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Configuring 877w and 7940 with SIP firmware for VOIP

jared.j01
Level 1
Level 1

Hello,

I have a Cisco 877w and Cisco 7940 IP phone.

The IP phone has SIP firmware loaded and I have configured the correct username, password and SIP server address details on the phone.

On the router everything seems to be working and I have also configured NAT.

I cannot seem to get the phone registering with the SIP server (as far as I can tell), and I cannot make or receive calls.

I will post my config shortly, but I am after some general guidelines as to what configuration I need to make on the router.

Do I have to forward any ports?

I have the IP phone plugged into Fa0/0 and it is getting an address assigned by DHCP.

Thanks for your assistance.


Sent from Cisco Technical Support Android App

6 Replies 6

Leo Laohoo
Hall of Fame
Hall of Fame

Your 7940 need to talk to a Call Server.  Cisco's solution is called Cisco Unified Call Manager (CUCM) and all other alterations.

If you are doing this for home and you don't have the funds to purchase a CUCM, then you need to search for an alternative solution like Asterisk.


Asterisk comes in multiple flavours:  You can have the bare CLI version or other alterations such as FreePBX or TrixBox, etc.

You can even run Asterisk on a very affordable Raspberry Pi or a Beagle Bone. 

Sorry for the delayed response with this!

I have made good progress. I previously configured the phone via the keypad and entered the authentication name, password and proxy address etc.

It would not work until I also enabled "register with proxy". Now I can call the VoIP phone from my mobile and it rings and I can have a conversation.

 

I deleted my previous port forwarding so there is nothing configured now.

The problem is when I try to make an outgoing call, my mobile phone rings but I cannot hear anything.

 

I just entered "NAT Address" via the keypad on the phone under SIP Configuration and made it my WAN IP address, but it did not solve the issue.

Any ideas?

 

The Cisco IP phone's IP address is 192.168.2.2 and is allocated via DHCP. I have not made it static at the moment.

Below are the IP NAT translations when making an outgoing call (no voice can be heard)

udp MYWANIP:1030 192.168.2.1:5060 203.221.91.30:5060 203.221.91.30:5060

udp MYWANIP:16386 192.168.2.1:16386 203.221.91.30:18190 203.221.91.30:18190

udp MYWANUP:16387 192.168.2.1:16387 203.221.91.30:18191 203.221.91.30:18191

 

 

IP NAT translations when making an incoming call (no problems with the call)

udp MYWANIP:1032 192.168.2.1:5060 203.221.91.30:5060 203.221.91.30:5060

udp MYWANIP:16388 192.168.2.1:16388 203.221.91.30:18308 203.221.91.30:18308

udp MYWANIP:16389 192.168.2.1:16389 203.221.91.30:18309 203.221.91.30:18309

 

Also how can I telnet or SSH into the phone and how can get the phone's SipDefault.cnf and other configuration files?

 

Thank you very much for the assistance!

 

 

jared.j01
Level 1
Level 1

I thought a Cisco 7940 with SIP firmware can work with a SIP VOIP provider?

Apart from the router configuration, I thought it would work very similar to a SIP softphone as in username, password and SIP server address settings.


Sent from Cisco Technical Support Android App

I thought a Cisco 7940 with SIP firmware can work with a SIP VOIP provider?

Yes and no.  Depends on your VoIP Servce Provider (VSP).

Some VSP, like mine, will only allow one session at any time.  This means if I have two or more handsets, like mine, only one handset will be able to authenticate with the VSP.  The other handset will be able to authenticate but will be unable to receive any incoming call.

Now back to your issue, check with your VSP for their settings.  Run a debug from the phone and find out what is going on.  One easy option is to install a software-based "phone" and get that to authenticate with your VSP.  Once you iron out the configuration issue with your sofware-based "phone" and your VSP, you can easily "export" the config to your hardware.  Does this make any sense to you?

Hi Jared.

You are correct. You don't have to configure any specific NAT for your IP phone to be able to talk to the VSP.

Please, as soon as you can, post your SipDefault.cnf file.

Regards

Carlo

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Hello and thank you for your replies.

I have been cofiguring the IP phone via the keypad buttons, so I am not sure how to export the

SipDefault.cnf file.

I will work on it again over the weekend and post my findings.

Good idea, I will try a Cisco softphone and see if I can get that working and go from there.

Will report back soon.

Thanks for the help.