Please, I beg of you, some help to avoid driving my car over a cliff in sheer desperation!
This is not an urgent issue (or was not when I started), I am now tearing my hair out.
I have had two engineers look at this as well as me.
I/c calls get a couple of beeps in the callers ear and then clear down, this is the message:
808160mS SIP Rx: UDP xx.245.6.81:5060 -> xxx.231.216.140:5060
INVITE sip:02080901280@xxx.231.216.140 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:02080901280@xx.245.6.81:5060>
From: <sip:07973673265@xx.245.6.81>;tag=3517057571-578992
Remote-Party-Id: <sip:7973673265@xx.245.6.81>;screen=yes;privacy=off
Call-ID: 1291655-3517057571-578987@MSX11.test.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 83.245.6.81:5060;x-route-tag="tgrp:test-INGRESS-TRKGRP";branch=z9hG4bK84c47011c63d2022d41708c4fea692b3
Contact: <sip:07973673265@83.xxx.6.81:5060>
Expires: 180
Call-Info: <sip:xx.245.6.81>;method="NOTIFY;Event=telephone-event;Duration=1000"
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 260
v=0
o=MSX11 5577 7987 IN IP4 xx.245.6.81
s=sip call
c=IN IP4 xx.245.6.82
t=0 0
m=audio 36202 RTP/AVP 18 100 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
808663mS SIP Rx: UDP xx.245.6.81:5060 -> xx.231.216.140:5060
INVITE sip:02080901280@xxx.231.216.140 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:02080901280@xx.245.6.81:5060>
From: <sip:07973673265@xx.245.6.81>;tag=3517057571-578992
Remote-Party-Id: <sip:7973673265@xx.245.6.81>;screen=yes;privacy=off
Call-ID: 1291655-3517057571-578987@MSX11.test.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP xx.245.6.81:5060;x-route-tag="tgrp:test-INGRESS-TRKGRP";branch=z9hG4bK84c47011c63d2022d41708c4fea692b3
Contact: <sip:07973673265@xx.245.6.81:5060>
Expires: 180
Call-Info: <sip:xx.245.6.81>;method="NOTIFY;Event=telephone-event;Duration=1000"
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 260
v=0
o=MSX11 5577 7987 IN IP4 xx.245.6.81
s=sip call
c=IN IP4 xx.245.6.82
t=0 0
m=audio 36202 RTP/AVP 18 100 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
809663mS SIP Rx: UDP xx.245.6.81:5060 -> xxx.231.216.140:5060
INVITE sip:02080901280@xxx.231.216.140 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:02080901280@xx.245.6.81:5060>
From: <sip:07973673265@xx.245.6.81>;tag=3517057571-578992
Remote-Party-Id: <sip:7973673265@xx.245.6.81>;screen=yes;privacy=off
Call-ID: 1291655-3517057571-578987@MSX11.test.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP xx.245.6.81:5060;x-route-tag="tgrp:test-INGRESS-TRKGRP";branch=z9hG4bK84c47011c63d2022d41708c4fea692b3
Contact: <sip:07973673265@xx.245.6.81:5060>
Expires: 180
Call-Info: <sip:xx.245.6.81>;method="NOTIFY;Event=telephone-event;Duration=1000"
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 260
v=0
o=MSX11 5577 7987 IN IP4 xx.245.6.81
s=sip call
c=IN IP4 xx.245.6.82
t=0 0
m=audio 36202 RTP/AVP 18 100 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Trunk says out of service.
Have configured as per this document as well:
It just will not work, there have been issues with them but they say that all is well their end.
Any help would be greatly appreciated.
Many thanks
Desperate Dan
This is not an urgent issue (or was not when I started), I am now tearing my hair out.
I have had two engineers look at this as well as me.
I/c calls get a couple of beeps in the callers ear and then clear down, this is the message:
808160mS SIP Rx: UDP xx.245.6.81:5060 -> xxx.231.216.140:5060
INVITE sip:02080901280@xxx.231.216.140 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:02080901280@xx.245.6.81:5060>
From: <sip:07973673265@xx.245.6.81>;tag=3517057571-578992
Remote-Party-Id: <sip:7973673265@xx.245.6.81>;screen=yes;privacy=off
Call-ID: 1291655-3517057571-578987@MSX11.test.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 83.245.6.81:5060;x-route-tag="tgrp:test-INGRESS-TRKGRP";branch=z9hG4bK84c47011c63d2022d41708c4fea692b3
Contact: <sip:07973673265@83.xxx.6.81:5060>
Expires: 180
Call-Info: <sip:xx.245.6.81>;method="NOTIFY;Event=telephone-event;Duration=1000"
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 260
v=0
o=MSX11 5577 7987 IN IP4 xx.245.6.81
s=sip call
c=IN IP4 xx.245.6.82
t=0 0
m=audio 36202 RTP/AVP 18 100 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
808663mS SIP Rx: UDP xx.245.6.81:5060 -> xx.231.216.140:5060
INVITE sip:02080901280@xxx.231.216.140 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:02080901280@xx.245.6.81:5060>
From: <sip:07973673265@xx.245.6.81>;tag=3517057571-578992
Remote-Party-Id: <sip:7973673265@xx.245.6.81>;screen=yes;privacy=off
Call-ID: 1291655-3517057571-578987@MSX11.test.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP xx.245.6.81:5060;x-route-tag="tgrp:test-INGRESS-TRKGRP";branch=z9hG4bK84c47011c63d2022d41708c4fea692b3
Contact: <sip:07973673265@xx.245.6.81:5060>
Expires: 180
Call-Info: <sip:xx.245.6.81>;method="NOTIFY;Event=telephone-event;Duration=1000"
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 260
v=0
o=MSX11 5577 7987 IN IP4 xx.245.6.81
s=sip call
c=IN IP4 xx.245.6.82
t=0 0
m=audio 36202 RTP/AVP 18 100 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
809663mS SIP Rx: UDP xx.245.6.81:5060 -> xxx.231.216.140:5060
INVITE sip:02080901280@xxx.231.216.140 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:02080901280@xx.245.6.81:5060>
From: <sip:07973673265@xx.245.6.81>;tag=3517057571-578992
Remote-Party-Id: <sip:7973673265@xx.245.6.81>;screen=yes;privacy=off
Call-ID: 1291655-3517057571-578987@MSX11.test.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP xx.245.6.81:5060;x-route-tag="tgrp:test-INGRESS-TRKGRP";branch=z9hG4bK84c47011c63d2022d41708c4fea692b3
Contact: <sip:07973673265@xx.245.6.81:5060>
Expires: 180
Call-Info: <sip:xx.245.6.81>;method="NOTIFY;Event=telephone-event;Duration=1000"
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 260
v=0
o=MSX11 5577 7987 IN IP4 xx.245.6.81
s=sip call
c=IN IP4 xx.245.6.82
t=0 0
m=audio 36202 RTP/AVP 18 100 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Trunk says out of service.
Have configured as per this document as well:
It just will not work, there have been issues with them but they say that all is well their end.
Any help would be greatly appreciated.
Many thanks
Desperate Dan