Calls drop after 10 seconds, after SIP AUTH added on Windstream

[ScottK]

Having issues after setting up Authentication on Windstream.
Was required to set up SIP AUTH id/password after some hacking on a DID (their issue). Once the SIP AUTH was in place, and provider activated it, outbound calls will drop after 10 seconds. Inbound and internal calls are fine.

In short, The initial exchange is fine. SDP sent, ACK rec'd. Call dials and picks up. Then a PRACK is sent out, comes back OK.3CX sends PRACK and receives a 401 Unauthorized. 3CX doesn't respond, and call is hung up

I'm looking through troubleshooting but it's as clear as mud as I've been dropped into this.

3CX SW is up-to-date (auto-update)
Provider is Windstream, we are talking to a Edgewater gateway

Sample - Dialed phone#, get ~10 seconds of talk time
Call Log Entry:
Date/Time From To Talking
09/30/2021 2:32:55 PM Scott Kolesar (600) 8779997705 Not Answered

Event Entry:
Call or Registration to 18779997705@(Ln.10000@Windstream) has failed. 10.0.0.9 replied: 500 Internal Server Error; from IP:10.0.0.9:5060

Outbound Parameters. I tried using OutHostPort for Request Line URI : Host Part as suggested elsewhere with no change.

SIP Field: Variable
Request Line URI : User Part "CalledNum" number that has been dialed (default: To->user)
Request Line URI : Host Part "GWHostPort" gateway/provider host/port
Contact : User Part "AuthID" authentication
Contact : Host Part "ContactUri" usually, content of Contact field
To : Display Name "CalledName" name that has been dialed (default: To->display name)
To : User Part "CalledNum" number that has been dialed (default: To->user)
To : Host Part "GWHostPort" gateway/provider host/port
From : Display Name "OutboundCallerId" Outbound caller Id taken from Extension settings in management console
From : User Part "CallerNum" caller's number (default: From->user)
From : Host Part "GWHostPort" gateway/provider host/port
User Agent : Text String Leave default value
Remote Party ID - Called Party : Display Name Leave default value
Remote Party ID - Called Party : User Part Leave default value
Remote Party ID - Called Party : Host Part Leave default value
Remote Party ID - Calling Party : Display Name Leave default value
Remote Party ID - Calling Party : User Part Leave default value
Remote Party ID - Calling Party : Host Part Leave default value
P-Asserted Identity : Display Name Leave default value
P-Asserted Identity : User Part Leave default value
P-Asserted Identity : Host Part Leave default value
P-Preferred Identity : Display Name Leave default value
P-Preferred Identity : User Part Leave default value
P-Preferred Identity : Host Part Leave default value
P-Called-Party-ID : Display Name Leave default value
P-Called-Party-ID : User Part Leave default value
P-Called-Party-ID : Host Part Leave default value

Provider Trace

 

[ScottK]

UPDATE: Sorry. Too late to edit or delete. I spilled out a harried post looking for help. Just tell me to pay the $75. Or, tell me which chapter to RTFM, which should have been my question.

 

[Evolute IT]

@cobaltit and @leejor are good with 3rd-party trunks.

But this sounds like a AuthTrunk issue, which I am not sure if 3CX even supports both methods.

 

[leejor]

Is this the same Windstream that offers a virtual PBX, competing with 3CX?

They may very well be, now, doing something that is no longer supported by 3CX. Unfortunately, that is always a possibility when using an unsupported provider. Since 3CX is sending the SIP: PRACK, and there is no response, perhaps Windstream can tell you why that is happening....

"oh... 3CX should not be sending it"
"we don't receive it"
"we don't respond to that"

 

[ScottK]

perhaps Windstream can tell you why that is happening....

According to Windstream, They send a second SIP: 401 Unauthorized after 3CX sends the PRACK, then waits for another response. [Mind you the call is in picked up and talking] It then drops the call after timing out, waiting for 3CX to send the new auth, which never happens (hence the series of SIP:183 messages leading to BYE)

>>Is there some sort of setting related to a re-auth during a call?

 

[leejor]

Could something like this, be what they have implemented, given they were having security issues?

https://www.researchgate.net/publication/274393691_Two_Step_Mutual_Authentication_Scheme_for_SIP

Perhaps you can find out the name of the change they made.

 

[ScottK]

I can pass that link along. It looks theoretical though; has that been formally implemented?

During our testing, I entered ID and password into Authentication. Once it was set up on the Windstream end, the only Type of Authentication option that allowed a call out was Do Not Require - IP Based. He did see the initial authentication to allow the call out. Using the Register/Account based option did not even open a line.

find out the name of the change they made

Direct Response from Windstream:
"
Scott,
We follow the RFC for SIP

https://www.ietf.org/rfc/rfc3261.txt
See Sections 22.1-22.4
"

 

[YiannisH_3CX]

That call flow looks very strange to me as 3CX does not support PRACK and I have never seen a PBX system sending PRACK. Can you send me a p.m. and share that capture with me so I can have a look? I would also prefer to see a capture from the 3CX side as well. Something really strange is going on here.