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NexVortex SIP Trunk on IPO 4

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dsm600rr

IS-IT--Management
Nov 17, 2015
1,444
US
Hello all,

I am going through the NexVortex SIP Trunk Document and have a question as the Document is from R9.1 and I am on R11 and as you know some of the SIP Settings have changed.

On the SIP URI Tab, they have the "Local URI", "Contact" and "Display Name" as "*" and the PAI and "Use Internal Data"

On R11 you cannot input a "*"

Under the "Call Details" Tab I do see the "Local URI, Display / Contact" and "Contact, Display / Contact" tabs which are set to "Auto"

Also "SIP Line Appearance" field below the "SIP URI's", Suggestions here?

URI_tmez7y.png


SIP_Line_Appearences_fx6rax.png





ACSS
 
I set my R11 SIP trunks up as

Local URI - Auto Auto Caller Caller Called
Contact - Auto Auto Caller Caller Called
P Asserted ID - Auto Auto Caller Caller Called
Diversion Header - Auto Auto Caller Caller None

Seems to work for me

As for SIP Line Appearances - never touched it and have no plans to.

| ACSS SME |
 
Auto is the same as * was.

Stuck in a never ending cycle of file copying.
 
Thanks Gents

I am assuming this is firewall related, (we have some sort of Ubiquity Firewall) however wanted to run it buy you guys before I get them involved and they swear up and down the street the issue is on my end.

Outbound calls on the SIP Trunk work fine, however they send the CID of my extension number rather than the TN. This was fixed with NSi on the ARS Table. Was curious if this was due to me not having something configured correctly.

Inbound calls have no TX or RX Audio on either end.

ACSS
 
With Auto as the setting any outgoing calls will automatically use the DDI assigned to that extension on the ICR as the outgoing number - if no DDI assigned then yes you need an NS (i not needed) to send a specific number otherwise your user SIP tab details will be sent.

Inbound calls not having speech will 99% of the time be the firewall.

| ACSS SME |
 
Pepp77: Thank you. Appreciate the Info. As far as the users "SIP" Tab, what exactly are the "SIP Name", "SIP Display Name" and "Contact" doing? The "Help" Tab leaves a bit to be desired here.

Also on random outbound calls I will get "Call Rejected". Thoughts? I will run a Trace now.

ACSS
 
SIP Name and Display Name I believe are used for internal Caller ID - between local SIP trunks.
Contact is the one that matches the contact entry on the SIP URI tab.

For the outbound calls you would need to run monitor and capture a trace of the failed call (and the responses from the SIP provider) and compare it to a call that worked.

Usually with SIP its because of a low dial delay time and people dont dial quickly enough so the whole phone number doesnt get sent to the provider and so the call is rejected.

| ACSS SME |
 
Hi Pepp77: Again, thank you for your input. This is on a test PBX and I am hitting redial as well as dialing the number and I will sometimes get the "Call Rejected Error"

The "Call Rejected" From what I have noticed is always on a follow up call immediately after a successful one.

I am getting a "403 Forbidden" on my trace:

2063296mS SIP Rx: UDP 104.219.xxx.xxx:5060 -> 192.168.xxx.xxx:5060
[highlight #EDD400] SIP/2.0 403 Forbidden[/highlight]
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;received=50.245.xxx.xxx;rport=5060;branch=z9hG4bKe084a7cc54e840ae1b40c14ba12a77c0
From: "xxx-xxx-xxxx" <sip:xxxxxxxxxx@nexvortex.com>;tag=bf164d77164747ed
To: <sip:xxxxxxxxxxx@nexvortex.com>;tag=134e05171e7c22f4d450ef7f18a602bc.0ffe
Call-ID: f09b9e7efb754b8c7abcf9004071f153
CSeq: 1318401284 INVITE
Content-Length: 0


ACSS
 
Just for reference, unchecking the 'Cache Auth Credentials' button in the 'SIP Advanced' tab corrected the "Call Rejected Error"

ACSS
 
what exactly are the "SIP Name", "SIP Display Name" and "Contact" doing?"

Those values are only used if the SIP URI routing the call in or out is set to "Use Internal Data". If your SIP URIs are using the default Auto setting then they can be ignored (In fact I would love Avaya to hide them unless there is a User Internal Data URI in the config. They're too visible and I had to deal with too many cases where people go straight to those settings and waste a lot of time rather than going to the SIP line settings.)

Stuck in a never ending cycle of file copying.
 
sizbut: Appreciate the information.

Few other fun things. I was working with NexVortex yesterday and on their end they were showing the Internal (private) IP Address of my IPO being sent to them.

So they send me a IPO/NexVortex Config Document and on the WAN side of the IPO they mention inputting a Public IP Address. I have never had to do this with any other SIP trunks I have worked with (Coredial, WhiteLabel, etc.) Thoughts?

Our invite to you, request line URI:
2019-11-25 21:26:49 +0000 : nVproxy:5060 -> [//104.219.xxx.xxx]104.219.xxx.xxx:5060 INVITE sip:1586xxxxxxx@50.245.xxx.xxx SIP/2.0
Your 200OK back to us:
2019-11-25 21:26:51 +0000 : 50.245.xxx.xxx:5060 -> 104.219.xxx.xxx:5060SIP/2.0 200 OK
Your contact header and connection part of SDP portion from your 200OK:
Contact: <sip:1586xxxxxxx@192.168.xxx.xxx:5060;transport=udp>
c=IN IP4 [//192.168.xxx.xxx]192.168.xxx.xxx

192.168.xxx.xxx is the IPO WAN IP Currently.

Public_nr8cjy.png



ACSS
 
You use Network Topology to rewrite SIP headers to the public IP.

Tell them to rework that document, the IP Office should not have a public IP address on it's LAN interfaces.

"Trying is the first step to failure..." - Homer
 
janni78: Thank you for the info, I did not think that was correct.

Under "Network Topoligy" I do see a spot to put a "Public IP Address" however I am unsure on how to exactly "use Network Topology to rewrite SIP headers to the public IP"

ACSS
 
Usually you put Firewall/NAT Type to Static Port Block and then type in the Public IP Address.
Then you select the LAN interface under SIP Line -> Transport -> Use Network Topology Info.

"Trying is the first step to failure..." - Homer
 
janni78: Thank you a ton - all is working now!

ACSS
 
Anytime a SIP provider says to put an interface on a public IP address I counter with this screenshot

Public_IP_Platform_Security_Guidelines_IPO_eestjj.jpg


The truth is just an excuse for lack of imagination.
 
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