Some SIP Trunks work others don't

[libertyinstaller]

Spectrum gave me a block of 20 (New) DID numbers that all worked prior to my cut. After the cut only my 4 BCP (analogue) trunks that were cut over work.

Spectrum states that I have a problem!

If, Spectrum gets the new numbers to work the 4 transferred trunks go dead. if the 4 transfered numbers work the new numbers go dead.

Remember, Spectrum states that this is my problem.

When the new or transferred trunks I'm not at the office or in Manager changing the configuration.

I'm running an I.P. Office 500 V2 R 9.1 with 2 digital 8's, 1-ATA, 1-VCM-32 and a UC module.
I have 2-5 license SIP packages
Essential License
VM Pro 4 Port

I have formatted and recreated the SD Card, tried on both Wan and Lan ports,
Rebuilt the ICP several times as well as the SIP line.

Any Ideas?

 

[intrigrant]

Spectrum gave me a block of 20 (New) DID numbers that all worked prior to my cut. After the cut only my 4 BCP (analogue) trunks that were cut over work.

Nice, i have no idea what you mean.
Cut what? The roses? Your hair?

I have 2-5 license SIP packages

2 or 5 OR 2 and 5 gives 7, SIP Phones or SIP trunks?

Give something to go on please.

 

[libertyinstaller]

Cut over (Porting) Business class lines to SIP.
I have 2, 5 trunk Channel license packages installed. I'm receiving 8 channels from Spectrum.

 

[intrigrant]

Right so basically the "cut"have nothing to do with your problem, you just cannot get the SIP trunk working.

Did you ever setup a SIP trunk before?
Did you setup the SIP Trunk properly?
Did you made the necessry changes to the customers firewall?
Do you have a IP route towards the providers SIP registrar?
Did you enable SIP Trunking in System>LANx>VoIP?

 

[critchey]

I am not 100% sure I understand but I think I know what you mean...

You were given 20 DIDs that were new to you and you set up a SIP line using these 20 DIDs and they all worked.
You then ported over 4 analog numbers to SIP, and those 4 lines works but the original 20 DIDs that previously worked stopped working... is this correct?

If that is the case I guess my question is did you change anything in the SIP line or did you simply add the new DIDs to your incoming call route (or SIP tab or whatever you use)? Are they failing inbound or outbound or both ways? What does monitor trace say when you use a working DID and what does it say when you use a non working DID? If you you try a non working DID does it even hit the phone system (monitor will tell you)?

You provided a lot of information but not really the right information... specifically nothing you did to troubleshoot other then reformatting and recreating the SD card and re-doing the lines and ICR. You should always run monitor when you have these type of issues and you should know if the call hit your phone system and what messages you are getting (or not getting). If you came here and said "I am getting this message in monitor what does it mean" then we can help you. Till then you need to do diagnostics.

The truth is just an excuse for lack of imagination.

 

[libertyinstaller]

Q1: Yes
Q2: Simply added new numbers to incoming call route
Q3: 16 Numbers don't work inbound Error 404 unallocated number, all 20 work out bound.
Q4: answered above
Q5: In Monitor it looks like a good call is connecting, but then you see a few different (ie quit, end, bye) reasons and the call terminates ending with unallocated (unrecognized) number.

 

[janni78]

Unallocated Number means that it can't route the number so it could be your ICR, SIP URIs etc.

It's hard to tell without trace and config and you shouldn't post the config.

"Trying is the first step to failure..." - Homer

 

[libertyinstaller]

intrigrant

Q1: Yes, about 50 times in the past 2 years.
Q2: Yes, 50 times!
Q3: Yes as this time it's my own firewall.
Q4: Yes, static
Q5: Yes, residing on Lan 1 of the IP Office.

As I said earlier either the 20 original numbers work or the ported 4 numbers work. But not at the same time.

 

[libertyinstaller]

CANCEL sip.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.60:5060;branch=z9hG4bK-d8754z-02ae86b009bd0361-1---d8754z-;rport
Max-Forwards: 70
To: <@192.168.1.60>
From: Call-ID: ZjcwMzM2NjE0MDQ0ZWQyNWZlMThkNjAxZTQ4ZTRlZDk.
CSeq: 1 CANCEL
User-Agent: ESBC9378-4B-2.0.13.0-Build8
Content-Length: 0

17:22:42 2941326mS SIP Tx: UDP 192.168.1.50:5060 -> 192.168.1.60:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.60:5060;branch=z9hG4bK-d8754z-02ae86b009bd0361-1---d8754z-;rport
From:
Call-ID: ZjcwMzM2NjE0MDQ0ZWQyNWZlMThkNjAxZTQ4ZTRlZDk.
CSeq: 1 CANCEL
Supported: timer
Server: IP Office 9.1.11.0 build 202
To: <sip192.168.1.60>;tag=bfb393b768d2ffea
Content-Length: 0

17:22:42 2941327mS SIP Tx: UDP 192.168.1.50:5060 -> 192.168.1.60:5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.60:5060;branch=z9hG4bK-d8754z-02ae86b009bd0361-1---d8754z-;rport
From:
Call-ID: ZjcwMzM2NjE0MDQ0ZWQyNWZlMThkNjAxZTQ4ZTRlZDk.
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.11.0 build 202
To: <@192.168.1.60>;tag=bfb393b768d2ffea
Content-Length: 0

 

[intrigrant]

I did not wanted to offence you in any way but you just did not ask a clear question and you did not provide any usefull information.

What you *should* have asked to start with is something like *i have a working SIP trunk and have added 20 DID numbers of which 16 don't work inbound*
Then there would not be any miscommunication to start with.

So, having said that it is truly odd if four of them do work.
The piece of the trace is TBH as useful as your initial question, there is nothing to go on.

 

[libertyinstaller]

NONE TAKEN!

Thank you for rephrasing the question, I'm never to old to stop learning.

Every time I ordered a pizza, locally. I would ask for less than normal cheese and I would always get it with about the normal amount of cheese. So one day I asked the owner what I need to do? His reply was; we put 6 ounces of cheese on that size. So the next time you order, ask for 4 ounces of cheese vs 6 ounces and ever since it comes out perfect, now!

Now back to SIP as you can see one call made it to the system and gave me a half ring but terminated before I could answer.

 

[intrigrant]

Make a complete trace of a failing inbound call and enable the defaurl filter plus SIP EXCEPT sip TX and RX as it will put a lot of garbage in the trace.
Add the trace as a text file and not pasting it in the response, it makes it incredily hard to read.

 

[IPGuru]

simple question
can you see the failing calls being presented in the sip trace?

Yes - it is probably your fault & you need to recheck your configuration
NO - it is probably the sip providers fault.

this assumes that the SIP provider has ported the numbers onto the existing sip trunk & not configured a new one with separate credentials



Do things on the cheap & it will cost you dear