Hi,
I have a FreePBX system up and running, with everything seemingly working fine, based on the ability to call one computer from another computer using SIP. Now I want to connect a Cisco 7960 phone to the system and this is where I’m running into problems… I can’t place a call nor receive a call.
The phone is loading it’s firmware fine off a TFTP server, on the same system as the FreePBX server. The extension for the phone is set up OK and I’ve confirmed (multiple times) that the name/password are correct, i.e., FreePBX matches the phone configuration.
The extension uses the CHAN_SIP driver and when placing a call from the phone, the following error message is noted in the Asterisk logs:
[2014-07-06 19:49:19] NOTICE[8178] res_pjsip/pjsip_distributor.c: Request from '"200" <sip:[email protected]>' failed for '192.168.1.178:50905' (callid: [email protected]) - No matching endpoint found
[2014-07-06 19:49:20] NOTICE[10189] res_pjsip/pjsip_distributor.c: Request from '"200" <sip:[email protected]>' failed for '192.168.1.178:50905' (callid: [email protected]) - No matching endpoint found
[2014-07-06 19:49:22] NOTICE[6087] res_pjsip/pjsip_distributor.c: Request from '"200" <sip:[email protected]>' failed for '192.168.1.178:50905' (callid: [email protected]) - No matching endpoint found
(No messages are noted in the logfiles if I use Generic PJSIP driver.)
Any help would be appreciated - I’m not sure where I’m going wrong with the set-up. All that I’ve read would suggest that the phone should simply be working now.
An answer to the question: I (stupidly) assumed that the beta version of the system would overall work OK. So I re-installed everything to:
FreePBX 2.11.0.37
Asterisk 11.10.2
and it now works sort of OK. (I was running the server on a VirtualBox system.) And as I say, stupidly…
However, there is a new problem:
I can make calls from the phone to a different soft-phone extension (and this includes a *65 which identifies the phone’s extension), but am unable to call the phone from a different extension. The phone’s extension doesn’t appear as a connected “IP phone online” either on the FreePBX base administration page.
The Cisco 79XX phones just dont play nice period with SIP in Asterisk. I am not a expert on them but I know you have to make sure I believe qualify and nat is off on the extension page.
Thanks @tonyclewis - I already had these settings per what you stated, because I had read as much elsewhere. But still good to get confirmation.
What I believe was the issue to the latest problem was: I didn’t have the “dynamic TFTP server address” (dyn_tftp_addr) set correctly in the default SIP configuration file (SIPDefault.cnf) for the phone. It was the only change I made, a reset of the phone and everything was working! (There are a number of indicators - not to mention which that the phone now shows the correct time, given that I set up an appropriate SNTP server address in the configuration file long ago…)
I’ve confirmed the situation by not only by making/receiving calls, but also from the Asterisk CLI, using the command ‘sip show peers’ and the phone is definitely there. It is “unmonitored” - I don’t think I’m exactly game to change the settings to change this situation - if it’s at all important, particularly given what you’ve already mentioned!
The Cisco 7960 works fine with Astetrisk FreePBX, you need to understand all that it entails, though. You need a tftp server that has several files to support the config of the phone. You will need the SIP firmware ( I am using P0S3-8-12-00) available from http://www.jtech.net/ip_phone/cisco/Cisco_firmwre.aspx. You will also require several asci files you can edit for config (XMLDefault.cnf.xml, SIPDefault.cnf, OS79XX.TXT, and a SIP{MAC ADRESS OF YOUR PHONE}.cnf for each phone you have). I have my DHCP server configured with dhcp_option_66 which points to the tftp server ip address. When the Cisco 7960 powers up, it gets its IP Address from my DHCP server, then goes to the TFPT server (the server running FreePBX, or Asterisk) and gets the remainder of the boot up config stuff. The SIP{MAC ADDRESSOF YOUR PHONE}.cnf file contains the extensions and passwords for your phones to connect to Asterisk.
If you can’t configure your dhcp server to have option 66, you can unlock the 7960 using the default password and select Alternate TFTP in Network Confiruation and set it to YES, then scroll up to the TFTP Server IP address and edit it to point to your Asterisk PBX ip address.
Hope that helps.
smarcuss