[asterisk-dev] One sip stack to rule them all...

[Eric Klein]

>
> Date: Sun, 8 Oct 2017 19:47:32 -0400
> From: James Finstrom <jfinstrom at gmail.com>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Subject: Re: [asterisk-dev] One sip stack to rule them all....
>
> A large percentage of "PJSIP" Sucks comes down to comfort.  I talked to
> several users at astricon and the summary is:
>
> Every provider that actually provides documentation only gives a chan_sip
> block
> We don't understand how to configure it.
> My customers need ccss.
>


James,
You bring up an issue that was discussed at Devcon. We, as a community,
need to step up and provide this kind of documentation, best practices, and
examples so people can use Asterisk (and in this case PJSIP) properly and
with confidence.

If we want people to use it, we need to show them how to do it in a
supported and stable way.

Eric Klein
VP Operations
Greenfield
Main US +1 805 410 1010
Main UK +44 203 746 6000
Main Il    +972  73 255 7799
Mobile    +972 54 666 0933

*Email *Eric at greenfield.tech
Skype: EricLKlein
Web: www.greenfield.tech
         www.cloudonix.io
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