Author Topic: Yate & RFC-2806  (Read 4307 times)

ogogon

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Yate & RFC-2806
« 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.

marian

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Re: Yate & RFC-2806
« Reply #1 on: November 14, 2014, 03:42:40 AM »
Hi,

Can you describe what exactly do you need?

ogogon

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Re: Yate & RFC-2806
« Reply #2 on: November 14, 2014, 09:17:24 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>
« Last Edit: November 14, 2014, 09:20:16 AM by ogogon »

marian

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Re: Yate & RFC-2806
« Reply #3 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.

ogogon

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Re: Yate & RFC-2806
« Reply #4 on: November 14, 2014, 09:15:10 PM »
Thank you very much.

Ogogon.