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


(Turritopsis Dohrnii Teo En Ming) #1

Good day from Singapore,

I can’t post everything here due to a cap on the maximum number of characters. My post is too lengthy. Please refer to the following link instead.

http://lists.digium.com/pipermail/asterisk-users/2020-December/295555.html

Thank you very much.

Mr. Turritopsis Dohrnii Teo En Ming


#2

By default, FreePBX has pjsip listening on port 5060 and chan_sip on port 5160. Your system is still set that way, which is good.

But your Cisco is set to register to port 5060 and you are showing the log with ‘sip set debug on’, which refers to chan_sip. sip show peers also refers to chan_sip.

If you have set up pjsip extensions for Cisco and Bria, then you must issue
pjsip set logger on
to see registration attempts, etc., and give the appropriate pjsip commands to view the state of the extension in Asterisk.

OTOH, if you set up a chan_sip extension for Cisco (not recommended), it’s apparently not there (it would appear in sip show peers even if the phone were not connected). And, the Cisco is attempting to register to the wrong port.

When you post new logs, etc., please paste them at pastebin.freepbx.org and post the link here.


(TheJames) #3

79XX phones are awful
That said see…


(Dave Burgess) #4

79xx phones in SIP mode are awful.
That said, see…


(Turritopsis Dohrnii Teo En Ming) #5

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:

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.

Mr. Turritopsis Dohrnii Teo En Ming
Singapore


#6

In the Advanced settings for the extension, set both Rewrite Contact and Force rport to No. Reboot phone.

If it still won’t register, at the Asterisk command prompt type
pjsip set logger on
paste the Asterisk log for an attempted registration at pastebin.freepbx.org and post the link here.


(Turritopsis Dohrnii Teo En Ming) #7

Thank you Stewart1,

I have set Force rport to No for my pjsip extension. But I did not set Rewrite Contact. Now my Cisco 7960 IP phone registers successfully on my Asterisk PBX server.

Thank you very much.

Mr. Turritopsis Dohrnii Teo En Ming
Singapore


(Turritopsis Dohrnii Teo En Ming) #8

Hi,

I have finally managed to get my Cisco 7960 IP phone to register on my
Asterisk PBX appliance on Christmas Eve 2020.

You can read my guide here:

Guide: Teo En Ming’s Guide to Configuring Asterisk/FreePBX with Cisco 7960 IP Phones
Link: lists.digium.com-pipermail-asterisk-users-2020-December-295581.html

However, there is still a problem. Please read Section 12. Do you know
how to solve it??

SECTION 12: CAVEATS

I can now make outgoing phone calls on my Cisco 7960 IP phone.

However, if Voicemail in my extension configuration is set to Disabled,
and when I try to dial my DID number, it says “Number not valid”.

Only when I change Voicemail to Enabled, and when I try to dial my DID
number, it says:

“The person at extension 1600 is unavailable. Please leave your message
after the tone, when done hang up or press the # key.”

It seems that incoming calls will not be routed to extension 1600 on my Cisco 7960 IP phone.

I will need to do more troubleshooting on this at a future date.

If you know how to solve this, please let me know.

Thank you very much!

Merry Christmas 2020!

Mr. Turritopsis Dohrnii Teo En Ming
Singapore


#9

For the 7960 to receive calls, both Force rport and Rewrite Contact must be set to No.

If you have these settings but it’s not working, try calling it from another extension. If that also fails, paste the Asterisk log for an internal call, including SIP trace, as described earlier in this thread. If internal calls work but external incoming calls fail, paste the log for an external call.


(Turritopsis Dohrnii Teo En Ming) #10

Hi Stewart1,

My final issue has been resolved.

Thank you very much.

Merry Christmas 2020!


(Turritopsis Dohrnii Teo En Ming) #11

Subject: Addendum to Teo En Ming’s Guide to Configuring Asterisk/FreePBX with Cisco 7960 IP Phones

Author: Mr. Turritopsis Dohrnii Teo En Ming

Country: Singapore

Date: 25 Dec 2020 Friday Singapore Time

Type of Publication: Plain Text

Document version: 20201225.01

=========================================================================================

Please refer to Teo En Ming’s earlier guide.

Guide: Teo En Ming’s Guide to Configuring Asterisk/FreePBX with Cisco 7960 IP Phones
Link:

In the above mentioned guide, under Section 12: Caveats, it was mentioned as follows:

"I can now make outgoing phone calls on my Cisco 7960 IP phone.

However, if Voicemail in my extension configuration is set to Disabled,
and when I try to dial my DID number, it says “Number not valid”.

Only when I change Voicemail to Enabled, and when I try to dial my DID
number, it says:

“The person at extension 1600 is unavailable. Please leave your message
after the tone, when done hang up or press the # key.”

It seems that incoming calls will not be routed to extension 1600.

I will need to do more troubleshooting on this at a future date."

Now, I am pleased to announce that, this final issue has been resolved on Christmas Day 2020.

The steps taken to resolve this final issue are as follows:

Login to Teo En Ming’s FreePBX GUI at .

Click Applications > Extensions.

Click the Pencil icon to edit the PJSIP extension 1600.

Click Advanced tab.

For Rewrite Contact, you MUST change the setting from Yes to No.

Click Voicemail tab.

Change Enabled to Yes.

For Require From Same Extension, click Yes.

Click Submit.

You MUST click Apply Config for the changes to take effect!

Voila! All issues have been solved.

Thank you Stewart1!

Merry Christmas 2020!


(system) closed #12

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