SIP POLYCOM

[colec12]

Strange issue, I am trying to deploy a Polycom Duo conference phone and can't seem to get it working properly. I can make calls to the phone with out issue but cannot make calls from Polycom.
Running CM 6.0 ans ASM 6.1. Looks like an issue with either the config in the phone or ASM.

Thanks for your help.

 

[gwebster]

Looks like an issue with either the config in the phone or ASM". That is certainly two possibilities. It could also be bug in either the phone of ASM. Do you have any other SIP devices working on this system? If yes, can it register with the user/password that you set up for the Polycom and make/receive calls?

 

[john8lee]

what are you seeing when you li trac tac <trunk tac code>?
is the digitmap on the polycom built correctly. I just installed a Polycom Trio and had to fight the digitmap

 

[colec12]

There is at least one Sip Polycom working. The phone does register.
I did not see a place to configure the digitmap on the phone itself.

 

[john8lee]

are you able to trace the trunk to see what CM is seeing? What error are you hearing/seeing?
The Polycom trio has a wegpage configurator which is where I found the digitmap under SIP setting tab

 

[Speewah]

The Digitmap is configured via the web interface of the phone.
This guide may help

http://support.polycom.com/global/d...s/voice/Understanding_Digit_Maps_Tech_Tip.pdf

 

[john8lee]

This is the digitmap we use in a 4-digit extn environment with ARS triggered on 9
911|0T|011xxx.T|9[1-9]xxxxxxxxxx|9[2-8]xxxxxxxxx|[1-8]xxxT

 

[colec12]

List trac tac shows the following:


time data

12:56:44 TRACE STARTED 05/03/2017 CM Release String cold-00.1.510.1-21061
12:57:26 SIP<INVITE sip:7237@hvcc.edu:5060;transport=tcp;user=phone
12:57:26 SIP<SIP/2.0
12:57:26 Call-ID: 17bb9c6a-2eb391fd-ed641544@10.2.231.250
12:57:26 SIP>SIP/2.0 404 Not Found
12:57:26 Call-ID: 17bb9c6a-2eb391fd-ed641544@10.2.231.250

 

[john8lee]

You have other SIP endpoints that work in both directions?
i know setting up the routing tab on System Manager is a pain, but with it working in one direction it sounds like the outbound routing is right but the inbound is missing something

 

[colec12]

I do have other sip end points working in both directions. I get a fast busy as soon as I try to make a call from this particular end point.

 

[john8lee]

were you able to find the digit map?
can you dial the polycom like a cell phone by entering all numbers then hitting the dial button?

 

[Speewah]

Compare the digit map with a working phone. Check traceSM where you will get more verbose output than 'list trace tac' on CM

 

[G van Hamburg]

Check your aar too.

Plan your work............Work your plan

 

[colec12]

Digit mapping looks good and so does the aar table. I am about to give in and call support.

 

[G van Hamburg]

My previous answer was a bit short. I’d start in system manager and compare the working user with the non-working user. See if you added the application sequence and also a CM profile. If the users match then look in CM if the stations look the same and if an OPS exists pointing to AAR. Last thing is to login into the polycom phones and compare the sip settings. When the SIP signaling, trunks, aar and route-pattern is ok and tested, adding a sip phone should be easy. I have done this a million times.

Plan your work............Work your plan