Asterisk does not use registered port for invitation since upgrade

jssilva · May 16, 2016, 10:47am

Hello,

Since a while back, asterisk stopped being able to call one of my extensions, an android phone with csipsimple. It can place calls though. Asterisk and all extensions are on a local lan, no nat or firewalls. Version is 13.6.0-1 running on a openwrt router since a number of years with only few problems.

As much as I can see, the extension registers using a random port, not 5060, but asterisk invites using 5060 which the extension is not listening.

sip show peer SGS4C

Name : SGS4C
Description :
Secret :
MD5Secret :
Remote Secret:
Context : MySets
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : default
Language :
Tonezone :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : “” <32>
MaxCallBR : 384 kbps
Expire : 104
Insecure : no
Force rport : No
Symmetric RTP: No
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Path support : No
Path : N/A
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
Outb. proxy : 192.168.1.214
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 192.168.1.214
Addr->IP : 192.168.1.214:49364
Defaddr->IP : 192.168.1.214:5060
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: SGS4C
SIP Options : (none)
Codecs : (alaw|ulaw)
Auto-Framing : No
Status : UNREACHABLE
Useragent : CSipSimple_jflte-21/r2470
Reg. Contact : sip:SGS4C@192.168.1.214:49364;ob
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

=== Registration ===

<— SIP read from UDP:192.168.1.214:49364 —>
REGISTER sip:192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.214:49364;rport;branch=z9hG4bKPjFbGEvIcONqRggRn3cuqBlB4kCu1fXdRd
Route: sip:192.168.1.2:5060;transport=udp;lr
Max-Forwards: 70
From: sip:SGS4C@192.168.1.2;tag=bBnizTdFzrxc3TpsP43Bu36RversqlcY
To: sip:SGS4C@192.168.1.2
Call-ID: x.OZJIO0t9uc4cRta3LpA7h1xJrfDssu
CSeq: 11903 REGISTER
User-Agent: CSipSimple_jflte-21/r2470
Contact: sip:SGS4C@192.168.1.214:49364;ob
Expires: 0
Authorization: Digest username=“SGS4C”, realm=“MyPBX”, nonce=“0a89bcff”, uri=“sip:192.168.1.2:5060”, response=“8c4b44515dfd168d34febadbdbb032ce”, algorithm=MD5
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.214:49364 (no NAT)

<— Transmitting (no NAT) to 192.168.1.214:49364 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.214:49364;branch=z9hG4bKPjFbGEvIcONqRggRn3cuqBlB4kCu1fXdRd;received=192.168.1.214;rport=49364
From: sip:SGS4C@192.168.1.2;tag=bBnizTdFzrxc3TpsP43Bu36RversqlcY
To: sip:SGS4C@192.168.1.2;tag=as0adbee88
Call-ID: x.OZJIO0t9uc4cRta3LpA7h1xJrfDssu
CSeq: 11903 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 0
Date: Mon, 16 May 2016 10:25:31 GMT
Content-Length: 0

=== Invitation === (ext. 31 calling 32)

Reliably Transmitting (no NAT) to 192.168.1.214:5060:
OPTIONS sip:SGS4C@192.168.1.214:49364;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4b21e801
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.2;tag=as55e01ed0
To: sip:SGS4C@192.168.1.214:49364;ob
Contact: sip:asterisk@192.168.1.2:5060
Call-ID: 5fbd9f4d6ba23fce62740d6c37ed5d9a@192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: ZON ZON Phone 1.0.0 (1)
Date: Mon, 16 May 2016 10:34:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
…

<------------>
Really destroying SIP dialog ‘52a1f73f35ad33334e4d94c3212325a8@192.168.1.2:5060’ Method: INVITE
[May 16 10:35:03] WARNING[1226][C-00000011]: app_dial.c:2411 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
Scheduling destruction of SIP dialog ‘80adc468-c0a801fe-13c4-45028-db5bc2-11fb37d4-db5bc2’ in 6400 ms (Method: INVITE)

