthesis-it.co.uk:5060>;tag=1115306992
To: <sip:115@LH.voip.synthesis-it.co.uk:5060>
Call-ID: 0_2082199486@192.168.1.8
CSeq: 1 INVITE
Contact: <sip:102@192.168.1.8:12159;transport=TCP>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T23G 44.84.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 304
v=0
o=- 20003 20003 IN IP4 192.168.1.8
s=SDP data
c=IN IP4 192.168.1.8
t=0 0
m=audio 12272 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
¶Ä_^qÐ ƒ ƒ RÛIO E sûˆ@ @«Š%;B3T\Ö§Ä/,m7 h`ÎPc“× SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.8:12159;branch=z9hG4bK3823221383;received=84.92.214.167;rport=12159
From: "Julie" <sip:102@LH.voip.synthesis-it.co.uk:5060>;tag=1115306992
To: <sip:115@LH.voip.synthesis-it.co.uk:5060>
Call-ID: 0_2082199486@192.168.1.8
CSeq: 1 INVITE
User-Agent: FreeSWITCH
Content-Length: 0
¶Ä_^©Ó D D RÛ@ E 4ž@ @m©%;B334”vÄÔMœÚüDQL€¬/?
XÚÐZ‹¶Ä_^‡è œ œ RÛag E Œû‰@ @©p%;B3T\Ö§Ä/,m8Kh`ÎPc•ð SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TCP 192.168.1.8:12159;branch=z9hG4bK3823221383;received=84.92.214.167;rport=12159
Max-Forwards: 70
From: "Julie" <sip:102@LH.voip.synthesis-it.co.uk:5060>;tag=1115306992
To: <sip:115@LH.voip.synthesis-it.co.uk:5060>;tag=yaae5cXtvBB6g
Call-ID: 0_2082199486@192.168.1.8
CSeq: 1 INVITE
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Remote-Party-ID: "115" <sip:115@LH.voip.synthesis-it.co.uk>;party=calling;privacy=off;screen=no
¶Ä_^V' > > $Ä ŸR E (Öi@ 6ÛôT\Ö§%;B3/Äh`Î,m8KPÚëV ¶Ä_^S4 D D $Ä ¿¹ E 4M–@ 4Ê34”v%;B3ÔÄDQLMœÚý€ ÁF
ZÈXÚжÄ_^^? > > $Ä E (Öj@ 6ÛóT\Ö§%;B3/Äh`Î,m;¯PŒæ@ ¶Ä_^ÚS x x $Ä pp E hÖk@ 6Ú²T\Ö§%;B3/Äh`Î,m;¯PŒ ñ ACK sip:115@LH.voip.synthesis-it.co.uk:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.8:12159;branch=z9hG4bK3823221383
From: "Julie" <sip:102@LH.voip.synthesis-it.co.uk:5060>;tag=1115306992
To: <sip:115@LH.voip.synthesis-it.co.uk:5060>;tag=yaae5cXtvBB6g
Call-ID: 0_2082199486@192.168.1.8
CSeq: 1 ACK
Content-Length: 0
So we have a cloud hosted Multi Tennat Fusion PBX and yesterday we had an issue with one of our clients. They have been experiencing power cuts over the last couple of days. When they restart they had issues with outbond calls on the phones. Intially we sorted this it was down to their external IP address changing.
~However today we had the same issue but now cat resolve it. Calls can be made in but not out. We have an external phone connected to the same domain and that one is fine. They can call internally and I can call in from the external extention. they cant call external phone (obviously as its external...)
I cant see any traffic going out so im now a bit lost. I have tried to to a pcap in a call and thats below : Any Ideas?
´Ä_^ñ
> > $Ä : E (üö@ 4ëªR!¤Ÿ%;B3/ÄiÒ_*ç–’ŒP(—£ ¶Ä_^—Ï ë ë $Ä E ÛÖh@ 6ØBT\Ö§%;B3/Äh],m7 PÚ] INVITE sip:115@LH.voip.synthesis-it.co.uk:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.8:12159;branch=z9hG4bK3823221383
From: "Julie" <sip:102@LH.voip.synthesis-it.co.uk:5060>;tag=1115306992
To: <sip:115@LH.voip.synthesis-it.co.uk:5060>
Call-ID: 0_2082199486@192.168.1.8
CSeq: 1 INVITE
Contact: <sip:102@192.168.1.8:12159;transport=TCP>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T23G 44.84.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 304
v=0
o=- 20003 20003 IN IP4 192.168.1.8
s=SDP data
c=IN IP4 192.168.1.8
t=0 0
m=audio 12272 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
¶Ä_^qÐ ƒ ƒ RÛIO E sûˆ@ @«Š%;B3T\Ö§Ä/,m7 h`ÎPc“× SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.8:12159;branch=z9hG4bK3823221383;received=84.92.214.167;rport=12159
From: "Julie" <sip:102@LH.voip.synthesis-it.co.uk:5060>;tag=1115306992
To: <sip:115@LH.voip.synthesis-it.co.uk:5060>
Call-ID: 0_2082199486@192.168.1.8
CSeq: 1 INVITE
User-Agent: FreeSWITCH
Content-Length: 0
¶Ä_^©Ó D D RÛ@ E 4ž@ @m©%;B334”vÄÔMœÚüDQL€¬/?
XÚÐZ‹¶Ä_^‡è œ œ RÛag E Œû‰@ @©p%;B3T\Ö§Ä/,m8Kh`ÎPc•ð SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TCP 192.168.1.8:12159;branch=z9hG4bK3823221383;received=84.92.214.167;rport=12159
Max-Forwards: 70
From: "Julie" <sip:102@LH.voip.synthesis-it.co.uk:5060>;tag=1115306992
To: <sip:115@LH.voip.synthesis-it.co.uk:5060>;tag=yaae5cXtvBB6g
Call-ID: 0_2082199486@192.168.1.8
CSeq: 1 INVITE
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Remote-Party-ID: "115" <sip:115@LH.voip.synthesis-it.co.uk>;party=calling;privacy=off;screen=no
¶Ä_^V' > > $Ä ŸR E (Öi@ 6ÛôT\Ö§%;B3/Äh`Î,m8KPÚëV ¶Ä_^S4 D D $Ä ¿¹ E 4M–@ 4Ê34”v%;B3ÔÄDQLMœÚý€ ÁF
ZÈXÚжÄ_^^? > > $Ä E (Öj@ 6ÛóT\Ö§%;B3/Äh`Î,m;¯PŒæ@ ¶Ä_^ÚS x x $Ä pp E hÖk@ 6Ú²T\Ö§%;B3/Äh`Î,m;¯PŒ ñ ACK sip:115@LH.voip.synthesis-it.co.uk:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.8:12159;branch=z9hG4bK3823221383
From: "Julie" <sip:102@LH.voip.synthesis-it.co.uk:5060>;tag=1115306992
To: <sip:115@LH.voip.synthesis-it.co.uk:5060>;tag=yaae5cXtvBB6g
Call-ID: 0_2082199486@192.168.1.8
CSeq: 1 ACK
Content-Length: 0