SOUNDSTATION IP5000 WITH H323

cooleyr
Occasional Visitor

SOUNDSTATION IP5000 WITH H323

I have 6 Sounstation IP 5000 and have configured and registered them on Cisco Call Manager 9.1. I have tested these for internal and external calls using MGCP gateways to ISDN 30 and all works fine.

 

I have a problem though with calls through H323 gateways which are registered to the same Call Manager.

 

Is this a known problem?  Any help appreciated.

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2 REPLIES 2
SteffenBaierUK
Polycom Employee & Community Manager

Re: SOUNDSTATION IP5000 WITH H323

Hello cooleyr,

welcome to the Polycom Community.

The SSIP Phones have been tested and verified against the CUCM platform so I am not aware of any issues.

 

  • What trouble are you experiencing and do other 3rd party SIP Phones have the same issue ?

  • Has Cisco been consulted yet?


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

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The title Poly Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. All posts and words are my own & do not represent the views of Employer.

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Message 2 of 3
cooleyr
Occasional Visitor

Re: SOUNDSTATION IP5000 WITH H323

Steffen

 

Thank you for the information.

 

I ahve other polycom phones but they are Analogue phone and are connected through an ATA.  We also have the Cisco 7936 phones which also work. However we have been advised that only the Polycom Soundstation phones are now available and these are the first I have created using Third Party SIP.

 

These are configured and work correctly through MGCP gateways however when I make a call through an H.323 gateway I get an engaged tone.  I was wondering is there was any extra configuration required to allow this communication to work through H.323 gateways.

 

I have not yet been to Cisco with this issue because I configure a Cisco phone with the same credentials except for Third Party SIP it works correctly.  I though this to be a Polycom configuration requirement.

 

Regards

 

Bob

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