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Beginner

SIP/2.0 500 Server Internal Error

Hi guys

 

I'm not able to make outbound calls from my CME SIP gateway , below is the debug ccsip messages output, I'm getting a SIP/2.0 500 Server Internal Error as shown in the output below (IPs changed for security reasons), below the ccsip output is my outbound SIP config

 

INVITE sip:+254725101240@10.10.10.10:5060 SIP/2.0
Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bK2CDAE5
Remote-Party-ID: "Lounge" <sip:+254709655000@20.20.20.20>;party=calling;screen=no;privacy=off
From: "Lounge" <sip:+254709655000@20.20.20.20>;tag=A5AAD12C-19F2
To: <sip:+254725101240@10.10.10.10>
Date: Tue, 24 Jul 2018 14:51:29 EAT
Call-ID: BCA9FD88-8E6E11E8-B5FFC93F-47B3019@20.20.20.20
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 86400
Cisco-Guid: 3046617042-2389578216-3053111615-0075182105
User-Agent: Cisco-SIPGateway/IOS-15.6.3.M2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1532433089
Contact: <sip:+254709655000@20.20.20.20:5060>
Expires: 180
Allow-Events: telephone-event
Session-ID: ef310bfacdac5fea9c8929e1a2618dfa;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 317

v=0
o=CiscoSystemsSIP-GW-UserAgent 192 472 IN IP4 20.20.20.20
s=SIP Call
c=IN IP4 20.20.20.20
t=0 0
m=audio 17194 RTP/AVP 0 8 18 101
c=IN IP4 20.20.20.20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

016223: Jul 24 14:51:29.772: //14092/B597AFD2B5FA/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bK2CDAE5
Call-ID: BCA9FD88-8E6E11E8-B5FFC93F-47B3019@20.20.20.20
From: "Lounge"<sip:+254709655000@20.20.20.20>;tag=A5AAD12C-19F2
To: <sip:+254725101240@10.10.10.10>
CSeq: 101 INVITE
Content-Length: 0


016224: Jul 24 14:51:29.796: //14092/B597AFD2B5FA/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bK2CDAE5
Record-Route: <sip:10.10.10.10:5060;transport=udp;lr>
Call-ID: BCA9FD88-8E6E11E8-B5FFC93F-47B3019@20.20.20.20
From: "Lounge"<sip:+254709655000@20.20.20.20>;tag=A5AAD12C-19F2
To: <sip:+254725101240@10.10.10.10>;tag=sbc0402vwcmjnvu-CC-1005-OFC-100
CSeq: 101 INVITE
Content-Length: 0


016225: Jul 24 14:51:29.796: //14092/B597AFD2B5FA/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:+254725101240@10.10.10.10:5060 SIP/2.0
Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bK2CDAE5
From: "Lounge" <sip:+254709655000@20.20.20.20>;tag=A5AAD12C-19F2
To: <sip:+254725101240@10.10.10.10>;tag=sbc0402vwcmjnvu-CC-1005-OFC-100
Date: Tue, 24 Jul 2018 14:51:29 EAT
Call-ID: BCA9FD88-8E6E11E8-B5FFC93F-47B3019@20.20.20.20
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: 00000000000000000000000000000000;remote=ef310bfacdac5fea9c8929e1a2618dfa
Content-Length: 0

==========================================================================

                                                    SIP CONFIG

===========================================================================

dial-peer voice 17 voip
corlist outgoing SAFSIPLIST
description "SAFSIP OUTGOING"
translation-profile outgoing OUTGOINGSAFSIP
destination-pattern 7T
session protocol sipv2
session target ipv4:10.10.10.10
session transport udp
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad

 

 

1 ACCEPTED SOLUTION

Accepted Solutions
Highlighted

Hi guys 

 

 

so I found out what the problem was the parameter "Min-SE: 86400" was too high and thus the provider was dropping the call. Different providers work with different figures , in my case I set it to 1800 and the calls started working.

View solution in original post

14 REPLIES 14
Highlighted
Beginner

You are getting SIP/2.0 500 Server Internal Error from remote device. What is the device with IP 10.10.10.10? Do you have access to that?

 

Thank you,

Mikolaj

 

**** PLEASE RATE IF USEFUL ****

Highlighted
Cisco Employee

The INVITE looks fine. What's located at the remote end? Do you have access to the remote device?

Highlighted

Hi Mikolaj and  Sreekanth

 

I don't have access to that device as it is the Telcos Sip server, so only way to proceed I guess is to get in touch with them?

Highlighted

That's right. You could try removing the Session ID header using sip-profiles and apply it to the dial-peer. Perhaps that header isn't understood by the Telco server.
Do you have registration with the provider? Is the trunk registered?
Highlighted

the trunk is registered , if I use a sip client on my laptop I can make outgoing calls, problem is when I connect it to the router (2911). Incoming calls are working fine , however the outgoing bring about the error posted above and as usual Telcos ,dont want to corporate

Highlighted

Where are these calls originating from ? You have IP Phones registered on the CME. If your ITSP does not want to assist you with troubleshooting, time to change it and move to a better one. For now, escalate to their Tier 3 and have them review the logs.
Also, the 10.10.10.10 is a private IP. Is there a ITSP box at the CE ?

Nipun Singh Raghav
"We cannot solve our problems with the same thinking we used when we created them"
Highlighted

the calls are originating from ip phones registered with the CME , the addresses 10.10.10.10 and 20.20.20.20 are replacing the real ip addresses being used , just thought its not best practice to post real envirenment IPs here.

