Polycom Phone Question

[TheBigEye]

I have a few Polycom phones. They have the function to transfer calls and forward calls. When I enable the forwarding and put my phone number for forwarding, nothing happens. I call the extension, the extension doesn't even ring... just goes dead. From the call log on the phone, I can see that it logged the call, but it's not sending the call out. Anybody got a clue? I know that when I make calls from the phone, I have to hit the send button after I dial the number I want to call. Anyway to set the dial pattern to automatically send the call?

 

[5qg4]

TheBigEye,

Please post following information.
3CX version?
Phone model?
3CX Server status log (console log).

 

[TheBigEye]

Polycom 601
3cx Free Version (unsure of version number, but recent within the last month)

10:38:58.128 ServRegs::checkExpiration [CM113000] Registration for sip:[email protected] has expired 
10:29:43.691 StunClient::process [CM215001] STUN failed to resolve external IP using server 69.25.47.164 
10:09:40.276 StunClient::process [CM215001] STUN failed to resolve external IP using server 69.25.47.164 
09:56:13.947 StratLink::onHangUp [CM104001] Call(1480): Ln:10002@Junction Networks hung up call; cause: BYE; from IP:66.227.100.20 
09:56:11.072 CallLegImpl::onConnected [CM103001] Call(1480): Created audio channel for Ln:10002@Junction Networks (66.227.100.16:17112) with Media Server (69.153.217.182:49154) 
09:56:11.057 StratInOut::onConnected [CM104005] Call(1480): Setup completed for call from Ln:10002@Junction Networks to Ln:10002@Junction Networks 
09:56:11.057 CallLegImpl::onConnected [CM103001] Call(1480): Created audio channel for Ln:10002@Junction Networks (66.227.100.17:11858) with Media Server (69.153.217.182:65533) 
09:55:41.074 CallConf::onProvisional [CM103003] Call(1480): Ext.100 is ringing 
09:55:40.808 CallConf::onIncoming [CM103002] Call(1480): Incoming call from 17136681717 (Ln:10002@Junction Networks) to sip:[email protected] 
09:54:03.064 StratLink::onHangUp [CM104001] Call(1479): Ln:10002@Junction Networks hung up call; cause: BYE; from IP:66.227.100.20 
09:53:59.361 CallLegImpl::onConnected [CM103001] Call(1479): Created audio channel for Ln:10002@Junction Networks (66.227.100.16:10182) with Media Server (69.153.217.182:65506) 
09:53:59.361 StratInOut::onConnected [CM104005] Call(1479): Setup completed for call from Ln:10002@Junction Networks to Ln:10002@Junction Networks 
09:53:59.361 CallLegImpl::onConnected [CM103001] Call(1479): Created audio channel for Ln:10002@Junction Networks (66.227.100.17:19816) with Media Server (69.153.217.182:65501) 
09:53:00.224 CallConf::onProvisional [CM103003] Call(1479): Ext.100 is ringing 
09:52:59.990 CallConf::onIncoming [CM103002] Call(1479): Incoming call from 17136681717 (Ln:10002@Junction Networks) to sip:[email protected] 
09:52:42.303 StratInOut::onHangUp [CM104007] Call(1478): Call from Ln:10002@Junction Networks to 800 has been terminated by Ln:10002@Junction Networks; cause: CANCEL; from IP:66.227.100.20 
09:52:26.632 CallConf::onIncoming [CM103002] Call(1478): Incoming call from 17136681717 (Ln:10002@Junction Networks) to sip:[email protected] 
09:52:26.632 MediaServerReporting::STUN [MS101003] Call(1478) Ln:10002@Junction Networks: Possible firewall problem. Address mapping failed on STUN server 69.90.168.14:3478 for local address ":9004"

 

[TheBigEye]

The problem is with the Digitmap I think. I currently don't have one and can't find what to put in there.

 

[5qg4]

Digitmap should be under SIP > LOCAL SETTING

 

[TheBigEye]

What do I put for the digit maps? What should go there??

 

[5qg4]

Please found the setup guide at http://www.3cx.com/sip-phones/polycom-soundpoint.html

For Digitmap:
1. Connect the Polycom SoundPoint IP 601 SIP phone to the network and mains power supply. The phone will boot up, and providing you have DHCP installed on your network, will obtain an IP address. To find the IP address, using the phone keypad, go to “menu -> status -> network -> tcp-ip parameters” and view its IP address.

2. Now you can access the phone’s web configuration page by going to http://x.x.x.x where x.x.x.x is the IP address of the phone from step 2.

3 The factory default username and password for accessing web configuration on Polycom phones is “Polycom” and “456” (make sure Polycom starts with a capital P).

4. Click SIP
5. Click Local Setting

You will found the Digitmap over here.
eg.
If your extension with 3 digits, start with 8 and "9" outbound rule to PSTN
[8]xx|[9]xxxx.T
xxxx, xx - min. digits
[8]xx - If press 801, it will issue a SIP call to extension without press dial.
[9]xxxx.T - If press 912345678, it will issue a SIP call to PSTN number 12345678 without press dial.

It works for our IP330, Hope it works at IP601 as well. Since them almost the same setup.

 

[TheBigEye]

That works! THANKS!!!

 

[5qg4]

Actually, we are not using web interface to configure all the Polycom phones. All of them are using FTP to auto download the config and update the FW during reboot.

What we need to config the phone, we just need key in the FTP server IP address into the phone setting. And we also need put all necessary config files and FW file to that FTP server.

It should be support HTTP and TFTP to download the config files as well..

 

[TheBigEye]

I dont' know how to setup the config files for ftp download. I'd like to, so that I can put a graphic on the phone as well.