thr3ads.net - asterisk users - [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI [Dec 2020]

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Turritopsis Dohrnii Teo En Ming

2020-Dec-24 03:09 UTC

[asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Thank you for your replies, Duncan Turnbull.

I am going to run tcpdump on my Asterisk PBX server.

By the way, I found a Youtube video.

Youtube video: Cisco 7942g IP Phone Configuration on FreePBX 
In-Depth(Without Endpoint Manager)

Link: https://www.youtube.com/watch?v=gk6w8O3fZlc&feature=youtu.be

 From the above youtube video, it seems that I cannot use pjsip extension 
for my Cisco 7960 IP phone. I need to delete the pjsip extension, and 
then create a legacy chan_sip extension, it seems.

These are the notes I have taken after watching the above Youtube video:

1. Cannot use pjsip extension, need to use legacy chan_sip extension

2. Display name: Your name

3. secret is 8 char only, must be numeric

4. Voicemail: Enabled

5. Require from same extension: yes

6. Go to Advanced, nat mode: never

7. Port 5060

8. Qualify: No

9. Send RPID: Send Remote-Party-ID header

10. Go to Settings > Asterisk SIP Settings > SIP Legacy Settings 
(chan_sip)

11. NAT: No

12. Enable SRV Lookup: No

13. Edit SEP<mac address>.cnf.xml, sipPort: 5160

14. Line #1, port: 5160



On 2020-12-23 17:55, Duncan Turnbull wrote:> 
> 
> Sent from my iPad
> 
>> On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming 
>> <ceo at teo-en-ming.com> wrote:
>> 
>> Hi Duncan Turnbull,
>> 
>> You can watch my Youtube video of my Cisco 7960 IP phone.
>> 
>> The link is: https://www.youtube.com/watch?v=ip_F08jmmio
>> 
>> My Youtube video shows the Network Configuration settings, SIP 
>> Configuration settings and Status of my Cisco 7960 IP Phone.
> The phone looks like it has picked up the configs however in the
> status there are two error messages re parsing SipDefault.cnf and the
> specific SIP..MAC.. file - you should try and remedy those errors .
> Otherwise most of the settings look to be there
> 
> I would suggest cutting out as much of the config as you can
> 
> I would also suggest you run tcpdump on the 192.168.1.9 box and
> monitor any traffic at all coming from your phone which is now on
> 192.168.1.130.  You may see the SIP messages there
> 
> Cheers Duncan
> 
>> Did you see anything wrong?
>> 
>> 
>> 
>> 
>>> On 2020-12-23 12:38, Duncan Turnbull wrote:
>>> Hi there
>>>>> On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming
>>>>> <ceo at teo-en-ming.com> wrote:
>>>> Good morning Duncan Turnbull,
>>>> I have posted my Asterisk PBX server debugging output
previously in
>>>> my original post. The link is:
>>>>
http://lists.digium.com/pipermail/asterisk-users/2020-December/295555.html
>>>> I saw many REGISTER requests. Are these REGISTER requests from
my
>>>> Cisco 7960 IP phone? Could you help me to check? Thank you very
>>>> much.
>>> If they come from the phone they will have the phones ip address.
The
>>> phone will also try and register with the extension you have given 
>>> it.
>>> None of the registration messages appear to have the up or the
>>> extension so you will need to figure out what’s gone wrong with the
>>> phones config
>>> That’s why checking the phone settings to see fit they have changed
>>> helps understand if your configs were correct. You can do this via 
>>> the
>>> phone screen or telnet. It will take you some time to become
familiar
>>> with this but it’s worth it
>>> Good luck
>>>> I shall reproduce my Asterisk PBX server debugging output
below.
>>>> SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
>>>> ============================================>>>> #
asterisk -vvvr
>>>> sip set debug on
>>>> freepbx*CLI>
>>>> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893
sip_reregister:
>>>> -- Re-registration for  60751 at sip.sg.didlogic.net
>>>> REGISTER 12 headers, 0 lines
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
>>>> Max-Forwards: 70
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 165 REGISTER
>>>> Supported: replaces, timer
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>> nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
>>>> response="bfadacdd4e745fd4b9e12046e6ce2afc",
qop=auth,
>>>> cnonce="2b1b6d13", nc=00000003
>>>> Expires: 120
>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To:
>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 165 REGISTER
>>>> Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
>>>> IP>:26462>;expires=120;received="sip:<CORPORATE
OFFICE PUBLIC
>>>> IP>:26462"
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (8 headers 0 lines) ---
>>>> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961
>>>> handle_response_register: Outbound Registration: Expiry for
>>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration in
105 s)
>>>> Really destroying SIP dialog
>>>> '005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1' Method:
REGISTER
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport
>>>> Max-Forwards: 70
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as41ddf4a6
>>>> To: <sip:sip.sg.didlogic.net>
>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606 at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Date: Sun, 20 Dec 2020 07:07:07 GMT
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>> INFO, PUBLISH, MESSAGE
>>>> Supported: replaces, timer
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK51105854;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as41ddf4a6
>>>> To: 
>>>>
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924
>>>> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606 at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (7 headers 0 lines) ---
>>>> Really destroying SIP dialog
>>>> '0b0605df4f4ca7b03402c9fd5a869606 at 192.168.1.9:5160'
Method: OPTIONS
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK4ff08179;rport
>>>> Max-Forwards: 70
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as004073f3
>>>> To: <sip:sip.sg.didlogic.net>
>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>> Call-ID: 500c8eb32071e3fc462be80f243d38fc at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Date: Sun, 20 Dec 2020 07:08:07 GMT