=== extract of sip.conf ===

[general]
sipdebug=yes
recordhistory=yes
dumphistory=yes
context=default ; default context for incoming calls
allowguest=no ; disable unauthenticated calls
srvlookup=yes ; enabled DNS SRV record lookup on outbound calls
tcpenable=no ; disable TCP support
transport=udp
bindport=5060 ; porta sempre antes do endereço
useragent=ZON ZON Phone 1.0.0 (1); ANDROIDv2.3.6; i9000
nat=no
defaultexpiry=300
localnet=192.168.0.0/255.255.0.0
qualify=yes
fromuser=MyPBX
defaultuser=MyPBX
…
; create a template for our devices
all-phones
type=friend
context=OtherSets
host=dynamic
nat=no
remotesecret=mypassword
secret=mypassword
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.0.0
qualify=200
canreinvite=no
…
SGS4C
outboundproxy=192.168.1.214
defaultip=192.168.1.214
callerid=32
defaultuser=SGS4C
context=MySets

This extension is only seldomly used, so I only noticed now but it used to work correctly and I suspect it stopped working after an asterisk upgrade, as I said before.

Can somebody help, please?
Thanks,
jss

david551 · May 16, 2016, 11:04am

Please mark your logs as preformatted text, otherwise everything between angle brackets gets lost, including important information, like the Contact header.

jcolp · May 16, 2016, 11:10am

The provided REGISTER seems to be an unregistration, not a registration. The Expires is set to 0. This would be why the device could not be dialed.

jssilva · May 16, 2016, 12:14pm

@david551 Quite right, sorry. Tried to edit but the buttons aren’t there. Will do so next time.

@jcolp Thank you for helping.
I wonder why that Expire: 0 is there. Tried once more, this time by deactivating and re-activating the account on the extension, the Expire is now 900 as set:

