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No Dial Tone when accessing line on SIP Trunk, 1

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ega218

MIS
Aug 21, 2004
133
US
I have configured a SIP Trunk and have used 3 instead of 9 for my short code to retreive a line from the SIP trunk. All incoming and outgoing calls are working great. However, I am not able to get a dial tone, either system or network tone when I press 3. It is just silence. I can still dial the number, but I would like to have a dial tone as I have when I press 9 for a regular line I have on my phone system on a PRI trunk.

I have not been able to find any mention on how to obtain a dial tone on a sip line, so the user knows that it is clear to make a phone call.

Any help will be greatly appreciated. Thanks in advance.
 
I have routed the 3N shortcode through an ARS and have set the secondary dial tone to SystemTone and NetworkTone, but neither give me a tone when I dial 3.

However, as I stated before I can still make a phone call just no tone.
 
Short Code
SC: 3N
Feature: Dial
Telephone Number: N
Line Group ID: "ARS Route ID & Name"
Locale: USA
Force Account Code: Unchecked

ARS:
Secondary Dial Tone: Checked. System and Network Tone have been selected with out any success of a dial tone.

Dial Delay Time: 6
Check User Call Barring: Checked
In Service: Checked

CODE: N;
Telephone Number: N"@SIP_IP_Address_From_Provider"
Feature: Dial
Lin Group: 100
 
Still not able to find a dial tone on my SIP line. Everything else works fine.

If anyone has any comments it is appreciated. Thanks.
 
does 3 overlap with anything else on your system? do you have any user short code that could be getting in the way.

Kevin Wing
ACSS Small and Medium Enterprise (SME) Communications
ACS- Implement IP Office
ACA- Implement IP Office
Carousel Industries
 
Hell, I don't even get secondary dial tone after I dial 9, it's just dead air. It's like this on nearly all of my sites, ranging from 4.1 to 6.0.. Luckily my users don't realize it's a problem and continue to dial anyway. These are all on PRI's.
 
You should remove the under score from your ARS table 51 "Sip_Main" it is just bad practice "SIP Main" or "SIPMain".

On your SIP line 17 change your * URI to a different outbound line ID. The * should be used for inbound ambiguous routing only not for outbound although I don't think this is the issue either. Also change the display name here to a *.

Get rid of the underscore characters in your Display Name in the numbered URI on line 17 as well. Again bad practice. I'd set them all to the telephone number for now just for testing.

Dump the dashes in your directory entries they can't be dialed by the system and they won't match on inbound so they should not be there.

After you've done all that let me know what happens. You might want to go back to your original programmer and smack them on the back of the head as all these are basic mistakes.

Kyle Holladay
ACSS SME Communications
ACE Implement: IP Office
MCP/MCTS Exchange 2007
Adtran ATSA, Aruba ACMA

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford
 
Thank you kwing112000,

I have resolved the problem. We are connected on a small community network. There are some other extensions on the other locaitons that start with the number 3, therefore, when would press 3 it would wait to see what the next numbers were before a dialtone appears. This is why I was not getting a dial tone. I change the shortcode dial number to access the ARS to 9 and all works well now.

Thanks
 
That's probably why mine doesn't work. I have SCN to 13 different sites, with every number 1-9 as the beginning of an extension.. O'well..
 
If you already use a 0 for secondary dailtone then you can change this to 01 and 02


Homo sapiens non urinat in ventum

honey, i fried the IP Office !!!

A yard full of tards
 
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