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>> INFO, PUBLISH, MESSAGE
>>>> Supported: replaces, timer
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK4ff08179;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as004073f3
>>>> To: 
>>>>
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.385d
>>>> Call-ID: 500c8eb32071e3fc462be80f243d38fc at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (7 headers 0 lines) ---
>>>> Really destroying SIP dialog
>>>> '500c8eb32071e3fc462be80f243d38fc at 192.168.1.9:5160'
Method: OPTIONS
>>>> [2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:15893
sip_reregister:
>>>> -- Re-registration for  60751 at sip.sg.didlogic.net
>>>> REGISTER 12 headers, 0 lines
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK6d85e46f;rport
>>>> Max-Forwards: 70
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 166 REGISTER
>>>> Supported: replaces, timer
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>> nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
>>>> response="074f1f037639144de751dc9231c191c9",
qop=auth,
>>>> cnonce="6eb58a86", nc=00000004
>>>> Expires: 120
>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 401 Unauthorized
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK6d85e46f;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To:
>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e426
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 166 REGISTER
>>>> WWW-Authenticate: Digest realm="sip.sg.didlogic.net",
>>>> nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
qop="auth"
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (8 headers 0 lines) ---
>>>> Responding to challenge, registration to domain/host name
>>>> sip.sg.didlogic.net
>>>> REGISTER 12 headers, 0 lines
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK7f0ddbdc;rport
>>>> Max-Forwards: 70
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 167 REGISTER
>>>> Supported: replaces, timer
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>> nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
>>>> response="c1184dab1dd50dad14cba70933b6bbaa",
qop=auth,
>>>> cnonce="4945f552", nc=00000001
>>>> Expires: 120
>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK7f0ddbdc;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To:
>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f557
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 167 REGISTER
>>>> Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
>>>> IP>:26462>;expires=120;received="sip:<CORPORATE
OFFICE PUBLIC
>>>> IP>:26462"
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (8 headers 0 lines) ---
>>>> [2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:24961
>>>> handle_response_register: Outbound Registration: Expiry for
>>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration in
105 s)
>>>> Really destroying SIP dialog
>>>> '005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1' Method:
REGISTER
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK0f99cec8;rport
>>>> Max-Forwards: 70
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as786e56f3
>>>> To: <sip:sip.sg.didlogic.net>
>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>> Call-ID: 49595bb227bd6e390ca40ce6508308aa at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Date: Sun, 20 Dec 2020 07:09:07 GMT
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>> INFO, PUBLISH, MESSAGE
>>>> Supported: replaces, timer
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK0f99cec8;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as786e56f3
>>>> To: 
>>>>
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e4fb
>>>> Call-ID: 49595bb227bd6e390ca40ce6508308aa at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (7 headers 0 lines) ---
>>>> Really destroying SIP dialog
>>>> '49595bb227bd6e390ca40ce6508308aa at 192.168.1.9:5160'
Method: OPTIONS
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> [2020-12-20 07:09:52] NOTICE[2366]: chan_sip.c:15893
sip_reregister:
>>>> -- Re-registration for  60751 at sip.sg.didlogic.net
>>>> REGISTER 12 headers, 0 lines
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK030dc571;rport
>>>> Max-Forwards: 70
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 168 REGISTER
>>>> Supported: replaces, timer
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>> nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
>>>> response="b6e96ab578798296e812139c383ebbac",
qop=auth,
>>>> cnonce="6313e774", nc=00000002
>>>> Expires: 120
>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK030dc571;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To:
>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2fbb
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 168 REGISTER
>>>> Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
>>>> IP>:26462>;expires=120;received="sip:<CORPORATE
OFFICE PUBLIC
>>>> IP>:26462"
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (8 headers 0 lines) ---
>>>> [2020-12-20 07:09:52] NOTICE[2366]: chan_sip.c:24961
>>>> handle_response_register: Outbound Registration: Expiry for
>>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration in
105 s)
>>>> Really destroying SIP dialog
>>>> '005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1' Method:
REGISTER
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK25d81863;rport
>>>> Max-Forwards: 70
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as3a8d9861
>>>> To: <sip:sip.sg.didlogic.net>
>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>> Call-ID: 4f9667021511db2968a0dc9964673ac9 at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Date: Sun, 20 Dec 2020 07:10:07 GMT
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>> INFO, PUBLISH, MESSAGE
>>>> Supported: replaces, timer
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK25d81863;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as3a8d9861
>>>> To: 
>>>>
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.a139
>>>> Call-ID: 4f9667021511db2968a0dc9964673ac9 at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (7 headers 0 lines) ---