=== Registration, please notice the Unreachable message ===

<--- SIP read from UDP:192.168.1.214:49364 --->
REGISTER sip:192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.214:49364;rport;branch=z9hG4bKPjLNwYtO-Rppc7x3qPMvanbll1Ih7wqvMU
Route: <sip:192.168.1.2:5060;transport=udp;lr>
Max-Forwards: 70
From: <sip:SGS4C@192.168.1.2>;tag=IW0Gio2vSqlfJ15-JK42H5EPi7cv6ZEz
To: <sip:SGS4C@192.168.1.2>
Call-ID: UTSzFGjqXnS.bP88y5zBqIQKJS50a2sW
CSeq: 59854 REGISTER
User-Agent: CSipSimple_jflte-21/r2470
Contact: <sip:SGS4C@192.168.1.214:49364;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.214:49364 (no NAT)
Sending to 192.168.1.214:49364 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.214:49364 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.214:49364;branch=z9hG4bKPjLNwYtO-Rppc7x3qPMvanbll1Ih7wqvMU;received=192.168.1.214;rport=49364
From: <sip:SGS4C@192.168.1.2>;tag=IW0Gio2vSqlfJ15-JK42H5EPi7cv6ZEz
To: <sip:SGS4C@192.168.1.2>;tag=as20c79ca1
Call-ID: UTSzFGjqXnS.bP88y5zBqIQKJS50a2sW
CSeq: 59854 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="MyPBX", nonce="0f2398b6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'UTSzFGjqXnS.bP88y5zBqIQKJS50a2sW' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.214:49364 --->
REGISTER sip:192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.214:49364;rport;branch=z9hG4bKPjlFz5YFk3gc6gTAmQnrkvfxDn4Jf51Bxj
Route: <sip:192.168.1.2:5060;transport=udp;lr>
Max-Forwards: 70
From: <sip:SGS4C@192.168.1.2>;tag=IW0Gio2vSqlfJ15-JK42H5EPi7cv6ZEz
To: <sip:SGS4C@192.168.1.2>
Call-ID: UTSzFGjqXnS.bP88y5zBqIQKJS50a2sW
CSeq: 59855 REGISTER
User-Agent: CSipSimple_jflte-21/r2470
Contact: <sip:SGS4C@192.168.1.214:49364;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="SGS4C", realm="MyPBX", nonce="0f2398b6", uri="sip:192.168.1.2:5060", response="960eaa0b7d51174b13bd48478f0195e1", algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.214:49364 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.214:5060:
OPTIONS sip:SGS4C@192.168.1.214:49364;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK382bda11
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as3486749f
To: <sip:SGS4C@192.168.1.214:49364;ob>
Contact: <sip:asterisk@192.168.1.2:5060>
Call-ID: 3e5d77ea2f7a81764eac6dc9268c781f@192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: ZON ZON Phone 1.0.0 (1)
Date: Mon, 16 May 2016 12:02:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.1.214:49364 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.214:49364;branch=z9hG4bKPjlFz5YFk3gc6gTAmQnrkvfxDn4Jf51Bxj;received=192.168.1.214;rport=49364
From: <sip:SGS4C@192.168.1.2>;tag=IW0Gio2vSqlfJ15-JK42H5EPi7cv6ZEz
To: <sip:SGS4C@192.168.1.2>;tag=as20c79ca1
Call-ID: UTSzFGjqXnS.bP88y5zBqIQKJS50a2sW
CSeq: 59855 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 900
Contact: <sip:SGS4C@192.168.1.214:49364;ob>;expires=900
Date: Mon, 16 May 2016 12:02:23 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'UTSzFGjqXnS.bP88y5zBqIQKJS50a2sW' in 32000 ms (Method: REGISTER)
[May 16 12:02:24] NOTICE[28632]: chan_sip.c:29343 sip_poke_noanswer: Peer 'SGS4C' is now UNREACHABLE!  Last qualify: 0
Really destroying SIP dialog '3e5d77ea2f7a81764eac6dc9268c781f@192.168.1.2:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.1.214:5060:
OPTIONS sip:SGS4C@192.168.1.214:49364;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5d6045c0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as497e69ab
To: <sip:SGS4C@192.168.1.214:49364;ob>
Contact: <sip:asterisk@192.168.1.2:5060>
Call-ID: 128e371c1a0db68c063a392b6a2fae53@192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: ZON ZON Phone 1.0.0 (1)
Date: Mon, 16 May 2016 12:02:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

…

<------------>
Really destroying SIP dialog '2e638da61a15037c55c432722dceb6ba@192.168.1.2:5060' Method: INVITE
[May 16 12:02:37] WARNING[6942][C-00000017]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Scheduling destruction of SIP dialog '80adc668-c0a801fe-13c4-45028-db7048-47d0a5da-db7048' in 6400 ms (Method: INVITE)

Looking forward to your appreciated help
jss

jcolp · May 16, 2016, 12:15pm

What is the actual console output including verbose and notice messages?

jssilva · May 16, 2016, 12:29pm

I’m not sure I understand you.
Do you mean to set sip debug off and copy the whole console output on a registration and call failure?
It can be very long because I have 4 trunks, several ata’s and extensions.

jcolp · May 16, 2016, 12:30pm

Yes, for example the Dial line doesn’t contain what was actually dialed - so I’d like to confirm that what you’re dialing is what is registered.

jssilva · May 16, 2016, 1:49pm

Sorry for the delay, I was expecting the site to be auto-refreshed, as it refreshes the time since posted, but it isn’t, and so I was waiting for your reply which was never seen until I hit refresh.
By the way, thank you for your time and prompt support.

So, I deactivated the extension 32, hit asterisk -r, activated the extension and dialed 32 from extension 31, which is an ata. I deleted some messages crossed with the trunks.

Asterisk 13.6.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.6.0 currently running on OpenWrt-casa (pid = 28585)

<--- SIP read from UDP:192.168.1.214:49364 --->
REGISTER sip:192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.214:49364;rport;branch=z9hG4bKPjEbuc4abKdKAebbqXuakP5InFX0bzUiB.
Route: <sip:192.168.1.2:5060;transport=udp;lr>
Max-Forwards: 70
From: <sip:SGS4C@192.168.1.2>;tag=SXmXH7gkZT1ppWomY-hDZ3OnnS5tLmfX
To: <sip:SGS4C@192.168.1.2>
Call-ID: BJk34138MPLpS4pA-wnErN2AcNfdGQ0f
CSeq: 45139 REGISTER
User-Agent: CSipSimple_jflte-21/r2470
Contact: <sip:SGS4C@192.168.1.214:49364;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.214:49364 (no NAT)
Sending to 192.168.1.214:49364 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.214:49364 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.214:49364;branch=z9hG4bKPjEbuc4abKdKAebbqXuakP5InFX0bzUiB.;received=192.168.1.214;rport=49364
From: <sip:SGS4C@192.168.1.2>;tag=SXmXH7gkZT1ppWomY-hDZ3OnnS5tLmfX
To: <sip:SGS4C@192.168.1.2>;tag=as16edce00
Call-ID: BJk34138MPLpS4pA-wnErN2AcNfdGQ0f
CSeq: 45139 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="MyPBX", nonce="4240b6a3"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'BJk34138MPLpS4pA-wnErN2AcNfdGQ0f' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.214:49364 --->
REGISTER sip:192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.214:49364;rport;branch=z9hG4bKPj8lHlyh0LIZnp91sYMufp.bG2q.i.0l2x
Route: <sip:192.168.1.2:5060;transport=udp;lr>
Max-Forwards: 70
From: <sip:SGS4C@192.168.1.2>;tag=SXmXH7gkZT1ppWomY-hDZ3OnnS5tLmfX
To: <sip:SGS4C@192.168.1.2>
Call-ID: BJk34138MPLpS4pA-wnErN2AcNfdGQ0f
CSeq: 45140 REGISTER
User-Agent: CSipSimple_jflte-21/r2470
Contact: <sip:SGS4C@192.168.1.214:49364;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="SGS4C", realm="MyPBX", nonce="4240b6a3", uri="sip:192.168.1.2:5060", response="c009839852e7a0ffab1267a930bdfb09", algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.214:49364 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.214:5060:
OPTIONS sip:SGS4C@192.168.1.214:49364;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3452eaa2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as6782d758
To: <sip:SGS4C@192.168.1.214:49364;ob>
Contact: <sip:asterisk@192.168.1.2:5060>
Call-ID: 7f8a5687235feab665f871c03641315c@192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: ZON ZON Phone 1.0.0 (1)
Date: Mon, 16 May 2016 13:34:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.1.214:49364 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.214:49364;branch=z9hG4bKPj8lHlyh0LIZnp91sYMufp.bG2q.i.0l2x;received=192.168.1.214;rport=49364
From: <sip:SGS4C@192.168.1.2>;tag=SXmXH7gkZT1ppWomY-hDZ3OnnS5tLmfX
To: <sip:SGS4C@192.168.1.2>;tag=as16edce00
Call-ID: BJk34138MPLpS4pA-wnErN2AcNfdGQ0f
CSeq: 45140 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 900
Contact: <sip:SGS4C@192.168.1.214:49364;ob>;expires=900
Date: Mon, 16 May 2016 13:34:37 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'BJk34138MPLpS4pA-wnErN2AcNfdGQ0f' in 32000 ms (Method: REGISTER)
[May 16 13:34:38] NOTICE[28632]: chan_sip.c:29343 sip_poke_noanswer: Peer 'SGS4C' is now UNREACHABLE!  Last qualify: 0
Really destroying SIP dialog '7f8a5687235feab665f871c03641315c@192.168.1.2:5060' Method: OPTIONS
Really destroying SIP dialog 'RULCHM.n7qYi-BJgnESL6fQf.5xzIKlg' Method: REGISTER
---
Really destroying SIP dialog '3292a9a70efe987349384f3d465443e4@192.168.1.2:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.1.254:5060 --->
INVITE sip:32@192.168.1.2 SIP/2.0
From: "31"<sip:ATAFXS2@192.168.1.2:5060>;tag=80ade218-c0a801fe-13c4-45028-db85eb-459d7b17-db85eb
To: "32"<sip:32@192.168.1.2:5060>
Call-ID: 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85eb-5983212b-6c1ed154
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SpeedTouch 780 Build 7.4.4.7
Contact: <sip:ATAFXS2@192.168.1.254:5060>
X-Serialnumber: CP0710LT87Z
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, UPDATE
Content-Type: application/sdp
Content-Length: 215

v=0
o=780 1463410283 1463410283 IN IP4 192.168.1.254
s=-
c=IN IP4 192.168.1.254
t=0 0
m=audio 10002 RTP/AVP 8 0 4 18 97
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.1.254:5060 (no NAT)
Sending to 192.168.1.254:5060 (no NAT)
Using INVITE request as basis request - 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
Found peer 'ATAFXS2' for 'ATAFXS2' from 192.168.1.254:5060

<--- Reliably Transmitting (no NAT) to 192.168.1.254:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85eb-5983212b-6c1ed154;received=192.168.1.254
From: "31"<sip:ATAFXS2@192.168.1.2:5060>;tag=80ade218-c0a801fe-13c4-45028-db85eb-459d7b17-db85eb
To: "32"<sip:32@192.168.1.2:5060>;tag=as1823911e
Call-ID: 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
CSeq: 1 INVITE
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="MyPBX", nonce="23b6c453"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.254:5060 --->
ACK sip:32@192.168.1.2 SIP/2.0
From: "31"<sip:ATAFXS2@192.168.1.2:5060>;tag=80ade218-c0a801fe-13c4-45028-db85eb-459d7b17-db85eb
To: "32"<sip:32@192.168.1.2:5060>;tag=as1823911e
Call-ID: 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85eb-5983212b-6c1ed154
Max-Forwards: 70
User-Agent: SpeedTouch 780 Build 7.4.4.7
Contact: <sip:ATAFXS2@192.168.1.254:5060>
X-Serialnumber: CP0710LT87Z
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.254:5060 --->
INVITE sip:32@192.168.1.2 SIP/2.0
From: "31"<sip:ATAFXS2@192.168.1.2:5060>;tag=80ade218-c0a801fe-13c4-45028-db85eb-459d7b17-db85eb
To: "32"<sip:32@192.168.1.2:5060>
Call-ID: 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85eb-59832149-5a126032
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SpeedTouch 780 Build 7.4.4.7
Contact: <sip:ATAFXS2@192.168.1.254:5060>
Authorization: Digest username="ATAFXS2",realm="MyPBX",nonce="23b6c453",uri="sip:32@192.168.1.2",response="b3096a44f25a81bb205c4eeed8b26b65",algorithm=MD5
X-Serialnumber: CP0710LT87Z
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, UPDATE
Content-Type: application/sdp
Content-Length: 215

v=0
o=780 1463410283 1463410283 IN IP4 192.168.1.254
s=-
c=IN IP4 192.168.1.254
t=0 0
m=audio 10002 RTP/AVP 8 0 4 18 97
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
<------------->
--- (15 headers 10 lines) ---
Sending to 192.168.1.254:5060 (no NAT)
Using INVITE request as basis request - 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
Found peer 'ATAFXS2' for 'ATAFXS2' from 192.168.1.254:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 97
Found audio description format telephone-event for ID 97
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.254:10002
Looking for 32 in MySets (domain 192.168.1.2)
sip_route_dump: route/path hop: <sip:ATAFXS2@192.168.1.254:5060>

<--- Transmitting (no NAT) to 192.168.1.254:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85eb-59832149-5a126032;received=192.168.1.254
From: "31"<sip:ATAFXS2@192.168.1.2:5060>;tag=80ade218-c0a801fe-13c4-45028-db85eb-459d7b17-db85eb
To: "32"<sip:32@192.168.1.2:5060>
Call-ID: 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
CSeq: 2 INVITE
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:32@192.168.1.2:5060>
Content-Length: 0


<------------>
Really destroying SIP dialog '7852e28154b1b1a66c760aee63c99a51@192.168.1.2:5060' Method: INVITE
[May 16 13:34:57] WARNING[12904][C-00000019]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Scheduling destruction of SIP dialog '80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 192.168.1.254:5060 --->
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85eb-59832149-5a126032;received=192.168.1.254
From: "31"<sip:ATAFXS2@192.168.1.2:5060>;tag=80ade218-c0a801fe-13c4-45028-db85eb-459d7b17-db85eb
To: "32"<sip:32@192.168.1.2:5060>;tag=as5dbc5b7e
Call-ID: 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
CSeq: 2 INVITE
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.254:5060 --->
ACK sip:32@192.168.1.2 SIP/2.0
From: "31"<sip:ATAFXS2@192.168.1.2:5060>;tag=80ade218-c0a801fe-13c4-45028-db85eb-459d7b17-db85eb
To: "32"<sip:32@192.168.1.2:5060>;tag=as5dbc5b7e
Call-ID: 80adce68-c0a801fe-13c4-45028-db85eb-2742c0e1-db85eb
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85eb-59832149-5a126032
Max-Forwards: 70
User-Agent: SpeedTouch 780 Build 7.4.4.7
Contact: <sip:ATAFXS2@192.168.1.254:5060>
X-Serialnumber: CP0710LT87Z
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.1.214:5060:
OPTIONS sip:SGS4C@192.168.1.214:49364;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK60e00654
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as62465c23
To: <sip:SGS4C@192.168.1.214:49364;ob>
Contact: <sip:asterisk@192.168.1.2:5060>
Call-ID: 6a116d883f8feead7c7c363c7db38b2f@192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: ZON ZON Phone 1.0.0 (1)
Date: Mon, 16 May 2016 13:34:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '6a116d883f8feead7c7c363c7db38b2f@192.168.1.2:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.1.254:5060 --->
REGISTER sip:192.168.1.2:5060 SIP/2.0
From: <sip:ATAFXS2@192.168.1.2:5060>;tag=80adecf8-c0a801fe-13c4-45028-da1724-6cbd2856-da1724
To: <sip:ATAFXS2@192.168.1.2:5060>
Call-ID: 80f38e14-c0a801fe-13c4-45028-da1724-2c97c830-da1724
CSeq: 1381 REGISTER
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85ef-598330f3-3852fef0
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SpeedTouch 780 Build 7.4.4.7
Contact: <sip:ATAFXS2@192.168.1.254:5060>
Expires: 120
Authorization: Digest username="ATAFXS2",realm="MyPBX",nonce="57babb20",uri="sip:192.168.1.2:5060",response="a1e4ab676f4c6d463ace59e114415b73",algorithm=MD5
X-Serialnumber: CP0710LT87Z
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.254:5060 (no NAT)
Sending to 192.168.1.254:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.254:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85ef-598330f3-3852fef0;received=192.168.1.254
From: <sip:ATAFXS2@192.168.1.2:5060>;tag=80adecf8-c0a801fe-13c4-45028-da1724-6cbd2856-da1724
To: <sip:ATAFXS2@192.168.1.2:5060>;tag=as4f784f3a
Call-ID: 80f38e14-c0a801fe-13c4-45028-da1724-2c97c830-da1724
CSeq: 1381 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="MyPBX", nonce="68ab6f71"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '80f38e14-c0a801fe-13c4-45028-da1724-2c97c830-da1724' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.254:5060 --->
REGISTER sip:192.168.1.2:5060 SIP/2.0
From: <sip:ATAFXS1@192.168.1.2:5060>;tag=80adf9a8-c0a801fe-13c4-45028-da1724-7af20321-da1724
To: <sip:ATAFXS1@192.168.1.2:5060>
Call-ID: 80f38e14-c0a801fe-13c4-45028-da1724-7eedfc4b-da1724
CSeq: 1381 REGISTER
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85ef-598330f3-65d1cb8e
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SpeedTouch 780 Build 7.4.4.7
Contact: <sip:ATAFXS1@192.168.1.254:5060>
Expires: 120
Authorization: Digest username="ATAFXS1",realm="MyPBX",nonce="5f047f64",uri="sip:192.168.1.2:5060",response="4de7136eb3d1286e8f0d6721fe51eb33",algorithm=MD5
X-Serialnumber: CP0710LT87Z
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.254:5060 (no NAT)
Sending to 192.168.1.254:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.254:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85ef-598330f3-65d1cb8e;received=192.168.1.254
From: <sip:ATAFXS1@192.168.1.2:5060>;tag=80adf9a8-c0a801fe-13c4-45028-da1724-7af20321-da1724
To: <sip:ATAFXS1@192.168.1.2:5060>;tag=as3418997c
Call-ID: 80f38e14-c0a801fe-13c4-45028-da1724-7eedfc4b-da1724
CSeq: 1381 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="MyPBX", nonce="66ae08a2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '80f38e14-c0a801fe-13c4-45028-da1724-7eedfc4b-da1724' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.254:5060 --->
REGISTER sip:192.168.1.2:5060 SIP/2.0
From: <sip:ATAFXS2@192.168.1.2:5060>;tag=80adecf8-c0a801fe-13c4-45028-da1724-6cbd2856-da1724
To: <sip:ATAFXS2@192.168.1.2:5060>
Call-ID: 80f38e14-c0a801fe-13c4-45028-da1724-2c97c830-da1724
CSeq: 1382 REGISTER
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85ef-598330fd-273c40c
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SpeedTouch 780 Build 7.4.4.7
Contact: <sip:ATAFXS2@192.168.1.254:5060>
Expires: 120
Authorization: Digest username="ATAFXS2",realm="MyPBX",nonce="68ab6f71",uri="sip:192.168.1.2:5060",response="c3fba40ddc9906623e710f328d9d880d",algorithm=MD5
X-Serialnumber: CP0710LT87Z
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.254:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.254:5060:
OPTIONS sip:ATAFXS2@192.168.1.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3fa36399
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as58fbd625
To: <sip:ATAFXS2@192.168.1.254:5060>
Contact: <sip:asterisk@192.168.1.2:5060>
Call-ID: 697297f158837e51610dc8033f494710@192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: ZON ZON Phone 1.0.0 (1)
Date: Mon, 16 May 2016 13:35:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.1.254:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85ef-598330fd-273c40c;received=192.168.1.254
From: <sip:ATAFXS2@192.168.1.2:5060>;tag=80adecf8-c0a801fe-13c4-45028-da1724-6cbd2856-da1724
To: <sip:ATAFXS2@192.168.1.2:5060>;tag=as4f784f3a
Call-ID: 80f38e14-c0a801fe-13c4-45028-da1724-2c97c830-da1724
CSeq: 1382 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: <sip:ATAFXS2@192.168.1.254:5060>;expires=120
Date: Mon, 16 May 2016 13:35:01 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '80f38e14-c0a801fe-13c4-45028-da1724-2c97c830-da1724' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.254:5060 --->
REGISTER sip:192.168.1.2:5060 SIP/2.0
From: <sip:ATAFXS1@192.168.1.2:5060>;tag=80adf9a8-c0a801fe-13c4-45028-da1724-7af20321-da1724
To: <sip:ATAFXS1@192.168.1.2:5060>
Call-ID: 80f38e14-c0a801fe-13c4-45028-da1724-7eedfc4b-da1724
CSeq: 1382 REGISTER
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85ef-5983311b-3a6fc66a
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SpeedTouch 780 Build 7.4.4.7
Contact: <sip:ATAFXS1@192.168.1.254:5060>
Expires: 120
Authorization: Digest username="ATAFXS1",realm="MyPBX",nonce="66ae08a2",uri="sip:192.168.1.2:5060",response="2f39abd41160b926b4d352f55b2b7a90",algorithm=MD5
X-Serialnumber: CP0710LT87Z
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.254:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.254:5060:
OPTIONS sip:ATAFXS1@192.168.1.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK60251e21
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as4bd2ac34
To: <sip:ATAFXS1@192.168.1.254:5060>
Contact: <sip:asterisk@192.168.1.2:5060>
Call-ID: 61310fdb1eb59bec120b77ab52b76e4a@192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: ZON ZON Phone 1.0.0 (1)
Date: Mon, 16 May 2016 13:35:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.1.254:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK-db85ef-5983311b-3a6fc66a;received=192.168.1.254
From: <sip:ATAFXS1@192.168.1.2:5060>;tag=80adf9a8-c0a801fe-13c4-45028-da1724-7af20321-da1724
To: <sip:ATAFXS1@192.168.1.2:5060>;tag=as3418997c
Call-ID: 80f38e14-c0a801fe-13c4-45028-da1724-7eedfc4b-da1724
CSeq: 1382 REGISTER
Server: ZON ZON Phone 1.0.0 (1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: <sip:ATAFXS1@192.168.1.254:5060>;expires=120
Date: Mon, 16 May 2016 13:35:01 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '80f38e14-c0a801fe-13c4-45028-da1724-7eedfc4b-da1724' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.254:5060 --->
SIP/2.0 200 OK
From: "asterisk"<sip:asterisk@192.168.1.2>;tag=as58fbd625
To: <sip:ATAFXS2@192.168.1.254:5060>;tag=80adf7d8-c0a801fe-13c4-45028-db85ef-698c90a8-db85ef
Call-ID: 697297f158837e51610dc8033f494710@192.168.1.2:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3fa36399
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, UPDATE
User-Agent: SpeedTouch 780 Build 7.4.4.7
X-Serialnumber: CP0710LT87Z
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '697297f158837e51610dc8033f494710@192.168.1.2:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.1.254:5060 --->
SIP/2.0 200 OK
From: "asterisk"<sip:asterisk@192.168.1.2>;tag=as4bd2ac34
To: <sip:ATAFXS1@192.168.1.254:5060>;tag=80adf098-c0a801fe-13c4-45028-db85f0-343ec0c6-db85f0
Call-ID: 61310fdb1eb59bec120b77ab52b76e4a@192.168.1.2:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK60251e21
Supported: replaces,100rel
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, UPDATE
User-Agent: SpeedTouch 780 Build 7.4.4.7
X-Serialnumber: CP0710LT87Z
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '61310fdb1eb59bec120b77ab52b76e4a@192.168.1.2:5060' Method: OPTIONS
CLI> exit

