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SIP/2.0 503 Service Unavailable 2

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trinetintl

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Mar 15, 2010
730
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Hi everybody, I have a customer on an IPO 500 V2 Rel 9.0.2.0 build 941 (according to monitor), the customer has SIP trunks delivered almost like a PRI on its own vendor provided session border controller and since the 29th at around 11:00 PM system status shows Trunk out of Service. Thankfully the SIP service is not in full production yet and we have not changed anything on our configuration. The carrier keeps insisting that it is our problem that everything will be back to normal once we clear the error because when you make an inbound call they get a an error that I assume is 503 as a reply to the invite. I do not see any traffic on SSA, nothing, simply Trunk Out of Service however while doing an inbound test call I did notice some traffic associated with the SIP and the call in Monitor where I see the error "SIP/2.0 503 Service Unavailable". That was a big surprise because it confirms the carrier's technical input but I do not know where else to look.

The setup/connection is very simple: Carrier provided Session Border Controller Box that has IP 192.168.43.1, connected to LAN2 on the IP office at 192.168.43.1, no gateways ow switches involved, nothing in the middle, LAN to LAN connection between devices.

Is this something that I need to check on our side, something that I may have inadvertently changed on the configuration? I double checked everything and in my mind we are good to go and the carrier is the one with the issue.

14:48:02 662835634mS SIP Rx: UDP 192.168.43.2:5060 -> 192.168.43.1:5060
INVITE sip:9565553333@192.168.43.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.2:5060;branch=z9hG4bK-d8754z-63c0c80a139f1bf2-1---d8754z-;rport
Max-Forwards: 68
Contact: <sip:19565554444@192.168.43.2:5060>
To: <sip:9565553333@192.168.43.2>
From: "Test Caller"<sip:19565554444@192.168.43.2>;tag=3239bdec
Call-ID: OWVhODQyNDU1ZDUyMjBkMThlZWNhNjFjZGJmYzg2ZGY.
CSeq: 1 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REGISTER, SUBSCRIBE, UPDATE
Content-Type: application/sdp
User-Agent: ESBC9378-4B-2.0.13.0-OR
Privacy: none
Content-Length: 206
History-Info: <sip:+19565553333@ims.rr.com;user=phone>;index=1

v=0
o=ESBC 2 0 IN IP4 192.168.43.2
s=ESBC
c=IN IP4 192.168.43.2
t=0 0
m=audio 62028 RTP/AVP 0 101
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
14:48:02 662835638mS SIP Tx: UDP 192.168.43.1:5060 -> 192.168.43.2:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.43.2:5060;branch=z9hG4bK-d8754z-63c0c80a139f1bf2-1---d8754z-;rport
From: "Test Caller" <sip:19565554444@192.168.43.2>;tag=3239bdec
Call-ID: OWVhODQyNDU1ZDUyMjBkMThlZWNhNjFjZGJmYzg2ZGY.
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.3.0 build 941
Reason: Q.850;cause=41;text="Temporary failure"
To: <sip:9565553333@192.168.43.2>
Content-Length: 0

14:48:02 662835646mS SIP Rx: UDP 192.168.43.2:5060 -> 192.168.43.1:5060
ACK sip:9565553333@192.168.43.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.2:5060;branch=z9hG4bK-d8754z-63c0c80a139f1bf2-1---d8754z-;rport
To: <sip:9565553333@192.168.43.2>
From: "Test Caller"<sip:19565554444@192.168.43.2>;tag=3239bdec
Call-ID: OWVhODQyNDU1ZDUyMjBkMThlZWNhNjFjZGJmYzg2ZGY.
CSeq: 1 ACK
Content-Length: 0

RE
APSS - SME
ACIS - SME
 
So what was your config error? I have a similar issue happeing and looking for ideas on what might be wrong.

Thanks
 
I've seen this issue a few times on server edition. There's a field under system->telephony where you need to set the maximum number of sip sessions for each primary/secondary/expansion system, otherwise you get the 503.

I have a vague recollection of if you have the trace option for sip set to terse, you'll also see something along the lines of license count =0 or license unavailable.

 
Carrier provided Session Border Controller Box that has IP 192.168.43.1, connected to LAN2 on the IP office at 192.168.43.1.
Was this the config issue? It would have been.

Dulce et decorum pro Avaya mori
 
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