>>>> Really destroying SIP dialog
>>>> '4f9667021511db2968a0dc9964673ac9 at 192.168.1.9:5160'
Method: OPTIONS
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK443d3196;rport
>>>> Max-Forwards: 70
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as5d241373
>>>> To: <sip:sip.sg.didlogic.net>
>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>> Call-ID: 201fab2b13e08922618f34911ce37ce0 at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Date: Sun, 20 Dec 2020 07:11:07 GMT
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>> INFO, PUBLISH, MESSAGE
>>>> Supported: replaces, timer
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK443d3196;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as5d241373
>>>> To: 
>>>>
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.19e3
>>>> Call-ID: 201fab2b13e08922618f34911ce37ce0 at 192.168.1.9:5160
>>>> CSeq: 102 OPTIONS
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (7 headers 0 lines) ---
>>>> Really destroying SIP dialog
>>>> '201fab2b13e08922618f34911ce37ce0 at 192.168.1.9:5160'
Method: OPTIONS
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> [2020-12-20 07:11:37] NOTICE[2366]: chan_sip.c:15893
sip_reregister:
>>>> -- Re-registration for  60751 at sip.sg.didlogic.net
>>>> REGISTER 12 headers, 0 lines
>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK292b942b;rport
>>>> Max-Forwards: 70
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 169 REGISTER
>>>> Supported: replaces, timer
>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>> nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
>>>> response="8305fa5eb2ec2396b7b618f40923e597",
qop=auth,
>>>> cnonce="043fc0dd", nc=00000003
>>>> Expires: 120
>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>> Content-Length: 0
>>>> ---
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>>
192.168.1.9:5160;branch=z9hG4bK292b942b;rport=26462;received=<CORPORATE
>>>> OFFICE PUBLIC IP>
>>>> From: <sip:60751 at sip.sg.didlogic.net>;tag=as6df6d977
>>>> To:
>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6c72
>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>> CSeq: 169 REGISTER
>>>> Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
>>>> IP>:26462>;expires=120;received="sip:<CORPORATE
OFFICE PUBLIC
>>>> IP>:26462"
>>>> Content-Length: 0
>>>> <------------->
>>>> --- (8 headers 0 lines) ---
>>>> [2020-12-20 07:11:37] NOTICE[2366]: chan_sip.c:24961
>>>> handle_response_register: Outbound Registration: Expiry for
>>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration in
105 s)
>>>> Really destroying SIP dialog
>>>> '005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1' Method:
REGISTER
>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>> <------------->
>>>> freepbx*CLI>
>>>>> On 2020-12-23 04:29, Duncan Turnbull wrote:
>>>>> Hi there
>>>>> That answer includes using tcpdump to check for SIP packets
and
>>>>> examine the register packet. At this point you have no SIP
packets
>>>>> coming from your phone so you are not upto that stage yet.
>>>>> You need to know why there are no SIP packets coming. My
guess is
>>>>> your
>>>>> config files have a typo in them. You can modify the ones I
sent to
>>>>> see if they work for you. You do need to validate your
configs. I
>>>>> recommend telnet to the phone, and checking the display
settings to
>>>>> see if it has picked up the settings. Equally check the
phone logs.
>>>>> It
>>>>> will tell you which files have errors.
>>>>> Until you get the settings loaded correctly nothing else
will
>>>>> matter
>>>>> Enjoy your break. If you want to use Voip you should
definitely
>>>>> spend
>>>>> some time to learn tcpdump. If you want to use Cisco you
need to be
>>>>> able to understand the configs yourself and get as much
info from
>>>>> the
>>>>> phone as possible. Not hard but it takes a little bit of
time.
>>>>> Cheers Duncan
>>>>>> On Tue, Dec 22, 2020 at 10:43 PM Turritopsis Dohrnii
Teo En Ming
>>>>>> <ceo at teo-en-ming.com> wrote:
>>>>>> Good day from Singapore,
>>>>>> I seem to have found the solution at FreePBX community
forums.
>>>>>> Please
>>>>>> check out the following discussion thread.
>>>>>> Discussion Thread: Cisco 7940 registration problem
RESOLVED
>>>>>> Link:
>>>>>
https://community.freepbx.org/t/cisco-7940-registration-problem-resolved/30285
>>>>>> But I don't understand very well what users at this
discussion
>>>>>> thread
>>>>>> are talking about. Can someone help me understand
better after
>>>>>> reading
>>>>>> through the above discussion thread?
>>>>>> For your information, I am using PJSIP extension
instead of
>>>>>> CHAN_SIP
>>>>>> extension.
>>>>>> I am planning to work on my Cisco 7960 IP phone
registration
>>>>>> problem
>>>>>> this coming Christmas 2020 weekends.
>>>>>> Thank you very much for your kind assistance.
>>>>>> On 2020-12-21 09:58, Duncan Turnbull wrote:
>>>>>>> Hi there
>>>>>>> I would normally highlight the part but the email
is so long I
>>>>>> thought
>>>>>>> I would just note what I can see
>>>>>>> It appears the Cisco is downloading files.
>>>>>>> None of the SIP traces show the IP of the phone of
the extension
>>>>>>> Your proxy is at 192.168.1.9
>>>>>>> Your phone is at 192.168.1.130
>>>>>>> These are the details you want the phone to pickup
>>>>>>> line1_name: "1600"
>>>>>>> line1_shortname: "TEO EN MING"
>>>>>>> line1_displayname: "TURRITOPSIS DOHRNII TEO EN
MING"
>>>>>>> line1_authname: "1600"
>>>>>>> line1_password: "IP Phone Extension
Password"
>>>>>>> I don't see any registration attempts from your
phone.
>>>>>>> The first thing is to use the phone screen display
to check if it
>>>>>>> actually has picked up the settings.
>>>>>>> To unlock the Cisco SIP IP phone, press **#
>>>>>>> You can also telnet to the phone usually cisco as
password, and
>>>>>> look
>>>>>>> at logs. Its quite possible some of your config
files are not
>>>>>> quite
>>>>>>> right. If they were all wrong the Cisco would keep
trying to TFTP
>>>>>> the
>>>>>>> files.
>>>>>>> This is an old SIPDefault.cnf I used to use in NZ
>>>>>>>
===============================================>>>>>>> ;
sip default configuration file
>>>>>>> #Image Version
