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Issues with SIP Trunk on R11.0.4.1.0, "Waiting for Line" and then Connects 2

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dsm600rr

dsm600rr

IS-IT--Management
Nov 17, 2015
1,444
US
All,

Having intermittent issues with my NexVortex SIP Trunk.

From what I can tell, this is only on Outbound calls. All testing is to the same outbound TN

Most of the time, when I place an outbound call, it connects right away as it should.

Other times, I will get "Waiting for Line" and the "Beep-Beep-Beep".

If I get the "Waiting for Line" and wait for: "Beep-Beep-Beep"-"Beep-Beep-Beep"...."Beep-Beep-Beep" and then it connects.

Thoughts?

I did not see anything much in SSA:

12651485mS SIP Rx: UDP 104.219.163.73:5060 -> 172.30.20.1:5060
SIP/2.0 407 Proxy Authentication Required


ACSS
 
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We have a customer in the UK with the same problem at version 9 using SIP trunks. I was able to capture the error as in the attached file via System Status. I made over 200 calls and it happened only around 4 times. We tried changing the SIP setting in the user section where it shows the 3 extension number to have the main CLI instead as the error log showed extension 249 was the outgoing CLI.

My file shows one error and two successful calls to a mobile number that I was dialling. Needless to say, that the problem is still happening and we have no idea if the issue is Avaya related or with the internet supplier / SIP trunk provider.

Next steps will be to change out the router and upgrade the IPO.

When I had the problem, I got the 3 bleeps, My mobile was ringing but I couldn't answer the call.

Firebird Scrambler

Nortel & Avaya Meridian 1 / Succession & BCM / Norstar Programmer

Website = linkedin
 
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Derfloh: that is what I meant. Monitor trace didn’t show much. Are there specific trace options I should select other that the default ones?

ACSS
 
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The default should show all needed information.

If you just want SIP messages, clear all filters and just enable SIP call rx/tx

IP Office remote service Fixed price SIP trunk configuration: CLI based cale blocking: SCN fallback over PSTN:
 
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I will do some more testing Monday and post the traces.

In the mean time, here is what NexVortex sent me from their end during testing:

Great speaking with you again! As we discussed, seems their is a delay with your outbound calls getting to us. We tested today with the below data:

12:40:38 pm ET...you pressed the last digit on your outbound call to 586488XXXX
12:40:47 pm ET...you hear the 'beep' on your side
12:40:52 pm ET or 17:40:52 GMT...we receive your initial invite, first packet received for the call
12:40:54 pm ET... I was able to grab the 183 Session Progress (ringing) that we sent back to you two seconds later...nice!

Call example:
27 Dec 2019 17:40:52 1586961XXXX 1586488XXXX

Your initial invite with time stamp:
2019-12-27 17:40:52 +0000 : 50.245.XXX.XXX:5060 -> 104.219.XXX.XXX:5060 INVITE sip:1586488XXXX@px1.nexvortex.com SIP/2.0
Call ID:
Call-ID: ce65c50f356e9a8b96221db54fe940ca
183 Session progress sent back to you two seconds later:
2019-12-27 17:40:54 +0000 : 104.219.XXX.XXX:5060 -> 50.245.XXX.XXX:5060 SIP/2.0 183 Session Progress

ACSS
 
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So I was able to do some testing this morning, and going through the monitor trace, only one thing really stuck out from the log (I can post the entire log if that may help)

I took about 20 successful test calls to get the issue to show.

Basically I will make the call, get "Waiting for Line", "Beep-Beep-Beep"-"Beep-Beep-Beep"...."Beep-Beep-Beep" and then it connects. While it is "Waiting for line", my phone is not ringing, Once it goes through the "Waiting for Line", "Beep-Beep-Beep"-"Beep-Beep-Beep"...."Beep-Beep-Beep" the call connects and my cell phone begins to ring.

Below is part of the trace that may be the issue:

261721133mS SIP Rx: UDP 104.219.XXX.XXX:5060 -> 172.30.20.1:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 50.245.XXX.XXX:5060;received=50.245.XXX.XXX;rport=5060;branch=z9hG4bKa6e2adaaf8adb7c19536b2a87c24f294
From: "586961XXXX" <sip:586961XXXX@px1.nexvortex.com>;tag=f051663cdedb41ee
To: <sip:1586488XXXX@px1.nexvortex.com>;tag=09c5247e8a9116fcdc76c940611a16eb.b0a0
Call-ID: 00bb219e7357e8cf769faba0de4130bb
CSeq: 1132117291 INVITE
Proxy-Authenticate: Digest realm="nexvortex.com", nonce="XgoXvl4KFpJ6W8iI0kFBq60chXnyeSR/"
Content-Length: 0

Thoughts?




ACSS
 
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That part is not unusual. IPO should send ACK and then another INVITE with an additional authentication header. We need a full trace and not only part of a trace.

IP Office remote service Fixed price SIP trunk configuration: CLI based cale blocking: SCN fallback over PSTN:
 
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SIP Rx: UDP 104.219.163.73:5060 -> 172.30.20.1:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 50.245.111.77:5060;rport=5060;branch=z9hG4bK2cc326eda2bc238f3a213e0c86494376;received=50.245.111.77
From: "5869616380" <sip:5869616380@px1.nexvortex.com>;tag=f051663cdedb41ee
To: <sip:15864885440@px1.nexvortex.com>
Call-ID: 00bb219e7357e8cf769faba0de4130bb
CSeq: 1132117292 INVITE
Content-Length: 0

This seems like the provider is sending you a "please continue to hold" message as the call is in transit. You got the above response 3 times during the course of your trace.

The truth is just an excuse for lack of imagination.
 
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If it is getting that message then disable the PRACK option on the SIP trunk. I had this happen and could not call certain numbers and it was because of the PRACK from the provider. It is on the VOIP tab of the SIP trunk.
Mike
 
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Side note, on successful test calls, "SIP/2.0 100 trying -- your call is important to us" still shows up in the trace.

ACSS
 
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teletechman: Should I disable "Re-Invite" and test?

ACSS
 
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I am not sure if this is a coincidence or not, however I was going through the document from NexVortex (although its done on R9.1 and I am on R11) and I noticed a few things that were not done or removed during the months I have been testing with NexVortex.

Either way I enabled "Caller ID from From Header" and I was able to make 100 successful outbound calls without getting any "Waiting for Line" issues.

I also noticed that I either never added or removed at some point:

Calls Route via Registrar: Disable (Mine is checked)
Separate Registrar: reg.nexvortex.com

I have not done the above yet, thoughts?

ACSS
 
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Maybe not related, but I recently started having similar issues with my outbound DIDWW Trunks. They just started hanging forever and eventually timing out.
I found out by accident how to fix it for me. In the user, go to sip tab, and make sure that SIP Name and Contact fields have actual 7 10 or 11 digit phone numbers in there and not just the extension.

Fixed the issue for me.


Travis
 
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