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Polycom IP450 - Dial from Missed Calls

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Occasional Advisor

Polycom IP450 - Dial from Missed Calls

Hi All,

 

We are unable to successfully dial numbers from our missed calls, Received Calls, Placed Calls or speed dial on the above handset. We have tried using firmware 4.03 4.04 and 4.10.

 

When a number is selected from any of the above lists, the phone does try to make the call.sip: 0203xxxxxx is displayed on the LCD. Then eventually times out.

 

I have included a SIP log just in case... Any help will be very much appreciated.

 

Thanks,

Synety

 

 

         INVITE sip:01134166007;transport=udp SIP/2.0

         Via: SIP/2.0/UDP 172.17.2.101;branch=z9hG4bK9c655e8c46941F9F;received=86.12.191.141

         Max-Forwards: 70

         Contact: <sip:441164244020@172.17.2.101;transport=udp>

         To: <sip:01134166007>

         From: "4020" <sip:441164244020@sip.synety.com>;tag=97EA806A-16495AB5

         Call-ID: 4246665e-78a35af9-41ccbc60@172.17.2.101

         CSeq: 1 INVITE

         Accept-Language: en

         Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

         Content-Type: application/sdp

         Supported: 100rel, replaces

         User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.1.0.84959

         Allow-Events: conference, talk, hold

         Content-Length: 294

 

 

Message 1 of 2
1 ACCEPTED SOLUTION

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Polycom Employee & Community Manager

Re: Polycom IP450 - Dial from Missed Calls

Hello Synety,

welcome back to the Polycom Community.

If you have a look at your INVITE To: Field.

 

According to the RFC3261:

 

To: The To header field contains the address of record whose
           registration is to be created, queried, or modified.  The To
           header field and the Request-URI field typically differ, as
           the former contains a user name.  This address-of-record MUST
           be a SIP URI or SIPS URI.

You are just sending the number and not a full URI.

 

This maybe simply caused by a misconfiguration of the phone and the community FAQ holds enough details on what a minimal configuration should contain. You should also check the registration status of the phone.

 

For further support please contact your Polycom reseller and / or Polycom support directly.

Best Regards

Steffen Baier

Polycom Global Services

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

View solution in original post

Message 2 of 2
1 REPLY 1
Highlighted
Polycom Employee & Community Manager

Re: Polycom IP450 - Dial from Missed Calls

Hello Synety,

welcome back to the Polycom Community.

If you have a look at your INVITE To: Field.

 

According to the RFC3261:

 

To: The To header field contains the address of record whose
           registration is to be created, queried, or modified.  The To
           header field and the Request-URI field typically differ, as
           the former contains a user name.  This address-of-record MUST
           be a SIP URI or SIPS URI.

You are just sending the number and not a full URI.

 

This maybe simply caused by a misconfiguration of the phone and the community FAQ holds enough details on what a minimal configuration should contain. You should also check the registration status of the phone.

 

For further support please contact your Polycom reseller and / or Polycom support directly.

Best Regards

Steffen Baier

Polycom Global Services

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

View solution in original post

Message 2 of 2