Yate Community Forum

Yate server => Other Yate server issues => Topic started by: ogogon on November 13, 2014, 06:59:05 PM

Title: Yate & RFC-2806
Post by: ogogon on November 13, 2014, 06:59:05 PM

Dear colleagues!

Tell me, please, whether the Yate has support of RFC-2806?
This additional schemes in SIP URI. For example: "tel:", "fax:", etc.

Unfortunately, provider's not opensource softswitch, on which I need to register, without these schemes can not. But the Asterisk from these schemes, falls into a riot.

Ogogon.

Title: Re: Yate & RFC-2806
Post by: marian on November 14, 2014, 03:42:40 AM

Hi,

Can you describe what exactly do you need?

Title: Re: Yate & RFC-2806
Post by: ogogon on November 14, 2014, 09:17:24 AM

Quote from: marian on November 14, 2014, 03:42:40 AM

Hi,

Can you describe what exactly do you need?

Will Yate properly understood in the incoming invite fields without "sip:"?
Е.g.,

Code: [Select]

From: <tel:81234567890;cpc-rus=1;phone-context=+7>;tag=sbc1308j3izzjz8-CC-2
P-Called-Party-ID: <tel:+71234567890>
P-Asserted-Identity: <tel:81234567890;phone-context=+7>

Title: Re: Yate & RFC-2806
Post by: marian on November 14, 2014, 09:22:37 AM

Yes.
In this case the number in From header 'tel:' uri will be set in 'caller' parameter.

Title: Re: Yate & RFC-2806
Post by: ogogon on November 14, 2014, 09:15:10 PM

Thank you very much.

Ogogon.