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SIP 503 Service Unavailable

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VinGtech

VinGtech

Systems Engineer
Feb 14, 2022
10
KE
Hi Guys,

I have created a SIP TRUNK between a GSM Gateway and Avaya IP Office and its registered and In IDLE state.

Having issues making an Inbound and Outbound calls where I get the error "SIP 503 Service Unavailable" via SIP Trunk.

The Trunk is in service and in IDLE state, but can get to route calls.



Below is the Log form system status
--------------------------------------------------------------

2/20/22 10:59:42 AM-834ms Line = 3, Channel = 1, SIP Message = Response, Direction = From Switch, From = admin@192.168.25.11, To = 2555@192.168.25.11, Response = 503 Service Unavailable
2/20/22 10:59:42 AM-866ms Line = 3, Channel = 1, SIP Message = Ack, Direction = To Switch, From = admin@192.168.25.11, To = 2555@192.168.25.11

This error pops up as well : 11:04:30 312383373mS CMMap: IP::SetCodec pcp[697]b0r0 0 -> f6805d98

Logs from system Monitor

--------------------------------------
11:04:29 312382592mS PRN: File unchanged, returning current value 7070
11:04:30 312383054mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
11:04:30 312383373mS SIP Rx: UDP 192.168.25.10:5060 -> 192.168.25.13:5060
INVITE sip:2555@192.168.25.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.25.10;branch=z9hG4bKefc645fb34c603f7be4b50ce103a849e;rport
From: 0710284680 <sip:admin@192.168.25.11:5060;user=phone>;tag=3b8ee9637b7e4ffcbfd61144a1c5b3f1
To: <sip:2555@192.168.25.11:5060>
Call-ID: 394aca57e88ff79081f1412c1d2d81ad@192.168.25.10
CSeq: 18949 INVITE
Contact: <sip:admin@192.168.25.10>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, NOTIFY, REFER
Content-Type: application/sdp
Max-Forwards: 70
Content-Length: 248

v=0
o=call 395926 395927 IN IP4 192.168.25.10
s=-
c=IN IP4 192.168.25.10
t=0 0
m=audio 8000 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=ptime:20
11:04:30 312383373mS CMCallEvt: 0000000000000000 0.1269.0 -1 BaseEP: NEW CMEndpoint f683f408 TOTAL NOW=1 CALL_LIST=0
11:04:30 312383373mS CMMap: IP::SetCodec pcp[697]b0r0 0 -> f6805d98
11:04:30 312383373mS SIP Tx: UDP 192.168.25.13:5060 -> 192.168.25.10:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.25.10;branch=z9hG4bKefc645fb34c603f7be4b50ce103a849e;rport
From: 0710284680 <sip:admin@192.168.25.11:5060;user=phone>;tag=3b8ee9637b7e4ffcbfd61144a1c5b3f1
Call-ID: 394aca57e88ff79081f1412c1d2d81ad@192.168.25.10
CSeq: 18949 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.0.0.0 build 237
Reason: Q.850;cause=41;text="Temporary failure"
Content-Length: 0
To: <sip:2555@192.168.25.11:5060>;tag=cfa4139d94623715

11:04:30 312383395mS SIP Rx: UDP 192.168.25.10:5060 -> 192.168.25.13:5060
ACK sip:2555@192.168.25.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.25.10;branch=z9hG4bKefc645fb34c603f7be4b50ce103a849e;rport
From: 0710284680 <sip:admin@192.168.25.11:5060;user=phone>;tag=3b8ee9637b7e4ffcbfd61144a1c5b3f1
To: <sip:2555@192.168.25.11:5060>;tag=cfa4139d94623715
Call-ID: 394aca57e88ff79081f1412c1d2d81ad@192.168.25.10
CSeq: 18949 ACK
Content-Length: 0

11:04:31 312384396mS CMCallEvt: c0a8190d000004f5 3.1269.1 -1 SIPTrunk Endpoint: StateChange: END=X CMCSIdle->CMCSDelete
11:04:31 312384396mS CMTARGET: c0a8190d000004f5 3.1269.1 -1 BaseEP: ~CMTargetHandler f68299f8 ep f683f408
11:04:31 312384396mS CMCallEvt: c0a8190d000004f5 3.1269.1 -1 BaseEP: DELETE CMEndpoint f683f408 TOTAL NOW=0 CALL_LIST=0
11:04:31 312384396mS CMMap: a=1.175 b=0.0 Mapper::FreeCodec freed CMRTVocoder(f6805d98) resource busy 0, total 3520
11:04:35 312388054mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
11:04:36 312389014mS PRN: File unchanged, returning current value 7070

********** Warning: Logging to Screen Stopped **********

 
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ICE_gbziem.png
call_details_isqod6.png
 
Upvote 0 Downvote
Why are you using auto for Local URI and PAI but 2555 for contact? How many digits is the provider expecting? Have you asked the provider why they sending a 503? Its must easier for them to tell you exactly why they are returning a 503 then us guessing.

