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The HP Community is where owners of HP products, like you, volunteer to help each other find solutions.
HP Recommended

Hi,

 

It's my first message in this community. I just bought a polycom soundstation IP 5000 and configured in right way that it's working fine. But my problem is like i'm not able to enter DTMF digits into any of the calls.

I want to give DTMF as Send in IN-BAND mode.But when i check with GUI and device config, i couldn't find any option to set the same.Please help me to solve the issue, as it's a prior one.

 

Regards

Randeep.

2 REPLIES 2
HP Recommended

Hello Randeep,

 

welcome to the Polycom community.

 

I would usually refer you to your reseller as without Wireshark Trace , Log Files and Configuration Files it is hard to verify your setup.

 

The SIP release 3.2.3 introduced a change for the Payload Type from 101 to 127.

 

The Standard tone.dtmf.rfc2833Payload="127" could be not compatible with your SIP Server and may needs to be changed.

 

This cannot be done via the GUI or the Web Interface.

 

If you are not already familiar with the Polycom Provisioning please make sure your read => this <= guide describing how to setup a Freeware FTP Server.

 

Please contact your Polycom Reseller for any further support.

 

Best Regards

 

Steffen Baier

------------------------------------------------
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.

Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN
HP Recommended

Hi Steffen.  I'm piggy-backing on this post as we have a current ticket very similar to this.  They just need to know HOW to edit a sip.cfg or any *.cfg for that matter.  I don't play with out VoiceStations & the Asterix box as breaking it may prove painful for me.

I've checked the VoIP & SIP FAQs & the Support.polycom.com Knowledge Base but all tey say is Edit the *.cfg file but no directions as to how.

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