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

[Steve Murphy]

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

[murftest12]
type=auth
auth_type=userpass
username=murftest12
password=SjU3

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:57969


[murftest12]    ; Cisco SPA514G mac=A4:93:4C:FE:1D:A2
type=endpoint
auth=murftest12
transport=transport-udp
aors=murftest12
moh_suggest=default
force_rport=yes
rewrite_contact=yes
rtp_symmetric=yes
dtmf_mode=rfc4733
disallow=all
allow=ulaw ; from phonetype
allow=g722 ; from phonetype
allow=alaw ; from phonetype
allow=alaw ; from phonetype (G.729 replaced with alaw)
direct_media=no
context=phone
rtp_timeout=120
set_var=__phoneid=12
set_var=__contacttypeid=4
set_var=__phonelineid=78
callerid="Steve Murphy" <101>
call_group=2
pickup_group=2
mailboxes=101 at murftest
language=en
send_rpid=yes
send_pai=yes

​OK, that completes the config (I hope).

Now, when I run "pjsip show endpoints, I get:​

SFO02-HostedPBXPJSip-Dev03*CLI> pjsip show endpoints

 Endpoint:  <Endpoint/CID.....................................>
<State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName............................................
...............>
        Aor:  <Aor............................................>
<MaxContact>
      Contact:  <Aor/ContactUri..........................> <Hash....>
<Status> <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>
<BindAddress..................>
   Identify:  <Identify/Endpoint..........................................
...............>
        Match:  <ip/cidr.........................>
    Channel:  <ChannelId......................................>
<State.....>  <Time(sec)>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
 ===========================================================
==============================

 Endpoint:  murftest12/101                                       Not in
use    0 of inf
     InAuth:  murftest12/murftest12
        Aor:  murftest12                                         2
      Contact:  murftest12/sip:murftest12 at 67.215.23.186:54 171a08228b
Unavail       0.000
      Contact:  murftest12/sip:murftest12 at 67.215.23.186:21 d9a15f4e35
Avail        50.514
  Transport:  transport-udp             udp      0      0  0.0.0.0:57969

​ Note that there are TWO Contact: entries! one Avail, the other Unavail...
the show endpoints doesn't display all the URL, but the show contacts does:

​  Contact:  murftest12/sip:murftest12 at 67.215.23.186:21800  d9a15f4e35
Avail        50.514
  Contact:  murftest12/sip:murftest12 at 67.215.23.186:54004  171a08228b
Unavail       0.000

None of my other phones have two contacts listed.... and this phone, a
cisco-spa-514, has just one sip account...

The trouble is, when I try to call it.... sometimes the INVITE is directed
to the "Unavail" entry, and the call never completes. The phone doesn't
even ring then. Any ideas? I tried to get the "Unavail" entry out... I
removed it from the db, I rebooted the phone, restarted asterisk, and it is
still there.

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?

many thanks...

murf
-- 

Steve Murphy


✉  murf at parsetree dot com
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