TelecomsTim
ISP
We are currently implementing a Server Edition system alongside an Avaya SBC and Hipcom SIP trunks.
After much playing around with the SBC we finally got calls to route in and out however incoming calls are dropping after 32 seconds. We have full speech path during those 32 seconds that the call is connected and outbound calls across the SIP are working perfectly.
As you will see below, the phone system is sending the BYE request and closing the call down.
(I have blanked out the numbers involved in the call for obvious reasons)
18:31:00 97196mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.254:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.254:5060;branch=z9hG4bK-s1632-000605106642-1--s1632-
Record-Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: <sip:07XXXXXXXXX@192.168.42.254:5060;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165562 INVITE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.3.0 build 941
To: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
Content-Type: application/sdp
Content-Length: 203
v=0
o=UserA 962444212 1033067916 IN IP4 192.168.42.1
s=Session SDP
c=IN IP4 192.168.42.1
t=0 0
m=audio 49154 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
18:31:02 99196mS SIP Call Tx: 17
BYE sip:07XXXXXXXXX@192.168.42.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK8f0751d9ec208fdca144a2915c56fa84
Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
To: <sip:07XXXXXXXXX@192.168.42.254;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165563 BYE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.3.0 build 941
Content-Length: 0
18:31:02 99196mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.254:5060
BYE sip:07XXXXXXXXX@192.168.42.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK8f0751d9ec208fdca144a2915c56fa84
Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
To: <sip:07XXXXXXXXX@192.168.42.254;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165563 BYE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.3.0 build 941
Content-Length: 0
18:31:02 99196mS Sip: SIPDialog f0627958 deleted, dialogs 0
18:31:02 99196mS CMCallEvt: 17.1004.1 1 SIPTrunk Endpoint: StateChange: END=A CMCSOGConnReq->CMCSCompleted
18:31:02 99196mS CMExtnTx: v=3501, p1=0
I also came across this on a Askerisk forum which pretty much matches my exact issue but I wondered in any of you guys have seen this on an IPO and how you resolved it.
I have an IP route setup on the system to route the call to the SIP provider using the SBC as the gateway.
We also have an issue where we receive silence when calling into voicemail which is likley to be linked to the issue above.
Any support would be kindly appreciated
After much playing around with the SBC we finally got calls to route in and out however incoming calls are dropping after 32 seconds. We have full speech path during those 32 seconds that the call is connected and outbound calls across the SIP are working perfectly.
As you will see below, the phone system is sending the BYE request and closing the call down.
(I have blanked out the numbers involved in the call for obvious reasons)
18:31:00 97196mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.254:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.254:5060;branch=z9hG4bK-s1632-000605106642-1--s1632-
Record-Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: <sip:07XXXXXXXXX@192.168.42.254:5060;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165562 INVITE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.3.0 build 941
To: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
Content-Type: application/sdp
Content-Length: 203
v=0
o=UserA 962444212 1033067916 IN IP4 192.168.42.1
s=Session SDP
c=IN IP4 192.168.42.1
t=0 0
m=audio 49154 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
18:31:02 99196mS SIP Call Tx: 17
BYE sip:07XXXXXXXXX@192.168.42.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK8f0751d9ec208fdca144a2915c56fa84
Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
To: <sip:07XXXXXXXXX@192.168.42.254;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165563 BYE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.3.0 build 941
Content-Length: 0
18:31:02 99196mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.254:5060
BYE sip:07XXXXXXXXX@192.168.42.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK8f0751d9ec208fdca144a2915c56fa84
Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
To: <sip:07XXXXXXXXX@192.168.42.254;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165563 BYE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.3.0 build 941
Content-Length: 0
18:31:02 99196mS Sip: SIPDialog f0627958 deleted, dialogs 0
18:31:02 99196mS CMCallEvt: 17.1004.1 1 SIPTrunk Endpoint: StateChange: END=A CMCSOGConnReq->CMCSCompleted
18:31:02 99196mS CMExtnTx: v=3501, p1=0
I also came across this on a Askerisk forum which pretty much matches my exact issue but I wondered in any of you guys have seen this on an IPO and how you resolved it.
I have an IP route setup on the system to route the call to the SIP provider using the SBC as the gateway.
We also have an issue where we receive silence when calling into voicemail which is likley to be linked to the issue above.
Any support would be kindly appreciated