>>>>>>> image_version:P0S3-08-6-00 ;
>>>>>>> #Proxy server address
>>>>>>> proxy1_address: 10.12.41.1 ;
>>>>>>> proxy_register: 1;
>>>>>>> logo_url: "http://10.12.41.1/Logo.bmp"   
; URL
>>>>>> for
>>>>>>> branding logo to be used on phone display
>>>>>>> time_format: 0 ;
>>>>>>> preferred_codec: g711alaw ;
>>>>>>> sntp_mode: unicast ;
>>>>>>> dial_template: dialplan
>>>>>>> sntp_server: 10.12.41.1 ;
>>>>>>> messages_uri: "*97"
>>>>>>> time_zone : NZST
>>>>>>> dst_auto_adjust : 1
>>>>>>> dst_offset : 01
>>>>>>> dst_start_month : September
>>>>>>> dst_start_day : 29
>>>>>>> dst_start_time : 02:00
>>>>>>> dst_stop_month : April
>>>>>>> dst_stop_day : 6
>>>>>>> dst_stop_time : 02:00
>>>>>>>
================================================>>>>>>>
This is a SIP Phone template - you can compare settings and notes
>>>>>> on
>>>>>>> the settings
>>>>>>> # SIP Configuration Generic File (start)
>>>>>>>
================================================>>>>>>> #
Proxy Server
>>>>>>> proxy1_address: "10.12.41.1"
>>>>>>> # Line 1 Settings
>>>>>>> line1_name: "EXTN"                     ;
Line 1 Extension\User ID
>>>>>>> line1_shortname: "0NXXXXXXX"           ;
Line 1 Short Name
>>>>>>> line1_displayname: "0NXXXXXXX"          
; Line 1 Display Name
>>>>>>> line1_authname: "EXTN"         ; Line 1
Registration
>>>>>> Authentication
>>>>>>> line1_password: "6gs72ha9"         ; Line
1 Registration Password
>>>>>>> phone_label: "Company Limited"    ; no
effect on SIP messaging
>>>>>>> # Line 2 Settings
>>>>>>> line2_name: ""                          ;
Line 2 Extension\User
>>>>>>> ID
>>>>>>> line2_displayname: ""                   ;
Line 2 Display Name
>>>>>>> line2_authname: "UNPROVISIONED"         ;
Line 2 Registration
>>>>>>> Authentication
>>>>>>> line2_password: "UNPROVISIONED"         ;
Line 2 Registration
>>>>>> Password
>>>>>>> # Line 3 Settings
>>>>>>> line3_name: ""                          ;
Line 3 Extension\User
>>>>>>> ID
>>>>>>> line3_displayname: ""                   ;
Line 3 Display Name
>>>>>>> line3_authname: "UNPROVISIONED"         ;
Line 3 Registration
>>>>>>> Authentication
>>>>>>> line3_password: "UNPROVISIONED"         ;
Line 3 Registration
>>>>>> Password
>>>>>>> # Line 4 Settings
>>>>>>> line4_name: ""                          ;
Line 4 Extension\User
>>>>>>> ID
>>>>>>> line4_displayname: ""                   ;
Line 4 Display Name
>>>>>>> line4_authname: "UNPROVISIONED"         ;
Line 4 Registration
>>>>>>> Authentication
>>>>>>> line4_password: "UNPROVISIONED"         ;
Line 4 Registration
>>>>>> Password
>>>>>>> # Line 5 Settings
>>>>>>> line5_name: ""                          ;
Line 5 Extension\User
>>>>>>> ID
>>>>>>> line5_displayname: ""                   ;
Line 5 Display Name
>>>>>>> line5_authname: "UNPROVISIONED"         ;
Line 5 Registration
>>>>>>> Authentication
>>>>>>> line5_password: "UNPROVISIONED"         ;
Line 5 Registration
>>>>>> Password
>>>>>>> # Line 6 Settings
>>>>>>> line6_name: ""                          ;
Line 6 Extension\User
>>>>>>> ID
>>>>>>> line6_displayname: ""                   ;
Line 6 Display Name
>>>>>>> line6_authname: "UNPROVISIONE"         ;
Line 6 Registration
>>>>>>> Authentication
>>>>>>> line6_password: "UNPROVISIONE"         ;
Line 6 Registration
>>>>>> Password
>>>>>>> # Emergency Proxy info
>>>>>>> proxy_emergency: ""
>>>>>>> proxy_emergency_port: "5060"
>>>>>>> # Backup Proxy info
>>>>>>> proxy_backup: ""
>>>>>>> proxy_backup_port: "5060"
>>>>>>> # Outbound Proxy info
>>>>>>> outbound_proxy: ""
>>>>>>> outbound_proxy_port: "5060"
>>>>>>> # NAT/Firewall Traversal
>>>>>>> nat_enable: "0"
>>>>>>> nat_address: ""
>>>>>>> voip_control_port: "5060"
>>>>>>> start_media_port: "10000"
>>>>>>> end_media_port:  "20000"
>>>>>>> nat_received_processing: "0"
>>>>>>> # Phone Label (Text desired to be displayed in
upper right
>>>>>>> corner)
>>>>>>> phone_label: "Name's phone"          
; Has no effect on SIP
>>>>>> messaging
>>>>>>> # Time Zone phone will reside in
>>>>>>> time_zone: NZST
>>>>>>> time_format: "D/M/Ya"
>>>>>>> # Telnet Level (enable or disable the ability to
telnet into this
>>>>>> phone
>>>>>>> telnet_level: "2"      ; 0-Disabled
(default), 1-Enabled,
>>>>>> 2-Privileged
>>>>>>> # Phone prompt/password for telnet/console session
>>>>>>> phone_prompt: ""                         
; Telnet/Console
>>>>>> Prompt
>>>>>>> phone_password: "cisco"                  
; Telnet/Console
>>>>>>> Password
>>>>>>> # Enable_VAD (1-enabled, 0-disabled)
>>>>>>> enable_vad: "0"
>>>>>>> # Network Media Type (auto, full100, full10,
half100, half10)
>>>>>>> network_media_type: "auto"
>>>>>>> user_info: none
>>>>>>> # URL for external Directory location
>>>>>>> directory_url:
"http://10.12.41.1/directory.html"
>>>>>>>
================================================>>>>>>> I
would then recommend tcpdump to monitor traffic coming from
>>>>>>> 192.168.1.130  Tcpdump is an important tool to
learn to use and
>>>>>> you
>>>>>>> can look at all traffic coming from the phone,
perhaps its using
>>>>>> TCP
>>>>>>> instead of UDP?
>>>>>>> Have you got a copy of the Cisco SIP IP Phone 7960
Administrator
>>>>>> Guide
>>>>>>> - it should be on the web somewhere
>>>>>>> Good luck, it will take a little time to get
familiar with your
>>>>>>> environment but its important to put the time in to
work out what
>>>>>>> means what
>>>>>>> Cheers Duncan
>>>>>> --
>>>>>
_____________________________________________________________________
>>>>>> -- Bandwidth and Colocation Provided by
http://www.api-digital.com
>>>>>> --
>>>>>> Check out the new Asterisk community forum at:
>>>>>> https://community.asterisk.org/
>>>>>> 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
>> 
>>>> --
>>>>
_____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by
http://www.api-digital.com
>>>> --
>>>> Check out the new Asterisk community forum at: 
>>>> https://community.asterisk.org/
>>>> 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
>> 
>> 
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> 
>> Check out the new Asterisk community forum at: 
>> https://community.asterisk.org/
>> 
>> 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
-- 
-----BEGIN EMAIL SIGNATURE-----

