Possible causes of error "503 Service Unavailable"

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Possible causes of error "503 Service Unavailable"

Postby jhepoy » Fri Apr 08, 2005 6:15 pm

We're currently trying to interconnect the Asterisk PBX with a ZTE Softswitch. The scenario is this, using the account and password provisioned in the ZTE Softswitch, I created a SIP Trunk from the Asterisk Box, then using an extension, we tried making a PSTN call going to the ZTE SIP Trunk, but apparently an error message "503 Service Unavailable" was the result.

It was verified on the ZTE side that the account has registered successfully.

Any help on this.

Thanks.
jhepoy
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Postby zmanea » Fri Apr 08, 2005 9:31 pm

It sounds like the ZTE switch is getting the call but does not know what to do with the extension specified. Watch the Asterisk command line when you make a call and look for a line like this: -- Called 1234567@peer . Does the ZTE switch know what to do with the extension (in my example it is 1234567) from Asterisk? Is the extension in the correct context/calling-search-space on the ZTE switch?
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Postby jhepoy » Sat Apr 09, 2005 3:03 am

here is the trace log from the asterisk. According to the ZTE operator that they were able to receive an incoming call request but the message they are seeing are "unauthorized".

Please find below the logs, thanks for the help.

9005 is my extension, 7192371 is the called pstn number.
203.177.174.172 is teh extionsion IP Address

