Cisco SPA504G - 503 Service unavailable for incoming call

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michaelsalzburg

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Hi,

we have a 3CX pbx V10 and are using Cisco SPA504G with firmware 7.5.2b.

Since a few weeks we have troubles with phones that are connected via the 3CX tunnel proxy (proxy manager). After starting/rebooting the phone everything works fine. But after some time (sometimes 10 minutes, sometimes 24 hours) incoming calls don't work any more. In the 3CX log the error "503 Service unavailable" is listed. Rebooting the phone solves the problem temporarly, but there should be a better solution.
Does anyone have an idea how to solve it?

3CX-Log:
09:16:22.927 [CM503002]: Call(180): Alerting sip:[email protected]:40600;rinstance=6447478e8f7a9981
09:16:22.698 [CM503025]: Call(180): Calling Unknown:Ext.EndCall@[Dev:sip:[email protected]:40600;rinstance=6447478e8f7a9981]
09:16:22.696 [MS210004] C:180.3:Offer provided. Connection(proxy mode): 127.0.0.1:7334(7335)
09:16:22.686 [CM503016]: Call(180): Attempt to reach <sip:[email protected]:5060> failed. Reason: Server Failure
09:16:22.684 [CM503003]: Call(180): Call to sip:[email protected] has failed; Cause: 503 Service Unavailable; warning: ; from IP:192.168.0.10:5080
09:16:21.093 [CM503025]: Call(180): Calling Ext:Ext.102@[Dev:sip:[email protected]:5060;transport=tcp]
09:16:21.091 [MS210002] C:180.2:Offer provided. Connection(transcoding mode): 192.168.0.10:7332(7333)
09:16:21.059 [CM503004]: Call(180): Route 1: Ext:Ext.102@[Dev:sip:[email protected]:5060;transport=tcp]
09:16:21.057 [CM503010]: Making route(s) to <sip:[email protected]:5060>
09:16:21.057 [MS210000] C:180.1:Offer received. RTP connection: 192.168.0.9:5000(5001)
09:16:21.055 Remote SDP is set for legC:180.1
09:16:21.051 [CM503001]: Call(180): Incoming call from 00664625xxxx@(Ln.10001@beronet Gateway Office) to <sip:[email protected]:5060>
09:16:21.035 [CM503012]: Inbound any hours rule (unnamed) for 10001 forwards to DN:102
09:16:21.035 Looking for inbound target: called=102; caller=00664625xxxx
09:16:21.032 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9;rport=5060;branch=z9hG4bKN5tNmjFyBKryH
Max-Forwards: 70
Contact: <sip:[email protected];transport=udp>
To: <sip:[email protected]>
From: "00664625xxxx"<sip:[email protected]>;tag=ZZp3H2v8B5vHe
Call-ID: 4c003f50-7103-1230-00a4-bd3ab2ba8fb2
CSeq: 33035433 INVITE
Min-SE: 120
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, REGISTER
Proxy-Authorization: Digest username="10001",realm="3CXPhoneSystem",nonce="414d535c06563bc478:1777ed4d72c5f98f8369afd016ba652c",algorithm=MD5,uri="sip:[email protected]",response="041aa583c6a83aada8b6d925eabcb71f"
Supported: timer, 100rel, replaces
User-Agent: Berofix VOIP Gateway (2.2)
Privacy: none
P-Asserted-Identity: "" <sip:[email protected];user=phone>
P-Preferred-Identity: "" <sip:[email protected];user=phone>
Content-Length: 0
Remote-Party-ID: "" <sip:[email protected];user=phone>;party=calling;privacy=off;screen=no

3CX-Tunnel-Log:
.\Bridge.cpp(419) : {09:14:53.900}(Trace5):Got SIP msg
brief=SipReq: INVITE [email protected]:5060 tid=a636946e8d0a137e cseq=INVITE [email protected]:5060 / 1 from(tu)
from tunnel's connection 93.83.20.182:5090
.\Bridge.cpp(434) : {09:14:53.900}(Trace5):Selected SIP stack on 10.10.0.1:5080
.\Bridge.cpp(721) : {09:14:53.900}(Trace5):Process tunnel request: INVITE
.\SipCore.cpp(430) : {09:14:53.900}(Trace5):Started new transaction: SipReq: INVITE [email protected]:5060 tid=a636946e8d0a137e cseq=INVITE [email protected]:5060 / 1 from(tu)
.\SipCore.cpp(472) : {09:14:53.900}(Trace5):DS:setOffer(tunnel)
.\SipCore.cpp(509) : {09:14:53.900}(Trace5):Set tunnel SDP
.\RTPSession.cpp(1038) : {09:14:53.900}(Trace5):SetTunnelSdp: modified SDP:
v=0
o=3cxPS 164483825664 130593849345 IN IP4 192.168.0.10
s=3cxPS Audio call
c=IN IP4 10.10.0.1
t=0 0
m=audio 10234 RTP/AVP 0 8 3 13 9 18 110 99 101
c=IN IP4 10.10.0.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 iLBC/8000
a=rtpmap:99 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

.\RTPSession.cpp(1346) : {09:14:53.900}(Trace5):NewOfferSockets:
[472,612]
newOfferAddresses:
[192.168.0.10:7332,192.168.0.10:7333]
newOfferMediaPorts:
[]
registeredSockets:
[]
registeredAddresses:
[]
sockToData:
[]
localToSock:
[]

.\Stack.cpp(162) : {09:14:53.900}(Trace5):Sending to stack(5080);
brief=SipReq: INVITE [email protected]:5060 tid=a636946e8d0a137e cseq=INVITE [email protected]:5060 / 1 from(tu)
.\Stack.cpp(102) : {09:14:54.947}(Trace5):Received from local SIP stack(5080):
brief=SipResp: 503 tid=a636946e8d0a137e cseq=INVITE / 1 from(wire)
.\Bridge.cpp(305) : {09:14:54.947}(Trace5):Processing SIP message (tid=a636946e8d0a137e) from SIP stack 10.10.0.1:5080
.\Bridge.cpp(668) : {09:14:54.947}(Trace5):Process local response: 503 Service Unavailable / INVITE
.\SipCore.cpp(704) : {09:14:54.947}(Trace5):DS:offerRejected
.\RTPSession.cpp(1228) : {09:14:54.947}(Trace5):EndOfSession
.\RTPSession.cpp(1228) : {09:14:54.947}(Trace5):EndOfSession
 
Re: Cisco SPA504G - 503 Service unavailable for incoming cal

Hi Michael
You refer to phone and phones ... So I'm not sure if this is a one- or many-phone problem??

If you have multiple SPA504g phones at the remote site all working through the same Proxy tunnel - it might help if you stagger the SIP ports that each SPA504G extension is using ... i.e. each phone will have extension 1 working on 5060 by default, so all phones will be working the same SIP port.

Try putting ext.1 of the first phone on 5061, ext.1 of the second phone on 5062, ext.1 of the third phone on 5063 etc. so that all working extensions have different SIP port settings (same for extensions 2, 3, or 4 if you have multiple extensions on the phones). You can find this 'SIP Port:' setting on each extension setting page of the 504G.

best regards
Graham Hill
Foxhall Solutions
 
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