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New Member

Polycom IP6000 call fails after 10 digits dialed.

Hi,
I have a Polycom IP6000 configured on our UCM 9.1 environment via SIP.  Calls inside the network and inbound work fine.  Outbound calls that are 7 digits work fine (9 + Local number).  Outbound calls with an area code fail on the 10th digit before I can dial the last digit in the phone number (9 + area code + local number).  It is like it is trying to match the route pattern before the complete number is entered.  Any suggestions on this?

1 ACCEPTED SOLUTION

Accepted Solutions

Usually SIP phones will have

Usually SIP phones will have some sort of configurable dialplan settings or they will pull down a dialplan file via TFTP.  Do you see anything configurable on the phone's webpage?  Most non-Cisco SIP phones send the digits enbloc so it's up to the phones to have knowledge of the full dial plan so they know to send the call after a certain number of digits have been dialed.

11 REPLIES

for me, it seems, overlapping

for me, it seems, overlapping patterns.

1) the CSS applied to polycom phone is same as other IP phones?

2) if not, verify the order of the partitions included in that CSS?

3) is there any route pattern starting with 9 and have urgent priority checked?

4) can you check that with Dialed Number Analyzer and post the DNA result logs here for this phone?

 

//Suresh

Please rate all the helpful posts

//Suresh Please rate all the useful posts.
New Member

It is using the same CSS as

It is using the same CSS as all of our other IP phones at this site.  I did DNA and it is routing our the correct end device (PRI) as I would expect it to.  That part looks normal.  I did just try something different though.  I have been testing doing off-hook dialing (that is when it fails).  I tried doing on hook dialing where I put in 9 (727)638-XXXX and press the dial button and the call goes through.  So it is something different between off-hook and on-hook dialing. 

Seems there may be a 8 or 9

Seems there may be a 8 or 9 digits matching pattern with urgent priority checked. We can collect the ccm traces for the working (on hook dialing) call and non-working (offhook dialing) call. Is this happening only for Polycom phones or cisco sip phones too? How about the SCCP phones?
//Suresh Please rate all the useful posts.

Usually SIP phones will have

Usually SIP phones will have some sort of configurable dialplan settings or they will pull down a dialplan file via TFTP.  Do you see anything configurable on the phone's webpage?  Most non-Cisco SIP phones send the digits enbloc so it's up to the phones to have knowledge of the full dial plan so they know to send the call after a certain number of digits have been dialed.

New Member

Hello Cisco Community,I'm

Hello Cisco Community,

I'm experiencing almost the same problem with a Polycom IP7000 on our CUCM 8.6.2.

We were able to make the polycom dial 7 digits call (with 9 as prefix), also calls inside the network (4 digits extensions), but we are not able to make this device dial long distance (1) or international calls (011). We use client matter codes for our LD and INT calls, and even though we have configured the same route pattern (they called it digitmap) on the SIP Phone web interface, after the 10+ digits, there's no tone to enter the CMC (Client Matter Code). 

Does anyone know how to fix this? 

Thank you for your help and support.

Sincerely,

PE

Are you dialing digit by

Are you dialing digit by digit with dialtone or are you sending the digits all at once? If doing digit by digit, pressing "#" should force the call to go out.  If that works, there's something wrong with the digitmap you have configured that is waiting for more digits.

New Member

Thank you for your answer

Thank you for your answer Brian Meade.

I'm doing it both ways (digit by digit and also all at once). Pressing # key force the call to go out (but there's no tone to enter the Client Matter Code). 

Using our phones cisco IP Phones (6921 - 8961 and 9971) there are no problems at all. 

I think I actually tested

I think I actually tested this once but it was using a SIP Trunk instead of 3rd party SIP device.  My findings were that the method for entering CMC/FAC seems to be Cisco proprietary and thus only works with Cisco endpoints.  You can get a capture and see if CUCM is sending the 183 Session Progress to play the tone but there may be no way to get this to work.  Some carriers allow you to set up FAC/CMC on their side for billing and that should work with all phones.

New Member

Thank you Brian. I will keep

Thank you Brian. I will keep testing and definitely update you with any progress. Thanks for your time.

PE

New Member

Thank you, didn't realize it

Thank you, didn't realize it had a web interface and its own dial plan.  I tweaked that and working!

New Member

I had the same issue. On the

I had the same issue. On the SoundStation 6000 there is a pre-configured dial plan. I removed the dial plan altogether and the polycom resorted to the dial plan that's configured in CM. Only issue with that is, it took longer for the call to go out i.e. 4 digit dialing internal, 10 digit dialing, and Intl dialing as well. I copied the the route patterns that were in CM into the Digit Map on the polycom and calling went out with an hitch. Hope this helps.

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