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murf at parsetree.com
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 Posted: Thu Sep 08, 2016 2:13 pm Post subject: [asterisk-users] PJSIP Weirdness, or just my weirdness? |
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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@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@67.215.23.186:54 171a08228b Unavail 0.000
Contact: murftest12/sip:murftest12@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@67.215.23.186:21800 d9a15f4e35 Avail 50.514
Contact: murftest12/sip:murftest12@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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gjoseph at digium.com
Guest
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| Posted: Thu Sep 08, 2016 4:28 pm Post subject: [asterisk-users] PJSIP Weirdness, or just my weirdness? |
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On Thu, Sep 8, 2016 at 1:12 PM, Steve Murphy <murf@parsetree.com (murf@parsetree.com)> wrote:
| Quote: | 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.
| Quote: |
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.
--
George JosephDigium, 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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amessina at messinet.com
Guest
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| Posted: Thu Sep 08, 2016 6:59 pm Post subject: [asterisk-users] PJSIP Weirdness, or just my weirdness? |
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On Thursday, September 8, 2016 1:12:36 PM CDT Steve Murphy wrote:
| Quote: | 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@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@67.215.23.186:54 171a08228b
Unavail 0.000
Contact: murftest12/sip:murftest12@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@67.215.23.186:21800 d9a15f4e35
Avail 50.514
Contact: murftest12/sip:murftest12@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.
|
I just resolved a similar issue with a new Yealink phone and PJSIP. It seems
that Asterisk (depending on many transcoding parameters and types of calls)
may send out a different codec on leg B than it receives on leg A. While less
than optimal for the end user, this is allowed by the RFCs. Yealink doesn't
seem to handle this well. The firmware referenced in this link fixed the
issue for me, as least with my T48G and DAHDI/PJSIP calls.
http://forum.yealink.com/forum/showthread.php?tid=8330&pid=39161#pid39161
| Quote: | 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
|
--
Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery
F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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murf at parsetree.com
Guest
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| Posted: Mon Sep 12, 2016 9:13 pm Post subject: [asterisk-users] PJSIP Weirdness, or just my weirdness? |
|
|
SOLVED!
Many THANKS to George and Anthony! See at the very end, my comments...
On Thu, Sep 8, 2016 at 5:58 PM, Anthony Joseph Messina <amessina@messinet.com (amessina@messinet.com)> wrote:
| Quote: | On Thursday, September 8, 2016 1:12:36 PM CDT Steve Murphy wrote:
| Quote: | 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@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@67.215.23.186:54 171a08228b
Unavail 0.000
Contact: murftest12/sip:murftest12@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@67.215.23.186:21800 d9a15f4e35
Avail 50.514
Contact: murftest12/sip:murftest12@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.
|
I just resolved a similar issue with a new Yealink phone and PJSIP. It seems
that Asterisk (depending on many transcoding parameters and types of calls)
may send out a different codec on leg B than it receives on leg A. While less
than optimal for the end user, this is allowed by the RFCs. Yealink doesn't
seem to handle this well. The firmware referenced in this link fixed the
issue for me, as least with my T48G and DAHDI/PJSIP calls.
http://forum.yealink.com/forum/showthread.php?tid=8330&pid=39161#pid39161
| Quote: | 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
|
|
George's previous message on this thread, plus this one were very helpful
in solving my problems.
I discovered:
1. Mystery #1 was a combination of factors:
1. The max_contacts=1 and remove_existing=yes did in fact get rid of the
dual contact problem! But I still could not get an INVITE to my phone.
2. The use of qualify_frequency=60 in my aor solved the access problem.
For some strange reason, my NAT path was shutting down before 120 sec
for this phone, which is considerably shorter than the other phones, for
some rather unobvious reasons.
2. Mystery #2, as it turned out, actually was because of codec choices. The phones
I was using were not in the list for firmware upgrades, so I had to take a different
approach: actually standardize the codec list for phones and trunks, to all be the
same. I did that, and all the 1-way audio problems went away.
Thanks for the help! I hope that others who have the same problems may find this
message helpful.
murf
--
Steve Murphy
✉ murf at parsetree dot com |
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