Code: Select all
Apr  9 09:55:47 DEBUG[20507]: Setting NAT on RTP to 4
Apr  9 09:55:48 DEBUG[20507]: Stopping retransmission on '91023D22-3F72-4355-864E-32E311F63695@203.177.174.172' of Response 27996: Found
Apr  9 09:55:48 DEBUG[20507]: Setting NAT on RTP to 4
Apr  9 09:55:48 DEBUG[20507]: Check for res for 9005
Apr  9 09:55:48 DEBUG[20507]: Call from user '9005' is 1 out of 0
Apr  9 09:55:48 DEBUG[20507]: build_route: Contact hop: <sip:9005@203.177.174.172:5060>
Apr  9 09:55:48 VERBOSE[20507]:     -- Executing Macro("SIP/9005-bb7e", "dialout|9|17192371") in new stack
Apr  9 09:55:48 DEBUG[20507]: Expression is '1'
Apr  9 09:55:48 VERBOSE[20507]:     -- Executing GotoIf("SIP/9005-bb7e", "1?4") in new stack
Apr  9 09:55:48 VERBOSE[20507]:     -- Goto (macro-dialout,s,4)
Apr  9 09:55:48 DEBUG[20507]: Expression is '0'
Apr  9 09:55:48 VERBOSE[20507]:     -- Executing GotoIf("SIP/9005-bb7e", "0?6") in new stack
Apr  9 09:55:48 DEBUG[20507]: Not taking any branch
Apr  9 09:55:48 VERBOSE[20507]:     -- Executing SetCallerID("SIP/9005-bb7e", "Digitel_3959012") in new stack
Apr  9 09:55:48 VERBOSE[20507]:     -- Executing SetVar("SIP/9005-bb7e", "length=1") in new stack
Apr  9 09:55:48 VERBOSE[20507]:     -- Executing Dial("SIP/9005-bb7e", "SIP/Digitel_3959012/7192371") in new stack
Apr  9 09:55:48 DEBUG[20507]: Setting NAT on RTP to 0
Apr  9 09:55:48 DEBUG[20507]: Outgoing Call for 7192371
Apr  9 09:55:48 DEBUG[20507]: 7192371 is not a local user
Apr  9 09:55:48 VERBOSE[20507]:     -- Called Digitel_3959012/7192371
Apr  9 09:55:49 DEBUG[20507]: Scheduled a registration timeout # 76814
Apr  9 09:55:49 DEBUG[20507]: Stopping retransmission on '5eea859b7392efa413ecd55f669bb551@127.0.0.1' of Request 710: Found
Apr  9 09:55:49 DEBUG[20507]: Stopping retransmission on '5eea859b7392efa413ecd55f669bb551@127.0.0.1' of Request 711: Found
Apr  9 09:55:49 DEBUG[20507]: Registration successful
Apr  9 09:55:49 DEBUG[20507]: Cancelling timeout 76814
Apr  9 09:55:54 WARNING[20507]: Maximum retries exceeded on call 5270a4bc621ecef83fe2662918d09436@202.73.169.68 for seqno 102 (Critical Request)
Apr  9 09:55:54 DEBUG[20507]: update_user_counter(7192371) - decrement outUse counter
Apr  9 09:55:54 DEBUG[20507]: 7192371 is not a local user
Apr  9 09:55:54 VERBOSE[20507]:   == No one is available to answer at this time
Apr  9 09:55:54 DEBUG[20507]: Exiting with DIALSTATUS=NOANSWER.
Apr  9 09:55:54 VERBOSE[20507]:     -- Executing Congestion("SIP/9005-bb7e", "") in new stack
Apr  9 09:55:54 VERBOSE[20507]:   == Spawn extension (macro-dialout, s, 8) exited non-zero on 'SIP/9005-bb7e' in macro 'dialout'
Apr  9 09:55:54 VERBOSE[20507]:   == Spawn extension (from-internal, 17192371, 1) exited non-zero on 'SIP/9005-bb7e'
Apr  9 09:55:54 VERBOSE[20507]:     -- Executing Macro("SIP/9005-bb7e", "hangupcall") in new stack
Apr  9 09:55:54 VERBOSE[20507]:     -- Executing ResetCDR("SIP/9005-bb7e", "w") in new stack
Apr  9 09:55:54 DEBUG[20507]: cdr_mysql: inserting a CDR record.
Apr  9 09:55:54 DEBUG[20507]: cdr_mysql: SQL command as follows:  INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2005-04-09 09:55:48','Digitel_3959012','9005','17192371','from-internal', 'SIP/9005-bb7e','SIP/Digitel_3959012-cd8e','ResetCDR','w',6,0,'NO ANSWER',3,'')
Apr  9 09:55:54 VERBOSE[20507]:     -- Executing NoCDR("SIP/9005-bb7e", "") in new stack
Apr  9 09:55:54 WARNING[20507]: CDR on channel 'SIP/9005-bb7e' not posted
Apr  9 09:55:54 WARNING[20507]: CDR on channel 'SIP/9005-bb7e' lacks end
Apr  9 09:55:54 VERBOSE[20507]:     -- Executing Wait("SIP/9005-bb7e", "5") in new stack
Apr  9 09:55:54 VERBOSE[20507]:   == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/9005-bb7e' in macro 'hangupcall'
Apr  9 09:55:54 VERBOSE[20507]:   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/9005-bb7e'
Apr  9 09:55:54 DEBUG[20507]: update_user_counter(9005) - decrement inUse counter
Apr  9 09:55:54 DEBUG[20507]: Stopping retransmission on '91023D22-3F72-4355-864E-32E311F63695@203.177.174.172' of Response 27997: Not Found
jhepoy
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Postby zmanea » Sat Apr 09, 2005 8:28 am

Is Asterisk or the ZTE switch behind a firewall? I think you are running into NAT problems. Can any other SIP devices, on the same network as Asterisk, make calls to the ZTE switch?
zmanea
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Postby jhepoy » Sun Apr 10, 2005 8:16 am

Hi, i've tested the same account from a different asterisk server and same result (503 service unavailable). With regards to NAT, i've tested the sip account provided by ZTE directly using a softphone (xlite and sjphone) and it works fine. Tested it with a public ip and a private ip too.

Thanks for the help.

Any idea of what's the problem here?
jhepoy
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Posts: 6
Joined: Thu Dec 21, 2006 10:56 pm

Postby zmanea » Mon Apr 11, 2005 10:24 am

So the Asterisk server is behind a NAT device? If so try adding externip &
localnet definitions to your sip.conf
zmanea
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Hi Jhepoy

Postby alkaline » Tue Mar 27, 2007 10:11 pm

Yes your registration is OK but your DIAL peer is not defined correctly at ZTE side.
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