 

Just for clarification , in the above SIP message any headers after the word "received " means this is coming from the Telcos end right?  for instance the below excerpt of the initial error posted. If so their sip server should be having logs of the invites I'm sending them right ?

 

Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 20.20.20.20:5060;branch=z9hG4bK2CDAE5
Record-Route: <sip:10.10.10.10:5060;transport=udp;lr>
Call-ID: BCA9FD88-8E6E11E8-B5FFC93F-47B3019@20.20.20.20
From: "Lounge"<sip:+254709655000@20.20.20.20>;tag=A5AAD12C-19F2
To: <sip:+254725101240@10.10.10.10>;tag=sbc0402vwcmjnvu-CC-1005-OFC-100
CSeq: 101 INVITE
Content-Length: 0

 

Highlighted

Yes the SIP server will have the logs of these calls. You are receiving this 500 from them
Highlighted

Not really. If you have a SIP IP Phone all messages being received from it will be a RECEIVED as well. This one looks to be from the ITSP but the best way is to mark the Call-ID to differentiate both the call legs and see which leg is sending what.

I also see a RR which would be the proxy that wants to stay in the path for the signalling. Which would be whatever IP you replaced with 10.10.10.10

Nipun Singh Raghav
"We cannot solve our problems with the same thinking we used when we created them"
Highlighted

I have seen that Internal Server Error actually be generated by the router, even though it looks like it is being received from the provider.  It has been generated by many random config issues or mismatches, but the one I saw recently was that the codec settings on the inbound and outbound call legs did not line up, and so the router refused to continue.  You really should have a specific inbound and outbound dial peer set up for any sip trunk, so that you match the parameters you think you should be matching, voip-wise - so your outbound dial peer towards the provider would be stating 'destination-pattern 7T' for instance, and your inbound call leg from the provider would be 'incoming called-number .T'  (you don't need destination pattern on inbound leg).  Then make sure both of those peers match as far as codec, session transport, dtmf, vad, etc.

I have seen other things cause the 500, but the dial peer matching is a big one.

I am usually looking at CUCM sip trunk to cube, and then cube to provider, and in that case, the same applies in the direction back towards CUCM, and if the cube tries to set up a call with one set of codecs on the list coming from CUCM, but missing one of those towards the provider, (or vice versa) that has also caused the error - the codec lists had to match in the cube in both directions.  This may be an IOS specific 'feature', but I have definitely experienced it in my own lab.

 

I also just found an old set of notes, where the CUBE was generating the internal server error until we removed the 'pass-through content sdp' from the sip parameters, since some impossible (to the router) scenario was attempting to be negotiated.

 

Mary Beth

Highlighted

We will get the answer if you compare the INVITE from your PC to the one from the router. The router is probably adding something else in the message that the PC isn't. The Session-ID header is one example
Highlighted

these are the first three sip messages (from wireshark) I get when I'm using a sip client to initiate the call and it works, any help in troubleshooting this will be highly appreciated its the third day I'm working on this.

 

Message Header
Via: SIP/2.0/UDP 20.20.20.20:62194;rport;branch=z9hG4bKPj3e4b31212cb340f1af360d69c89a80ba
Max-Forwards: 70
From: <sip:+254709655000@10.10.10.10>;tag=ea41aa6311564e12a4b1e0b3dfc9ad0c
To: <sip:0725101240@10.10.10.10>
Contact: <sip:+254709655000@20.20.20.20:62194;ob>
Call-ID: 4a154fb104d44f1e8c8ec42b9015648b
CSeq: 7330 INVITE
Sequence Number: 7330
Method: INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.18.5
Content-Type: application/sdp
Content-Length: 404

Message Header
Via: SIP/2.0/UDP 20.20.20.20:62194;branch=z9hG4bKPj3e4b31212cb340f1af360d69c89a80ba;rport=62194
Call-ID: 4a154fb104d44f1e8c8ec42b9015648b
From: <sip:+254709655000@10.10.10.10>;tag=ea41aa6311564e12a4b1e0b3dfc9ad0c
To: <sip:0725101240@10.10.10.10>
CSeq: 7330 INVITE
Sequence Number: 7330
Method: INVITE
Content-Length: 0


Message Header
Via: SIP/2.0/UDP 20.20.20.20:62194;branch=z9hG4bKPj3e4b31212cb340f1af360d69c89a80ba;rport=62194
Record-Route: <sip:10.10.10.10:5060;transport=udp;lr>
Call-ID: 4a154fb104d44f1e8c8ec42b9015648b
From: <sip:+254709655000@10.10.10.10>;tag=ea41aa6311564e12a4b1e0b3dfc9ad0c
To: <sip:0725101240@10.10.10.10>;tag=sbc0403zc9r9a8b-CC-1021-OFC-100
CSeq: 7330 INVITE
Sequence Number: 7330
Method: INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
Contact: <sip:10.10.10.10:5060>
Require: 100rel
RSeq: 1
Content-Length: 190
Content-Type: application/sdp

Highlighted
Engager

Hi,

 

Do you have the configuration output available to share (no passwords Please)

 

"SIP Calls Receives 500 Internal Server Error "Routing Failed" Event. A SIP call fails and a 500 Internal Server Error "Routing Failed" event is received. A SIP Trunk is not configured to receive or send calls to the IP Address from the source of the SIP INVITE event"

 

This sip ITSP use a proxy server to register the trunk with login and password?

 

Best regards

Daniel Sobrinho
Highlighted

Hi guys 

 

 

so I found out what the problem was the parameter "Min-SE: 86400" was too high and thus the provider was dropping the call. Different providers work with different figures , in my case I set it to 1800 and the calls started working.

View solution in original post