The Gospel for all Targeted Individuals (TIs):

[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers

Link: 
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html

********************************************************************************************

Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's 
Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the 
United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 
2019) and Australia (25 Dec 2019 to 9 Jan 2020):

[1] https://tdtemcerts.wordpress.com/

[2] https://tdtemcerts.blogspot.sg/

[3] https://www.scribd.com/user/270125049/Teo-En-Ming

-----END EMAIL SIGNATURE-----

Duncan Turnbull

2020-Dec-24 04:33 UTC

head link

[asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi Turritopsis

I think the key point maybe making sure the password doesn’t exceed the capacity
of the phone. So an 8 char password is a good idea

I would be surprised if pjsip doesn’t work but I haven’t tried it with a Cisco
phone

Whatever gets you working is what you want

Have a wonderful Xmas 

Cheers Duncan
> On 24/12/2020, at 1:11 PM, Turritopsis Dohrnii Teo En Ming <ceo at
teo-en-ming.com> wrote:
> 
> Thank you for your replies, Duncan Turnbull.
> 
> I am going to run tcpdump on my Asterisk PBX server.
> 
> By the way, I found a Youtube video.
> 
> Youtube video: Cisco 7942g IP Phone Configuration on FreePBX
In-Depth(Without Endpoint Manager)
> 
> Link: https://www.youtube.com/watch?v=gk6w8O3fZlc&feature=youtu.be
> 
> From the above youtube video, it seems that I cannot use pjsip extension
for my Cisco 7960 IP phone. I need to delete the pjsip extension, and then
create a legacy chan_sip extension, it seems.
> 
> These are the notes I have taken after watching the above Youtube video:
> 
> 1. Cannot use pjsip extension, need to use legacy chan_sip extension
> 
> 2. Display name: Your name
> 
> 3. secret is 8 char only, must be numeric
> 
> 4. Voicemail: Enabled
> 
> 5. Require from same extension: yes
> 
> 6. Go to Advanced, nat mode: never
> 
> 7. Port 5060
> 
> 8. Qualify: No
> 
> 9. Send RPID: Send Remote-Party-ID header
> 
> 10. Go to Settings > Asterisk SIP Settings > SIP Legacy Settings
(chan_sip)
> 
> 11. NAT: No
> 
> 12. Enable SRV Lookup: No
> 
> 13. Edit SEP<mac address>.cnf.xml, sipPort: 5160
> 
> 14. Line #1, port: 5160
> 
> 
> 
>> On 2020-12-23 17:55, Duncan Turnbull wrote:
>> 
>> Sent from my iPad
>>>> On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming
<ceo at teo-en-ming.com> wrote:
>>> Hi Duncan Turnbull,
>>> You can watch my Youtube video of my Cisco 7960 IP phone.
>>> The link is: https://www.youtube.com/watch?v=ip_F08jmmio
>>> My Youtube video shows the Network Configuration settings, SIP
Configuration settings and Status of my Cisco 7960 IP Phone.
>> The phone looks like it has picked up the configs however in the
>> status there are two error messages re parsing SipDefault.cnf and the
>> specific SIP..MAC.. file - you should try and remedy those errors .
>> Otherwise most of the settings look to be there
>> I would suggest cutting out as much of the config as you can
>> I would also suggest you run tcpdump on the 192.168.1.9 box and
>> monitor any traffic at all coming from your phone which is now on
>> 192.168.1.130.  You may see the SIP messages there
>> Cheers Duncan
>>> Did you see anything wrong?
>>>> On 2020-12-23 12:38, Duncan Turnbull wrote:
>>>> Hi there
>>>>>> On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En
Ming <ceo at teo-en-ming.com> wrote:
>>>>> Good morning Duncan Turnbull,
>>>>> I have posted my Asterisk PBX server debugging output
previously in my original post. The link is:
>>>>>
http://lists.digium.com/pipermail/asterisk-users/2020-December/295555.html
>>>>> I saw many REGISTER requests. Are these REGISTER requests
from my Cisco 7960 IP phone? Could you help me to check? Thank you very much.
>>>> If they come from the phone they will have the phones ip
address. The
>>>> phone will also try and register with the extension you have
given it.
>>>> None of the registration messages appear to have the up or the
>>>> extension so you will need to figure out what’s gone wrong with
the
>>>> phones config
>>>> That’s why checking the phone settings to see fit they have
changed
>>>> helps understand if your configs were correct. You can do this
via the
>>>> phone screen or telnet. It will take you some time to become
familiar
>>>> with this but it’s worth it
>>>> Good luck
>>>>> I shall reproduce my Asterisk PBX server debugging output
below.
>>>>> SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
>>>>>
============================================>>>>> # asterisk
-vvvr
>>>>> sip set debug on
>>>>> freepbx*CLI>
>>>>> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893
sip_reregister:
>>>>> -- Re-registration for  60751 at sip.sg.didlogic.net
>>>>> REGISTER 12 headers, 0 lines
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
>>>>> Max-Forwards: 70
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 165 REGISTER
>>>>> Supported: replaces, timer
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>>> nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
>>>>> response="bfadacdd4e745fd4b9e12046e6ce2afc",
qop=auth,
>>>>> cnonce="2b1b6d13", nc=00000003
>>>>> Expires: 120
>>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To:
>>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 165 REGISTER
>>>>> Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
>>>>>
IP>:26462>;expires=120;received="sip:<CORPORATE OFFICE PUBLIC
IP>:26462"
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (8 headers 0 lines) ---
>>>>> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961
>>>>> handle_response_register: Outbound Registration: Expiry for
>>>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration
in 105 s)
>>>>> Really destroying SIP dialog
>>>>> '005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1'
Method: REGISTER
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK51105854;rport
>>>>> Max-Forwards: 70
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as41ddf4a6
>>>>> To: <sip:sip.sg.didlogic.net>
>>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>>> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606 at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Date: Sun, 20 Dec 2020 07:07:07 GMT
>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>>> INFO, PUBLISH, MESSAGE
>>>>> Supported: replaces, timer
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK51105854;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as41ddf4a6
>>>>> To:
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924
>>>>> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606 at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (7 headers 0 lines) ---
>>>>> Really destroying SIP dialog
>>>>> '0b0605df4f4ca7b03402c9fd5a869606 at
192.168.1.9:5160' Method: OPTIONS
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK4ff08179;rport