The truth is just an excuse for lack of imagination.
 
Upvote 0 Downvote
I was just trying various options.

Initially I used auto, then I used internal data but the error is still the same.

503 Service Unavailable
 
Upvote 0 Downvote
Each one does something different. If you use internal data did you set the SIP tab for the user to a full DID used for CLI? If you used auto did you set the CLI via the shortcode? Have you asked the provider why they are sending a 503?

The truth is just an excuse for lack of imagination.
 
Upvote 0 Downvote

The source for the SIP Trunk is a GSM Gateway Box, created a SIP Trunk to Avaya IPO Server.

Can you please share the shortcode format?I configured internal data and it did not work.

 
Upvote 0 Downvote
From your own SIP trace:

From: 0710284680 <sip:admin@192.168.25.11:5060;user=phone>;tag=3b8ee9637b7e4ffcbfd61144a1c5b3f1
To: <sip:2555@192.168.25.11:5060>

This sure looks like you are still using internal data and have not setup the SIP tab for the user so its just using the user name of admin which will not work. You are also only sending 4 digits to the provider are you sure they accept 4 digits?

If using internal data you need to go to the SIP tab for the user that is calling out and setup the information there. If you are using auto you need to setup a shortcode:
9N
Dial
Nsi8005551212
50main

After making these changes run another trace. You should see the from being from a valid phone number not admin.

The truth is just an excuse for lack of imagination.
 
Upvote 0 Downvote
critchey

I can now see the number being sent as per below SIP trace, however I am still getting the same error. Where am I going wrong

I am using Internal data, on the SIP tab for the user what do I need to use? its currently setup as the extension number

17:28:02 680994073mS SIP Rx: UDP 192.168.25.102:5060 -> 192.168.25.13:5060
INVITE sip:2605@192.168.25.13:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.25.102:5060;branch=z9hG4bK37ab8c20;rport
Max-Forwards: 70
From: "0710284680" <sip:0710284680@192.168.25.102>;tag=as0c48cd82
To: <sip:2605@192.168.25.13:5060>
Contact: <sip:0710284680@192.168.25.102>
Call-ID: 19e0270e5f8cd5ca73549dce6c68e317@192.168.25.102
CSeq: 102 INVITE
Date: Thu, 24 Feb 2022 14:27:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 60913291 60913291 IN IP4 192.168.25.102
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.25.102
t=0 0
m=audio 11892 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
17:28:02 680994073mS CMCallEvt: 0000000000000000 0.1861.0 -1 BaseEP: NEW CMEndpoint f681d830 TOTAL NOW=1 CALL_LIST=0
17:28:02 680994073mS CMMap: IP::SetCodec pcp[798]b0r0 0 -> f6815538
17:28:02 680994073mS SIP Tx: UDP 192.168.25.13:5060 -> 192.168.25.102:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.25.102:5060;branch=z9hG4bK37ab8c20;rport
From: "0710284680" <sip:0710284680@192.168.25.102>;tag=as0c48cd82
Call-ID: 19e0270e5f8cd5ca73549dce6c68e317@192.168.25.102
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 11.1.0.0.0 build 237
Reason: Q.850;cause=41;text="Temporary failure"
Content-Length: 0
To: <sip:2605@192.168.25.13:5060>;tag=78d3b0ea2efbfd44

17:28:02 680994074mS SIP Rx: UDP 192.168.25.102:5060 -> 192.168.25.13:5060
ACK sip:2605@192.168.25.13:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.25.102:5060;branch=z9hG4bK37ab8c20;rport
Max-Forwards: 70
From: "0710284680" <sip:0710284680@192.168.25.102>;tag=as0c48cd82
To: <sip:2605@192.168.25.13:5060>;tag=78d3b0ea2efbfd44
Contact: <sip:0710284680@192.168.25.102>
Call-ID: 19e0270e5f8cd5ca73549dce6c68e317@192.168.25.102
CSeq: 102 ACK
Content-Length: 0
 
Upvote 0 Downvote
Looking at the trace above, this is an inbound call from an Asterisk PBX to the IP Office, and it's the IP Office sending the 503.

Have you got an Incoming Call route set up for calls to "2605" on that SIP trunk?

“Some humans would do anything to see if it was possible to do it.
If you put a large switch in some cave somewhere, with a sign on it saying 'End-of-the-World Switch. PLEASE DO NOT TOUCH'.
The paint wouldn't even have time to dry.”

Terry Pratchet
 
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