[asterisk-users] PJSIP Weirdness, or just my weirdness?

[George Joseph]

On Thu, Sep 8, 2016 at 1:12 PM, Steve Murphy <murf at parsetree.com> wrote:

> Hello!
>
> Oh, wise ones, ponder with me over two of the surprises that
> populate the universe!
>
>
> I have a phone, that I sometimes cannot reach, connected via pjsip.
> It can call other extensions just fine, it can call out over a
> trunk to my cell, all is well, but getting a call? Forget it most of the
> time.
>
> Here is all the config relevant to that phone:
>
>
> [murftest12]
> type=aor
> qualify_frequency=1992
> max_contacts=2
>

*max_contacts = 1*
*remove_existing = yes*
*This should take care of mystery 1.*


>
>
> MYSTERY #2:
>
> The above cisco-spa, when it calls out over the trunk, all is well,
> wonderful 2-way audio.
> But when I do the same operation from my yealink phones, I get my cell
> with one-way audio.
> It's a classic NAT situation: the phone system is in a droplet at digital
> ocean, but my phones are here at home behind a NAT. I see only 3 NAT
> related options:
>
> force_rport
> rtp_symmetric
> rewrite_contact
>
> and I set them all to "yes", and they can call each other, but as
> explained, in
> dialing out thru a trunk, the yealinks get one-way audio...
>
> Any more NAT options?
>

Do the phones have settings for NAT?  If so, turn them on.
Does your home router have a setting for SIP-ALG?  If so *TURN IT OFF!!*
Give that a shot.


>
> many thanks...
>
> murf
> --
>
> Steve Murphy
>
>
> ✉  murf at parsetree dot com
>
>
> --
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-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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