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 > <[email protected]> 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 >>>> <[email protected]> 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 >>>>>> <[email protected]> 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 [email protected] >>>>> 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:[email protected]>;tag=as6df6d977 >>>>> To: <sip:[email protected]> >>>>> Call-ID: [email protected] >>>>> 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:[email protected]: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:[email protected]>;tag=as6df6d977 >>>>> To: >>>>> <sip:[email protected]>;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21 >>>>> Call-ID: [email protected] >>>>> 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 >>>>> '[email protected]' 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:[email protected]:5160>;tag=as41ddf4a6 >>>>> To: <sip:sip.sg.didlogic.net> >>>>> Contact: <sip:[email protected]:5160> >>>>> Call-ID: [email protected]: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:[email protected]:5160>;tag=as41ddf4a6 >>>>> To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924 >>>>> Call-ID: [email protected]:5160 >>>>> CSeq: 102 OPTIONS >>>>> Content-Length: 0 >>>>> <-------------> >>>>> --- (7 headers 0 lines) --- >>>>> Really destroying SIP dialog >>>>> '[email protected]: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:[email protected]:5160>;tag=as004073f3 >>>>> To: <sip:sip.sg.didlogic.net> >>>>> Contact: <sip:[email protected]:5160> >>>>> Call-ID: [email protected]: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:[email protected]:5160>;tag=as004073f3 >>>>> To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.385d >>>>> Call-ID: [email protected]:5160 >>>>> CSeq: 102 OPTIONS >>>>> Content-Length: 0 >>>>> <-------------> >>>>> --- (7 headers 0 lines) --- >>>>> Really destroying SIP dialog >>>>> '[email protected]:5160' Method: OPTIONS >>>>> [2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:15893 sip_reregister: >>>>> -- Re-registration for [email protected] >>>>> 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:[email protected]>;tag=as6df6d977 >>>>> To: <sip:[email protected]> >>>>> Call-ID: [email protected] >>>>> 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:[email protected]: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:[email protected]>;tag=as6df6d977 >>>>> To: >>>>> <sip:[email protected]>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e426 >>>>> Call-ID: [email protected] >>>>> 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:[email protected]>;tag=as6df6d977 >>>>> To: <sip:[email protected]> >>>>> Call-ID: [email protected] >>>>> 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:[email protected]: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:[email protected]>;tag=as6df6d977 >>>>> To: >>>>> <sip:[email protected]>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f557 >>>>> Call-ID: [email protected] >>>>> 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 >>>>> '[email protected]' 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:[email protected]:5160>;tag=as786e56f3 >>>>> To: <sip:sip.sg.didlogic.net> >>>>> Contact: <sip:[email protected]:5160> >>>>> Call-ID: [email protected]: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:[email protected]:5160>;tag=as786e56f3 >>>>> To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e4fb >>>>> Call-ID: [email protected]:5160 >>>>> CSeq: 102 OPTIONS >>>>> Content-Length: 0 >>>>> <-------------> >>>>> --- (7 headers 0 lines) --- >>>>> Really destroying SIP dialog >>>>> '[email protected]: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 [email protected] >>>>> 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:[email protected]>;tag=as6df6d977 >>>>> To: <sip:[email protected]> >>>>> Call-ID: [email protected] >>>>> 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:[email protected]: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:[email protected]>;tag=as6df6d977 >>>>> To: >>>>> <sip:[email protected]>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2fbb >>>>> Call-ID: [email protected] >>>>> 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 >>>>> '[email protected]' 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:[email protected]:5160>;tag=as3a8d9861 >>>>> To: <sip:sip.sg.didlogic.net> >>>>> Contact: <sip:[email protected]:5160> >>>>> Call-ID: [email protected]: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:[email protected]:5160>;tag=as3a8d9861 >>>>> To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.a139 >>>>> Call-ID: [email protected]:5160 >>>>> CSeq: 102 OPTIONS >>>>> Content-Length: 0 >>>>> <-------------> >>>>> --- (7 headers 0 lines) --- >>>>> Really destroying SIP dialog >>>>> '[email protected]: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:[email protected]:5160>;tag=as5d241373 >>>>> To: <sip:sip.sg.didlogic.net> >>>>> Contact: <sip:[email protected]:5160> >>>>> Call-ID: [email protected]: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:[email protected]:5160>;tag=as5d241373 >>>>> To: <sip:sip.sg.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.19e3 >>>>> Call-ID: [email protected]:5160 >>>>> CSeq: 102 OPTIONS >>>>> Content-Length: 0 >>>>> <-------------> >>>>> --- (7 headers 0 lines) --- >>>>> Really destroying SIP dialog >>>>> '[email protected]: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 [email protected] >>>>> 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:[email protected]>;tag=as6df6d977 >>>>> To: <sip:[email protected]> >>>>> Call-ID: [email protected] >>>>> 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:[email protected]: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:[email protected]>;tag=as6df6d977 >>>>> To: >>>>> <sip:[email protected]>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6c72 >>>>> Call-ID: [email protected] >>>>> 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 >>>>> '[email protected]' 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 >>>>>>> <[email protected]> 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 -- _____________________________________________________________________ -- 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
![http://10.12.41.1/Logo.bmp"](http://10.12.41.1/Logo.bmp"){kind=link}