503 Server Error

Asterisk Asterisk Support
1

Sometimes Asterisk returns “503 Server Error” after it receives BYE command (and always when receiving OPTIONS, but that’s not that annoying at this very moment). This however results in calls stuck in the queue long after they were already terminated by the softphone client.

Does anyone have any ideas, why this is happening?

Thanks for suggestions in advance!

Here is the log (see “503 Server Error” at the bottom):

[code]>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (2765 Ms, To: 111.111.111.5:5060) >>>>
INVITE sip:991@111.111.111.5 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.164:5060
From: From101 sip:From101@111.111.111.5;tag=b50e036
To: sip:991@111.111.111.5
Contact: sip:From101@111.111.111.5:5060
Call-Id: 7aae81a4-294c-40d6-97ee-6a32893a12cc-00000c44@111.111.111.164
CSeq: 189874943 INVITE
Max-Forwards: 70
Organization: A0F5342B-C93D-44FD-8483-423CAB810C55
Content-Length: 210
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment.)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=From101 189847843 189847843 IN IP4 111.111.111.164
s=LanScape
c=IN IP4 111.111.111.164
t=0 0
m=audio 8002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (2781 Ms, From: 111.111.111.5:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.111.164:5060;received=111.111.111.164
From: From101 sip:From101@111.111.111.5;tag=b50e036
To: sip:991@111.111.111.5
Call-ID: 7aae81a4-294c-40d6-97ee-6a32893a12cc-00000c44@111.111.111.164
CSeq: 189874943 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:991@111.111.111.5
Content-Length: 0

<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (16 Ms, From: 111.111.111.5:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.164:5060;received=111.111.111.164
From: From101 sip:From101@111.111.111.5;tag=b50e036
To: sip:991@111.111.111.5;tag=as7b3ddc9b
Call-ID: 7aae81a4-294c-40d6-97ee-6a32893a12cc-00000c44@111.111.111.164
CSeq: 189874943 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:991@111.111.111.5
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 4267 4267 IN IP4 111.111.111.5
s=session
c=IN IP4 111.111.111.5
t=0 0
m=audio 16792 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (94 Ms, To: 111.111.111.5:5060) >>>>
ACK sip:991@111.111.111.5 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.164:5060;received=111.111.111.164
From: From101 sip:From101@111.111.111.5;tag=b50e036
To: sip:991@111.111.111.5;tag=as7b3ddc9b
Call-Id: 7aae81a4-294c-40d6-97ee-6a32893a12cc-00000c44@111.111.111.164
CSeq: 189874943 ACK
Max-Forwards: 70
Route: sip:991@111.111.111.5
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment.)
Content-Length: 0

TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (7469 Ms, To: 111.111.111.5:5060) >>>>
OPTIONS sip:991@111.111.111.5 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.164:5060
From: From101 sip:From101@111.111.111.164
To: 991 sip:991@111.111.111.5
Call-Id: 7aae81a4-294c-40d6-97ee-6a32893a12cc-00000c44@111.111.111.164
CSeq: 189845293 OPTIONS
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment.)
Content-Length: 0

<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (7531 Ms, From: 111.111.111.5:5060) <<<<
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 111.111.111.164:5060;received=111.111.111.164
From: From101 sip:From101@111.111.111.164
To: 991 sip:991@111.111.111.5;tag=as7b3ddc9b
Call-ID: 7aae81a4-294c-40d6-97ee-6a32893a12cc-00000c44@111.111.111.164
CSeq: 189845293 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:991@111.111.111.5
Content-Length: 0

TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (4781 Ms, To: 111.111.111.5:5060) >>>>
BYE sip:991@111.111.111.5 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.164:5060
From: From101 sip:From101@111.111.111.5;tag=b50e036
To: sip:991@111.111.111.5;tag=as7b3ddc9b
Call-Id: 7aae81a4-294c-40d6-97ee-6a32893a12cc-00000c44@111.111.111.164
CSeq: 189867348 BYE
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment.)
Content-Length: 0

<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (4782 Ms, From: 111.111.111.5:5060) <<<<
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 111.111.111.164:5060;received=111.111.111.164
From: From101 sip:From101@111.111.111.5;tag=b50e036
To: sip:991@111.111.111.5;tag=as7b3ddc9b
Call-ID: 7aae81a4-294c-40d6-97ee-6a32893a12cc-00000c44@111.111.111.164
CSeq: 189867348 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:991@111.111.111.5
Content-Length: 0[/code]

2

Hi,

I don’t know if you are still in trouble since your post is old but it may help others anyway.

This “503 Server Error” can occur when the CSeq field has not been incremented correctly.
You can notice than the CSeq from the invite is a higher number than the one from your BYE request. That should not happen, I think.

HTH

Guillaume
www.blueface.ie