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Windstream - Nuvox Makes a Change

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Windsream Formally Nuvox, has made a change on theor Sonus platform and three clients using windstream now fails, Windstream support in working late into the night to determine what has changed on their end.

 

After the invite, PBX provide 100 trying, then a progress 183, followed by a 400 Bad Request.

 

Absolutely nothing had changed on all three clients, each with dedicated SIP services and all three failed at same time... I hope to here something yet tonight or in the AM, hoping to get a heads up if possible..

 

 

The call looks like this skipping the details. I think an update on the Sonus warrants a trunk settings change.... Any advise helpful.

|Time | 74.223.147.173 |

 

| | | 173.221.11.118 |

 

|0.000 | INVITE SDP (g711U g711A g723 g729 g721 telepho...eventRTPType) |SIP From: "GREENVILLE SC" <sip:8649184278@74.223.147.173:5060 To:<sip:6566736005@171.222.11.118:5060

 

| |(5060) ------------------> (5060) |

 

|0.013 | 100 Trying| |SIP Status

 

| |(5060) <------------------ (5060) |

 

|0.083 | 183 Session Progress SDP (g711U telephone-even...PType-101) |SIP Status

 

| |(5060) <------------------ (5060) |

 

|0.143 | 400 Bad Request |SIP Status

 

| |(5060) <------------------ (5060) |

 

|0.200 | ACK | |SIP Request

 

| |(5060) ------------------> (5060) |

 

 

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PBX sends back 400 Bad Request response only on a new INVITE that does not have the contact header. But based on the flow you mentioned (that PBX sent back 100 & then 183), I am surprised about the 400 response. Is there any looping going on inside the PBX or are we getting another INVITE here? Can you check the PBX SIP log or wireshark trace please?

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I have no idea how to add attachments to this board. I've sent the trunk pcap file to KM and MD requested emergency help..

 

 

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Guest kevin

After working with windstream it appears they upgraded their Sonus switch which caused the issue. The swung us over to the non upgraded switch and it is working again. Looks like it is some configuration with the new load since they send the 400 message back pretty quickly on outbound calls with a syntax error and nothing changed on our side.

 

SIP/2.0 400 Bad Request

Via: SIP/2.0/UDP 75.1.2.3:5060;branch=z9hG4bK-5ca194d545ee80d31041fd89653a2d00;rport=5060

From: "caller" <sip:125@localhost>;tag=34164

To: <sip:12125551212@75.1.2.3;user=phone>

Call-ID: 0f244044@pbx

CSeq: 12262 INVITE

Error-Info: <sip:12125551212@75.1.2.3>;cause="[line 028] SIP syntax error"

Content-Length: 0

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Error-Info: <sip:12125551212@75.1.2.3>;cause="[line 028] SIP syntax error"

 

It would be intesting to see the INVITE, especially line 28. Then we might know what the problem is, and if it can be fixed by configuration.

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Guest kevin

Here it is with modified IP addresses

 

INVITE sip:3172222222@1.2.3.4;user=phone SIP/2.0

Via: SIP/2.0/UDP 4.3.2.1:5060;branch=z9hG4bK-5ca194d545ee80d31041fd89653a2d00;rport

From: "someones name" <sip:125@localhost>;tag=34164

To: <sip:3172222222@1.2.3.4;user=phone>

Call-ID: 0f244044@pbx

CSeq: 12262 INVITE

Max-Forwards: 70

Contact: <sip:125@4.3.2.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/4.2.1.4025

Remote-Party-ID: "someones name" <sip:125@10.1.1.251>;party=calling;screen=yes

Content-Type: application/sdp

Content-Length: 257

 

v=0

o=- 15773 15773 IN IP4 173.221.11.118

s=-

c=IN IP4 173.221.11.118

t=0 0

m=audio 49710 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=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

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