>>>>> Max-Forwards: 70
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as004073f3
>>>>> To: <sip:sip.sg.didlogic.net>
>>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>>> Call-ID: 500c8eb32071e3fc462be80f243d38fc at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Date: Sun, 20 Dec 2020 07:08:07 GMT
>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>>> INFO, PUBLISH, MESSAGE
>>>>> Supported: replaces, timer
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK4ff08179;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as004073f3
>>>>> To:
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.385d
>>>>> Call-ID: 500c8eb32071e3fc462be80f243d38fc at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (7 headers 0 lines) ---
>>>>> Really destroying SIP dialog
>>>>> '500c8eb32071e3fc462be80f243d38fc at
192.168.1.9:5160' Method: OPTIONS
>>>>> [2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:15893
sip_reregister:
>>>>> -- Re-registration for  60751 at sip.sg.didlogic.net
>>>>> REGISTER 12 headers, 0 lines
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK6d85e46f;rport
>>>>> Max-Forwards: 70
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 166 REGISTER
>>>>> Supported: replaces, timer
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>>> nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
>>>>> response="074f1f037639144de751dc9231c191c9",
qop=auth,
>>>>> cnonce="6eb58a86", nc=00000004
>>>>> Expires: 120
>>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 401 Unauthorized
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK6d85e46f;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To:
>>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e426
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 166 REGISTER
>>>>> WWW-Authenticate: Digest
realm="sip.sg.didlogic.net",
>>>>> nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
qop="auth"
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (8 headers 0 lines) ---
>>>>> Responding to challenge, registration to domain/host name
>>>>> sip.sg.didlogic.net
>>>>> REGISTER 12 headers, 0 lines
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK7f0ddbdc;rport
>>>>> Max-Forwards: 70
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 167 REGISTER
>>>>> Supported: replaces, timer
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>>> nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
>>>>> response="c1184dab1dd50dad14cba70933b6bbaa",
qop=auth,
>>>>> cnonce="4945f552", nc=00000001
>>>>> Expires: 120
>>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK7f0ddbdc;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To:
>>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f557
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 167 REGISTER
>>>>> Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
>>>>>
IP>:26462>;expires=120;received="sip:<CORPORATE OFFICE PUBLIC
IP>:26462"
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (8 headers 0 lines) ---
>>>>> [2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:24961
>>>>> handle_response_register: Outbound Registration: Expiry for
>>>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration
in 105 s)
>>>>> Really destroying SIP dialog
>>>>> '005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1'
Method: REGISTER
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK0f99cec8;rport
>>>>> Max-Forwards: 70
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as786e56f3
>>>>> To: <sip:sip.sg.didlogic.net>
>>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>>> Call-ID: 49595bb227bd6e390ca40ce6508308aa at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Date: Sun, 20 Dec 2020 07:09:07 GMT
>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>>> INFO, PUBLISH, MESSAGE
>>>>> Supported: replaces, timer
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK0f99cec8;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as786e56f3
>>>>> To:
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e4fb
>>>>> Call-ID: 49595bb227bd6e390ca40ce6508308aa at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (7 headers 0 lines) ---
>>>>> Really destroying SIP dialog
>>>>> '49595bb227bd6e390ca40ce6508308aa at
192.168.1.9:5160' Method: OPTIONS
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> [2020-12-20 07:09:52] NOTICE[2366]: chan_sip.c:15893
sip_reregister:
>>>>> -- Re-registration for  60751 at sip.sg.didlogic.net
>>>>> REGISTER 12 headers, 0 lines
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK030dc571;rport
>>>>> Max-Forwards: 70
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 168 REGISTER
>>>>> Supported: replaces, timer
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>>> nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
>>>>> response="b6e96ab578798296e812139c383ebbac",
qop=auth,
>>>>> cnonce="6313e774", nc=00000002
>>>>> Expires: 120
>>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK030dc571;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To:
>>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2fbb
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 168 REGISTER
>>>>> Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
>>>>>
IP>:26462>;expires=120;received="sip:<CORPORATE OFFICE PUBLIC
IP>:26462"
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (8 headers 0 lines) ---
>>>>> [2020-12-20 07:09:52] NOTICE[2366]: chan_sip.c:24961
>>>>> handle_response_register: Outbound Registration: Expiry for
>>>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration
in 105 s)
>>>>> Really destroying SIP dialog
>>>>> '005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1'
Method: REGISTER
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK25d81863;rport
>>>>> Max-Forwards: 70
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as3a8d9861
>>>>> To: <sip:sip.sg.didlogic.net>
>>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>>> Call-ID: 4f9667021511db2968a0dc9964673ac9 at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Date: Sun, 20 Dec 2020 07:10:07 GMT
>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>>> INFO, PUBLISH, MESSAGE
>>>>> Supported: replaces, timer
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK25d81863;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as3a8d9861
>>>>> To:
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.a139
>>>>> Call-ID: 4f9667021511db2968a0dc9964673ac9 at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (7 headers 0 lines) ---
>>>>> Really destroying SIP dialog
>>>>> '4f9667021511db2968a0dc9964673ac9 at
192.168.1.9:5160' Method: OPTIONS
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK443d3196;rport
>>>>> Max-Forwards: 70
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as5d241373
>>>>> To: <sip:sip.sg.didlogic.net>
>>>>> Contact: <sip:60751 at 192.168.1.9:5160>
>>>>> Call-ID: 201fab2b13e08922618f34911ce37ce0 at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Date: Sun, 20 Dec 2020 07:11:07 GMT
