Unable to change number of rings before voicemail picks up.

[rayg75]

I have a 7100 and a Polycom phone.  I followed the instructions in the admin guide to change the number of rings before voice mail picks up.  It does not appear to make any difference.  Any suggestions are appreciated.

[Anonymous]

Just to confirm you went to Voice > User Accounts > selected the extension > clicked Edit > and then changed this on the call coverage tab?  Can you reply with a screenshot for this extension? 

Thanks,

Matt

[rayg75]

Yes that is what I changed.  I tried using the default call coverage list and a custom list.  Attached are the screen shots. On the one with the custom call list I also tried putting in her extension and it did not make a difference.

[Anonymous]

Thanks for the confirmation.  If you can supply the current configuration and the output from both a debug voice verbose as well as debug sip stack messages while the issue is recreated, I would be happy to troubleshoot further with you.  You can submit them on our FTP server with the instructions below:

Open Internet Explorer web browser on their PC
Type the following URL:  ftp://ftp.adtran.com

Press the Alt key, click View, and then click Open FTP Site in Windows Explorer

Double-click the "Incoming" folder
Drag and drop files from PC into the Internet Explorer window

Reply to this post with the exact filenames used so we can retrieve the files

Thanks,

Matt

[rayg75]

I will be happy to provide you that information if you will tell me how to to do.  I know how to telnet into the system and get to the enable prompt.  If you could provide instructions from there I would appreciate it.   I also have some more information that might help. No one can leave a voice mail for the user.  When you dial her extension it rings twice and then beeps and disconnects.  They have three polycom phones and from what I can tell they are all identical, the other two users have not complained of any problems.  They also have several Adtran phones and they all work fine. I have uploaded the file config-*******-07-01-2013.cfg.  I tried to change the name of the file after I uploaded it but I could not.  After you read this post please edit this comment to remove the name of the file. It is extension 101 that is having the problem

Message was edited by: matt - edited filename as requested

[Anonymous]

Thanks for the information.  I retrieved the configuration file and edited out the filename from your last post as you requested. 

Here are the instructions to enable the debug:

Once you get into enable mode you need to enter the command debug sip stack messages and then separately debug voice verbose.  These commands will generate a lot of output, especially if there is a lot of voice traffic.  You should use a terminal program that will let you log directly to a file or make sure the scroll back buffer is set very high so you can catch all of the output.  After you have recreated the issue you can turn off the debugs by entering the command undebug all or u a for short.  Then copy and paste all of the debug output into a text document or pull the file the terminal program was logging directly to. As a best practice it is recommended to be ready to recreate the issue as soon as you enter the debug commands and turn off the debugs immediately after to minimize the length of the output.  It also helps to collect the debug after hours or another time when there is minimal voice traffic, but that is not completely necessary. 

If the debug does not shed any light I would be curious to see if rebooting makes a difference.  If not you may want to try deleting and recreating this user and possibly upgrading the firmware.

Thanks,

Matt

[rayg75]

I uploaded a file call Rglennputty with the debug files.  I am also going to create another ticket about not being able to transfer from the Adtran phones. You already have the cfg file and the debug files.  I will reference this ticket in the new one I create. 

[Anonymous · Accepted solution]

The debug revealed the following call flow:

• The call comes in on the PRI from a cell phone to the main DID number ending in 4232, which gets routed to ring group 8101 since it it configured with an alias of 4232.
• The ring group 8101 sends a call out to extensions 100, 101, 102, and 112.
• After the configured number of rings with no answer from the extensions the ring group coverages to the auto attendant at 8201.
• The caller uses the dial by extension feature of the auto attendant to input extension 101 and the call is transferred to extension 101.
• Extension 101 does not answer the call, and seems to be configured with a call forward to an invalid 9 digit external number ending in 3697 which appears to be causing the issue.

The call forward can be seen from the SIP 302 Moved Temporarily message in the debug:

22:56:33:774 SIP.STACK MSG     Rx: UDP src=10.10.20.2:5060 dst=10.10.20.1:5060

22:56:33:774 SIP.STACK MSG         SIP/2.0 302 Moved Temporarily

22:56:33:774 SIP.STACK MSG         Via: SIP/2.0/UDP 10.10.20.1:5060;branch=z9hG4bK-2a9c4-a6727b6-2965b2f

22:56:33:775 SIP.STACK MSG         From: "Cell Phone " <sip:XXXYYY6651@10.10.20.1:5060;transport=UDP>;tag=50dd4a8-a0a0a01-13c4-2a9c4-bce9102e-2a9c4

22:56:33:775 SIP.STACK MSG         To: "Name Replaced" <sip:101@10.10.20.1:5060>;tag=E2D61690-BFAAC60D

22:56:33:775 SIP.STACK MSG         CSeq: 1 INVITE

22:56:33:775 SIP.STACK MSG         Call-ID: 514b2f0-a0a0a01-13c4-2a9c4-ddde8266-2a9c4@10.10.20.1

22:56:33:775 SIP.STACK MSG         Contact: <sip:YYYZ3697@10.10.20.1>

22:56:33:775 SIP.STACK MSG         User-Agent: PolycomSoundPointIP-SPIP_670-UA/3.1.3.0439

22:56:33:776 SIP.STACK MSG         Accept-Language: en

22:56:33:776 SIP.STACK MSG         Diversion: "Name Replaced" <sip:101@10.10.20.1:5060>;reason=no-answer

22:56:33:776 SIP.STACK MSG         Content-Length: 0

If you go to the Polycom phone menu for extension 101 you can disable the call forward by going to Menu > Features > Forward > select No Answer > and select the Disable softkey.  I would also recommend making sure no other call forwards are enabled on the phone besides the one for No Answer.

Thanks,

Matt

[rayg75]

Thank you very much I don't think I would have ever found that. 

[Anonymous]

Glad to help.  Let me know if that does not fix the issue.