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Call to sip amplifier go out through sip trunk

Unanswered Question
Jun 1st, 2012
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I am re-wording this question. I have a cyberdata sip endpoint amp. It is registered, connected, has an extension. when i dial the extension it gives a fast busy. When i check the logs it shows that the call is going out through the sip trunk to our ITSP (nexvortex), where of course it fails. How do I create a dial plan to keep it inside?

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David Trad Sun, 06/03/2012 - 18:26
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    2013 Small Business

Hi Barry,


I think you have made the mistake that many of us made the first time we looked at a SIP amplifier


Firstly you do not want to use the SIP register for this, actually you want to use a SIP-DN which falls under your global DN registration, this then creates a SIP extension much like the SCCP extension (Which I believe Cisco will move down this path shortly).


When you create a SIP-DN it acts and behaves like a normal extension, and it also ensures that it will correctly hang-up once the user closes the circuit as it wont need to wait for a response back from the unit.


Here is the sticking point CCA does not support creating this as far as I am aware (In version 3.2), maybe the beta version 3.2.1 might?? So you will need to do this via CLI and this may get you into some trouble with SBSC if you are not Express certified, so maybe you can log a configuration assistance with SBSC on this and see if they will assist you with configuring this up.


What I can tell you is that this is the best way for it to work, creating a SIP registration like you would with an ITSP requires further configuration such as Dial Plans and Translation rules/profiles etc...etc.. Not the most ideal way to do it



Cheers,



David.

Barry Hunsinger Mon, 06/04/2012 - 06:48
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Thanks David. I will try to sbcs to cooperate. They don't like to help support 3rd party products. They may be willing to help create a sip-dn, and then I can take it from there.

Barry Hunsinger Mon, 06/04/2012 - 08:47
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I just called sbcs. they claim I don't have cli support. I didn't know that that was an option. Does anyone have the cli commands to create sip-dn?

David Trad Mon, 06/04/2012 - 16:31
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Hi Barry,


I am always obliged to help others out, however I must disclaimer the following:


  1. You do this at your own risk
  2. I am not responsible if you get denied support by SBSC due to the CLI changes you implement
  3. You will most likely make your SBSC contract void or in a non support configuration


Sorry if that sounds harsh, but I just need you to be clear that what I am about to do is to purely to help you out because I know that you are stuck, and that you realize what you are doing is not within your certified capacity under Cisco's terms and conditions (Even though many of us may not agree with these terms in whole).



voice register global

mode cme

source-address port 5060

max-dn 5

max-pool 5

dialplan-pattern 1 XXXXX... extension-length 3

authenticate register

date-format D/M/Y

file text

create profile

!

voice register dn 1

call-forward b2bua noan 399 timeout 45

number 300

allow watch

name Amp

label Amp-300

!

voice register pool 1

id mac 0001.0002.0003

type

number 1 dn 1

dtmf-relay sip-notify

username amp password 1234

allow watch

codec g711ulaw



NOTES:


  • The above configuration was done on CME version 4 & 7 not 8 so quite a few things might have changed since as a bit changed between version 4 and 7
  • This may not work the first time and you may need to play with it until you get it right, the above configuration took almost a day to get right for the SIP payphone I had to integrate into the system
  • You need to make sure that the settings on here matches that of the SIP end point, no match means no work and play
  • For the sake of it, THIS is not going to be a supported configuration for you and will most likely put your system out of scope PLEASE understand this before you impliment, and if you are concerned about this then do not proceed.


Good luck, and I highly encourage you to sit the Express UC certification exam before it expires (Not sure what it is being replaced with) and get yourself covered for these types of changes. I understand why Cisco are doing this, but what I don't understand is how their policy leaves an entity high and dry and denied basic support because of a system that is not fully functional and capable of supporting the UC (CCA I'm talking about here).




Cheers,


David.

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