>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY,
>>>>> INFO, PUBLISH, MESSAGE
>>>>> Supported: replaces, timer
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK443d3196;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: "Unknown" <sip:60751 at
192.168.1.9:5160>;tag=as5d241373
>>>>> To:
<sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.19e3
>>>>> Call-ID: 201fab2b13e08922618f34911ce37ce0 at
192.168.1.9:5160
>>>>> CSeq: 102 OPTIONS
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (7 headers 0 lines) ---
>>>>> Really destroying SIP dialog
>>>>> '201fab2b13e08922618f34911ce37ce0 at
192.168.1.9:5160' Method: OPTIONS
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> [2020-12-20 07:11:37] NOTICE[2366]: chan_sip.c:15893
sip_reregister:
>>>>> -- Re-registration for  60751 at sip.sg.didlogic.net
>>>>> REGISTER 12 headers, 0 lines
>>>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>>>> Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK292b942b;rport
>>>>> Max-Forwards: 70
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To: <sip:60751 at sip.sg.didlogic.net>
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 169 REGISTER
>>>>> Supported: replaces, timer
>>>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>>>> Authorization: Digest username="60751",
realm="sip.sg.didlogic.net",
>>>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>>>> nonce="X975g1/e+FcgTQnFYPwx5RQy4kH7puXkamn2zYA=",
>>>>> response="8305fa5eb2ec2396b7b618f40923e597",
qop=auth,
>>>>> cnonce="043fc0dd", nc=00000003
>>>>> Expires: 120
>>>>> Contact: <sip:6531590313 at 192.168.1.9:5160>
>>>>> Content-Length: 0
>>>>> ---
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/UDP
>>>>>
192.168.1.9:5160;branch=z9hG4bK292b942b;rport=26462;received=<CORPORATE
>>>>> OFFICE PUBLIC IP>
>>>>> From: <sip:60751 at
sip.sg.didlogic.net>;tag=as6df6d977
>>>>> To:
>>>>> <sip:60751 at
sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6c72
>>>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1
>>>>> CSeq: 169 REGISTER
>>>>> Contact: <sip:6531590313@<CORPORATE OFFICE PUBLIC
>>>>>
IP>:26462>;expires=120;received="sip:<CORPORATE OFFICE PUBLIC
IP>:26462"
>>>>> Content-Length: 0
>>>>> <------------->
>>>>> --- (8 headers 0 lines) ---
>>>>> [2020-12-20 07:11:37] NOTICE[2366]: chan_sip.c:24961
>>>>> handle_response_register: Outbound Registration: Expiry for
>>>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration
in 105 s)
>>>>> Really destroying SIP dialog
>>>>> '005dbc8238e06ac421ef613a3b55e134 at 127.0.0.1'
Method: REGISTER
>>>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>>>> <------------->
>>>>> freepbx*CLI>
>>>>>> On 2020-12-23 04:29, Duncan Turnbull wrote:
>>>>>> Hi there
>>>>>> That answer includes using tcpdump to check for SIP
packets and
>>>>>> examine the register packet. At this point you have no
SIP packets
>>>>>> coming from your phone so you are not upto that stage
yet.
>>>>>> You need to know why there are no SIP packets coming.
My guess is your
>>>>>> config files have a typo in them. You can modify the
ones I sent to
>>>>>> see if they work for you. You do need to validate your
configs. I
>>>>>> recommend telnet to the phone, and checking the display
settings to
>>>>>> see if it has picked up the settings. Equally check the
phone logs. It
>>>>>> will tell you which files have errors.
>>>>>> Until you get the settings loaded correctly nothing
else will matter
>>>>>> Enjoy your break. If you want to use Voip you should
definitely spend
>>>>>> some time to learn tcpdump. If you want to use Cisco
you need to be
>>>>>> able to understand the configs yourself and get as much
info from the
>>>>>> phone as possible. Not hard but it takes a little bit
of time.
>>>>>> Cheers Duncan
>>>>>>> On Tue, Dec 22, 2020 at 10:43 PM Turritopsis
Dohrnii Teo En Ming
>>>>>>> <ceo at teo-en-ming.com> wrote:
>>>>>>> Good day from Singapore,
>>>>>>> I seem to have found the solution at FreePBX
community forums.
>>>>>>> Please
>>>>>>> check out the following discussion thread.
>>>>>>> Discussion Thread: Cisco 7940 registration problem
RESOLVED
>>>>>>> Link:
>>>>>>
https://community.freepbx.org/t/cisco-7940-registration-problem-resolved/30285
>>>>>>> But I don't understand very well what users at
this discussion
>>>>>>> thread
>>>>>>> are talking about. Can someone help me understand
better after
>>>>>>> reading
>>>>>>> through the above discussion thread?
>>>>>>> For your information, I am using PJSIP extension
instead of CHAN_SIP
>>>>>>> extension.
>>>>>>> I am planning to work on my Cisco 7960 IP phone
registration problem
>>>>>>> this coming Christmas 2020 weekends.
>>>>>>> Thank you very much for your kind assistance.
>>>>>>> On 2020-12-21 09:58, Duncan Turnbull wrote:
>>>>>>>> Hi there
>>>>>>>> I would normally highlight the part but the
email is so long I
>>>>>>> thought
>>>>>>>> I would just note what I can see
>>>>>>>> It appears the Cisco is downloading files.
>>>>>>>> None of the SIP traces show the IP of the phone
of the extension
>>>>>>>> Your proxy is at 192.168.1.9
>>>>>>>> Your phone is at 192.168.1.130
>>>>>>>> These are the details you want the phone to
pickup
>>>>>>>> line1_name: "1600"
>>>>>>>> line1_shortname: "TEO EN MING"
>>>>>>>> line1_displayname: "TURRITOPSIS DOHRNII
TEO EN MING"
>>>>>>>> line1_authname: "1600"
>>>>>>>> line1_password: "IP Phone Extension
Password"
>>>>>>>> I don't see any registration attempts from
your phone.
>>>>>>>> The first thing is to use the phone screen
display to check if it
>>>>>>>> actually has picked up the settings.
>>>>>>>> To unlock the Cisco SIP IP phone, press **#
>>>>>>>> You can also telnet to the phone usually cisco
as password, and
>>>>>>> look
>>>>>>>> at logs. Its quite possible some of your config
files are not
>>>>>>> quite
>>>>>>>> right. If they were all wrong the Cisco would
keep trying to TFTP
>>>>>>> the
>>>>>>>> files.
>>>>>>>> This is an old SIPDefault.cnf I used to use in
NZ
>>>>>>>>
===============================================>>>>>>>>
; sip default configuration file
>>>>>>>> #Image Version
>>>>>>>> image_version:P0S3-08-6-00 ;
>>>>>>>> #Proxy server address
>>>>>>>> proxy1_address: 10.12.41.1 ;
>>>>>>>> proxy_register: 1;
>>>>>>>> logo_url:
"http://10.12.41.1/Logo.bmp"                    ; URL
>>>>>>> for
>>>>>>>> branding logo to be used on phone display
>>>>>>>> time_format: 0 ;
>>>>>>>> preferred_codec: g711alaw ;
>>>>>>>> sntp_mode: unicast ;
>>>>>>>> dial_template: dialplan
>>>>>>>> sntp_server: 10.12.41.1 ;
>>>>>>>> messages_uri: "*97"
>>>>>>>> time_zone : NZST
>>>>>>>> dst_auto_adjust : 1
>>>>>>>> dst_offset : 01
>>>>>>>> dst_start_month : September
>>>>>>>> dst_start_day : 29
>>>>>>>> dst_start_time : 02:00
>>>>>>>> dst_stop_month : April
>>>>>>>> dst_stop_day : 6
>>>>>>>> dst_stop_time : 02:00
>>>>>>>>
================================================>>>>>>>>
This is a SIP Phone template - you can compare settings and notes
>>>>>>> on