Regards,
jss

jcolp · May 16, 2016, 1:50pm

What is the dialplan? As well if you type “core set verbose 9” and try again do you get additional output, and if so can you provide it.

jcolp · May 16, 2016, 1:52pm

Actually upon looking at your version and stuff that has changed there was a major fix done for an issue which would look like this - where a device would go unreachable when it shouldn’t. I’d suggest upgrading as well before looking further. The latest is 13.9.1.

jssilva · May 16, 2016, 2:13pm

I guess I’ll have to wait then, because the latest version on openwrt repository is 13.6.0-1. I’ll be watching for an upgrade.

Thank you so, so much for your help. I wished there were a lot more people like you to make this a better world and erradicate selfishness.

Best regards,
jss

jssilva · May 19, 2016, 10:35pm

Just for your information, achieved to upgrade to 13.8.0 but problem remains.

EDIT: Just found a way to force the extension to listen on 5060 and it now works, obviously.
Anyway, problem remains if one is not able to do this forcing.

Topic		Replies	Views
Extension(Device) Port Asterisk Support	4	256	November 18, 2011
SIP pones registers on random port and not on 5060 Asterisk	6	1909	December 25, 2016
Remote extension will not register on port 5070 Asterisk Support	1	288	May 2, 2007
Clients not registering Asterisk Support	7	1103	December 29, 2008
Problem: Bindport and register port Asterisk Support	5	526	November 12, 2008

Asterisk does not use registered port for invitation since upgrade

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