>>>>>>>> the settings
>>>>>>>> # SIP Configuration Generic File (start)
>>>>>>>>
================================================>>>>>>>>
# Proxy Server
>>>>>>>> proxy1_address: "10.12.41.1"
>>>>>>>> # Line 1 Settings
>>>>>>>> line1_name: "EXTN"                   
; Line 1 Extension\User ID
>>>>>>>> line1_shortname: "0NXXXXXXX"         
; Line 1 Short Name
>>>>>>>> line1_displayname: "0NXXXXXXX"       
; Line 1 Display Name
>>>>>>>> line1_authname: "EXTN"         ; Line
1 Registration
>>>>>>> Authentication
>>>>>>>> line1_password: "6gs72ha9"         ;
Line 1 Registration Password
>>>>>>>> phone_label: "Company Limited"    ;
no effect on SIP messaging
>>>>>>>> # Line 2 Settings
>>>>>>>> line2_name: ""                       
; Line 2 Extension\User ID
>>>>>>>> line2_displayname: ""                
; Line 2 Display Name
>>>>>>>> line2_authname: "UNPROVISIONED"      
; Line 2 Registration
>>>>>>>> Authentication
>>>>>>>> line2_password: "UNPROVISIONED"      
; Line 2 Registration
>>>>>>> Password
>>>>>>>> # Line 3 Settings
>>>>>>>> line3_name: ""                       
; Line 3 Extension\User ID
>>>>>>>> line3_displayname: ""                
; Line 3 Display Name
>>>>>>>> line3_authname: "UNPROVISIONED"      
; Line 3 Registration
>>>>>>>> Authentication
>>>>>>>> line3_password: "UNPROVISIONED"      
; Line 3 Registration
>>>>>>> Password
>>>>>>>> # Line 4 Settings
>>>>>>>> line4_name: ""                       
; Line 4 Extension\User ID
>>>>>>>> line4_displayname: ""                
; Line 4 Display Name
>>>>>>>> line4_authname: "UNPROVISIONED"      
; Line 4 Registration
>>>>>>>> Authentication
>>>>>>>> line4_password: "UNPROVISIONED"      
; Line 4 Registration
>>>>>>> Password
>>>>>>>> # Line 5 Settings
>>>>>>>> line5_name: ""                       
; Line 5 Extension\User ID
>>>>>>>> line5_displayname: ""                
; Line 5 Display Name
>>>>>>>> line5_authname: "UNPROVISIONED"      
; Line 5 Registration
>>>>>>>> Authentication
>>>>>>>> line5_password: "UNPROVISIONED"      
; Line 5 Registration
>>>>>>> Password
>>>>>>>> # Line 6 Settings
>>>>>>>> line6_name: ""                       
; Line 6 Extension\User ID
>>>>>>>> line6_displayname: ""                
; Line 6 Display Name
>>>>>>>> line6_authname: "UNPROVISIONE"       
; Line 6 Registration
>>>>>>>> Authentication
>>>>>>>> line6_password: "UNPROVISIONE"       
; Line 6 Registration
>>>>>>> Password
>>>>>>>> # Emergency Proxy info
>>>>>>>> proxy_emergency: ""
>>>>>>>> proxy_emergency_port: "5060"
>>>>>>>> # Backup Proxy info
>>>>>>>> proxy_backup: ""
>>>>>>>> proxy_backup_port: "5060"
>>>>>>>> # Outbound Proxy info
>>>>>>>> outbound_proxy: ""
>>>>>>>> outbound_proxy_port: "5060"
>>>>>>>> # NAT/Firewall Traversal
>>>>>>>> nat_enable: "0"
>>>>>>>> nat_address: ""
>>>>>>>> voip_control_port: "5060"
>>>>>>>> start_media_port: "10000"
>>>>>>>> end_media_port:  "20000"
>>>>>>>> nat_received_processing: "0"
>>>>>>>> # Phone Label (Text desired to be displayed in
upper right corner)
>>>>>>>> phone_label: "Name's phone"      
; Has no effect on SIP
>>>>>>> messaging
>>>>>>>> # Time Zone phone will reside in
>>>>>>>> time_zone: NZST
>>>>>>>> time_format: "D/M/Ya"
>>>>>>>> # Telnet Level (enable or disable the ability
to telnet into this
>>>>>>> phone
>>>>>>>> telnet_level: "2"      ; 0-Disabled
(default), 1-Enabled,
>>>>>>> 2-Privileged
>>>>>>>> # Phone prompt/password for telnet/console
session
>>>>>>>> phone_prompt: ""                     
; Telnet/Console
>>>>>>> Prompt
>>>>>>>> phone_password: "cisco"              
; Telnet/Console
>>>>>>>> Password
>>>>>>>> # Enable_VAD (1-enabled, 0-disabled)
>>>>>>>> enable_vad: "0"
>>>>>>>> # Network Media Type (auto, full100, full10,
half100, half10)
>>>>>>>> network_media_type: "auto"
>>>>>>>> user_info: none
>>>>>>>> # URL for external Directory location
>>>>>>>> directory_url:
"http://10.12.41.1/directory.html"
>>>>>>>>
================================================>>>>>>>>
I would then recommend tcpdump to monitor traffic coming from
>>>>>>>> 192.168.1.130  Tcpdump is an important tool to
learn to use and
>>>>>>> you
>>>>>>>> can look at all traffic coming from the phone,
perhaps its using
>>>>>>> TCP
>>>>>>>> instead of UDP?
>>>>>>>> Have you got a copy of the Cisco SIP IP Phone
7960 Administrator
>>>>>>> Guide
>>>>>>>> - it should be on the web somewhere
>>>>>>>> Good luck, it will take a little time to get
familiar with your
>>>>>>>> environment but its important to put the time
in to work out what
>>>>>>>> means what
>>>>>>>> Cheers Duncan
>>>>>>> --
>>>>>>
_____________________________________________________________________
>>>>>>> -- Bandwidth and Colocation Provided by
http://www.api-digital.com
>>>>>>> --
>>>>>>> Check out the new Asterisk community forum at:
>>>>>>> https://community.asterisk.org/
>>>>>>> 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
>>>>> --
>>>>>
_____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by
http://www.api-digital.com --
>>>>> Check out the new Asterisk community forum at:
https://community.asterisk.org/
>>>>> 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
> 
>>> --
>>>
_____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
--
>>> Check out the new Asterisk community forum at:
https://community.asterisk.org/
>>> 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
> 
> -- 
> -----BEGIN EMAIL SIGNATURE-----
> 
> The Gospel for all Targeted Individuals (TIs):
> 
> [The New York Times] Microwave Weapons Are Prime Suspect in Ills of
> U.S. Embassy Workers
> 
> Link:
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html
> 
>
********************************************************************************************
> 
> Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's
Academic
> Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United
Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and
Australia (25 Dec 2019 to 9 Jan 2020):
> 
> [1] https://tdtemcerts.wordpress.com/
> 
> [2] https://tdtemcerts.blogspot.sg/
> 
> [3] https://www.scribd.com/user/270125049/Teo-En-Ming
> 
> -----END EMAIL SIGNATURE-----
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at:
https://community.asterisk.org/
> 
> 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

Steve Edwards

2020-Dec-24 05:06 UTC

head link

[asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

On Thu, 24 Dec 2020, Turritopsis Dohrnii Teo En Ming wrote:
> 3. secret is 8 char only, must be numeric
My my SIP<mac-address>.cnf file from 2007 contains:

 	image_version:			P0S3-8-12-00
 	line1_password:			346cc89a2526255839534c22ad7790c

and my notes say my 9760 only allowed up to 31 character passwords.

You may find it useful to use tcpdump with '-w' to write the packets to
a
file and then analyze with sngrep.

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
             https://www.linkedin.com/in/steve-edwards-4244281

asterisk users - Dec 2020 - HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI