Polycom IP Phones Intercom HELP Please!

[ciscotech2007]

Thank you for your time.

Picked up several Polycom 330 IP Phones for a good deal. I have worked with my Cisco 7940/7960's for awhile now so I have experience in setting up IP phones with 3CX. I was hoping the polycom phones would be easier but they are just as challenging if not more.

For the sake of this conversation, I have a 3CX server setup with (1) unmanaged switch and (1) Polycom IP Phone with SIP firmware. The phones are rather new but the firmware was V2 and the polycom website has the newer V3 SIP firmware. I booted the phones to my TFTP server and updated the firmware. I verified the update by checking the software version on the phone. I factory reset and local config clear'ed the phone right after the update to clear out any old configs since the phones were used for a few months.

For the most part the polycoms are similar to my cisco IP phones in that they can be configured through TFTP test (CFG) files or locally through the phone's interface. The cisco phones do not keep a web interface when converted to SIP though. The polycoms do have a web interface... however its clunky and requires a restart for every change you make.

Just so we start clean, the server is up, the phone is up and showing the SIP line I registered in 3CX. Once the polycom boots the 3CX server shows the phone registering. I can call the polycom from my cisco phones.. and I can call the cisco phones from the polycom.

There are several features I am trying to get working and I have been researching for hours and cannot seem to find;

We use the intercom(auto-answer 2-way audio) via our 3CX extensions(cisco hardphones currently).... How do you get intercom working on the polycom phones? On my ciscos it's EASY, just select auto-answer and that SIP line auto-answers! I have explored the SIP.CFG(general CFG file) file and found 'ring types' then saw 'AUTO ANSWER' listed in the types available and then explored the phone1.CFG(phone specific CFG file) and found my line #1 registration information and then found 'ring type' and it had no entry listed. This is a huge pain. Via the 3CX server PDF for installation, I have even tried to dial *90 THEN the extension and get a 'the person your trying to call is unavailable'.

Can anyone give me some information on enabling intercom on polycom phones? And if you provide an answer please give me a step by step explanation so I can clearly figure out what your trying to explain.

I have used some tips and tricks from here if anyone is curious:

http://www.computechgroup.com/?p=385

(I believe this solution is for trixbox though because there's no dialplan or syntax in 3CX that reads 'exten => _*90.,1,SIPAddHeader(Alert-Info: Ring Answer)')

http://sipx-wiki.calivia.com/index.php/HowTo_configure_Polycom_SoundPoint_IP_phones_with_sipX

(Yeah I know its SipX but I was just searching SIP server polycom setup and configuration forums and help)

Appears MOH(music on hold) works. Both calling the polycom and putting it on hold and the polycom calling other cisco phones and putting them on hold.

Voicemail on the phone works IF you do not answer the polycom when calling from a cisco. I set a basic forwarding rule in the 3CX to forward unanswered calls to the polycom's extension VM. But if you 'reject' the call from the polycom's softkeys my cisco phone gets 'three high beeps and a woman saying 'the person you are trying to call is unavailable''. Any ideas on why is is happening? I presume my pressing the reject softkey on the polycom sends back a SIP message to the 3CX server and my 3CX server does not know how to handle it...or send it to the polycom's VMbox.

Anyone know how to setup the screen animations or logos? Supposly most of the polycom phones support either a logo or Idle screen..... I have very limited information on this capability... though my SIP firmware download from polycom's site included several images but none seemed to be for my 330s... though I did not spend much time reviewing the images yet...

How do you access voicemail on the polycoms from the phone itself? On my cisco phones I simply press the 'messages' button, which I linked the messages URI to the 3CX's main voicemail box, which I think was 999, but I dont remember off the top of my head.

I am able to leave a VM and the MWI lights up indicating a VM has been left. (I am running enterprise demo which supports MWI). It appears after leaving a VM the phone also BEEPs every 30 seconds in addition to the MWI light on. After I left a VM the screen on the polycom scrolls missed calls and (1) new VM. I select the 'MSGS' (messages) softkey and that pulls up "1. Message Center' as the only option. I use the center of the arrow pad (which is a OK or ENTER key) and then it shows '1 MESSAGE 0 URGENT'. Then there are (2) options, CONNECT and CLEAR. I selected CONNECT and the polycom's extension pops up on the polycom's screen and I get a busy signal..... probably because I need to program the 3CX main VMbox number... but where ..in the phone? web interface? config files?

thanks for your time and please be full and complete with any advice...

 

[William400]

Hi

Thanks for your detailed explanations.

As this point in time Polycom intercom function cannot be acheived via the HTTP interface of the device. This can only be done via the CFG files. In 3CX V7, we have added support for the SoundPoint_IP range provisioning . This includes the intercom function. have you tried this out?

Since you have been trying our various things on the phone it would be ideal to flash it to factory default ensure you are on firmware 3.2 and then proceed from there.

As to the 'Reject' option it appears that unlike other phones the Polycom is sending a 603 rather than a 486 when the reject is used. We will need to investigate this.

As to the MWI, you would need to set this manually in the UI or provisioning files. We shall look into this also.

 

[ciscotech2007]

Thank you for your help!

Yes you are right... the HTTP and Phone menu options are there... but they are very limited. In contrast to my ciscos the HTTP and Phone menu programmable options are not good. They give little configuration capability. YES, you can use the CFG files for a very detailed configuration option... But thats the issue.. the CFG files editing is difficult and cumbersome. I would much rather config each phone manually through it's HTTP interface or locally on the phone itself(through the menu). Yes, I know CFG file downloads make provisioning quick for 20 phones but I normally work with 3 - 8 phones and dont mind manually configuring the phones to entire they have the correct values set. I guess if I had great working configuration CFG files and knew every option in the CFG file and just changed the variables per project TFTPing the CFG files would be FAST and EASY.

Can you describe how to use the provisioning option in 3CX? It looks rather simple, add the MAC of the phone, select the model and the interface the phone is on... then what? Set the phone TFTP server to the 3CX server and when the phone boots it contacts the 3CX server and it TFTP serves the 3CX-generated CFG file? How do you configure multiple lines since the provisioning is setup when you configure (1) 3CX SIP line? How do you enable or disable intercom function using the provisioning? Because the intercom 'auto-answers' calls how does the 3CX know to enable or disable that feature through provisioning?

Thanks and look forward to responses

 

[ciscotech2007]

Anyone?

OR Can anyone post a WORKING polycom ip phone configuration file (sip.CFG) and phone1.CFG file WITH INTERCOM working (i.e. call the 3CX extension that polycom is configured with and the phone 'auto-answers giving you 2-way audio without the person having to pickup the phone'...)

 

[ciscotech2007]

Here's what my CFG files have on my polycoms. Keep in mind, I have reset to phones via;

1. delete local config
2. delete local settings
3. Format file system
4. TFTP boot and installed latest SIP firmware

This process was completed to 'clear' out any local setting or values that may have existing on the phones. These values may have been added via the phone's menu on the phone itself or the phone's HTTP configuration webpage.

It appears there are (4) main files the polycom uses during boot;

1. [MAC].CFG / 0000000000000.CFG - used to identify the firmware load for possible upgrading
2. sip.ld - actual SIP firmware (39MB)
3. phone1.CFG - phone SIP registration configuration file for each phone
4. sip.CFG - global configuration file for all phones

I dont mess with #1 or #2 file and #4 really states global variables it appears. sip.CFG does have some IP address and other information for for the sake of testing I have only modified the phone1.CFG configuration file. This file appears to be where you input your environment settings such as the IP address of your 3CX server, port, SIP user ID, SIP auth password, etc. The field I am most interested in is the [ringtype=""] field. It appears this is the location where you change the ringtype associated with that specific SIP line configuration. Each ringtype is defined in the sip.CFG main file as seen here;

<ringType se.rt.enabled="1" se.rt.modification.enabled="1">
<DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="2" se.rt.1.callWait="6" se.rt.1.mod="1"/>
<VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual"/>
<AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/>
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>
<INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring" se.rt.5.ringer="2" se.rt.5.callWait="6" se.rt.5.mod="1"/>
<EXTERNAL se.rt.6.name="External" se.rt.6.type="ring" se.rt.6.ringer="2" se.rt.6.callWait="6" se.rt.6.mod="1"/>
<EMERGENCY se.rt.7.name="Emergency" se.rt.7.type="ring" se.rt.7.ringer="2" se.rt.7.callWait="6" se.rt.7.mod="1"/>
<CUSTOM_1 se.rt.8.name="Custom 1" se.rt.8.type="ring" se.rt.8.ringer="5" se.rt.8.callWait="7" se.rt.8.mod="1"/>
<CUSTOM_2 se.rt.9.name="Custom 2" se.rt.9.type="ring" se.rt.9.ringer="7" se.rt.9.callWait="7" se.rt.9.mod="1"/>
<CUSTOM_3 se.rt.10.name="Custom 3" se.rt.10.type="ring" se.rt.10.ringer="9" se.rt.10.callWait="7" se.rt.10.mod="1"/>
<CUSTOM_4 se.rt.11.name="Custom 4" se.rt.11.type="ring" se.rt.11.ringer="11" se.rt.11.callWait="7" se.rt.11.mod="1"/>
</ringType>

(this is just an exerpt from the sip.CFG file, specifcally the 'ringtypes' section)

Below is my phone1.CFG file;

<phone1>
<reg reg.1.displayName="666" reg.1.address="666" reg.1.label="666" reg.1.type="private" reg.1.lcs="" reg.1.csta="" reg.1.thirdPartyName="" reg.1.auth.userId="666" reg.1.auth.password="666" reg.1.auth.optimizedInFailover="" reg.1.musicOnHold.uri="" reg.1.server.1.address="192.168.0.200" reg.1.server.1.port="5060" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="60" reg.1.server.1.expires.overlap="" reg.1.server.1.register="" reg.1.server.1.retryTimeOut="" reg.1.server.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.server.1.lcs="" reg.1.outboundProxy.address="192.168.0.200" reg.1.outboundProxy.port="5060" reg.1.outboundProxy.transport="" reg.1.acd-login-logout="0" reg.1.acd-agent-available="0" reg.1.proxyRequire="" reg.1.ringType="3" reg.1.lineKeys="" reg.1.callsPerLineKey="" reg.1.bargeInEnabled="" reg.1.serverFeatureControl.dnd="" reg.1.serverFeatureControl.cf="" reg.1.strictLineSeize=""

(this is just an exerpt from the phone1.CFG file, specifcally the 'Line 1 registration' section, notice the bold section I made to point out the ringtype option)

In the default polycom firmware package, the phone1.CFG file contained the option [reg.1.ringType="2"]. So I referenced the sip.CFG file under ringtypes and could not coorlate "2" with the correct option... because "2" under ringtypes is actually 'visual'.. which means the phone flashes during calls but does not ring. So obviously, even though the ringtype is defined in the phone1.CFG file... the phone is not taking the factory default ringtype AND it's not taking my newly defined ringtype of "3"... why 3?... I dont know.. I just was trying to get something to work...

Keep in my the phone is registering on my 3CX and I can call and make calls.. only when I call the polycom it RINGS but will not AUTO ANSWER(aka intercom)

What in the heck am I doing wrong and why does polycom make this so difficult????

 

[ciscotech2007]

ok...MADE PROGRESS

In my testing, i changed ringtype values to other values and was testing if auto-answer was working. But i noticed something ODD.... the ringer sounded differently.... and then it came to me... in the phone1.CFG, under the LINE registration settings, under ringtype=x... that option is not for auto-answer/intercom ITS TO provision the ring TYPE, as in the phone's audible ringer... NOT the RING TYPE. AHH!

I tried to check this by changing the values on the phone back to "2" and then setting the phone1.CFG file ringtype to 6... only after a reboot and checking the phone's menu it showed ringer 2.. at first I was like... what the heck? Then i remember, once you make a setting change on the phone.. the phone's local settings override any provisioning from the TFTP server's CFG files...

nonetheless, I am fairly certain the ringtype is not the place to set auto-answer.... so now I am off on the search elsewhere in the CFG files to set auto-answer or intercom on these darm polycoms!

 

[ciscotech2007]

Here's auto-answer/intercom setup for polycom's on ASTERISK... DANG!... their solution works in asterisk but not in 3CX.... so close

http://www.voip-info.org/wiki/view/Polycom+auto-answer+config

 

[ciscotech2007]

ok... so in the meantime of waiting for support I went ahead and set the polycom to get it's configuration file from the 3CX server..

first I went into 3CX and went under my polycom extension.. went to the provisioning tab and entered the polycom's MAC/SIP firmware/and interface the phone is on.

Then I rebooted the polycom phone and reconfigured the boot settings away from my TFTP server.. and redirected it from TFTP to HTTP and pointed the setting to 'http://192.168.0.200/management/provisioning'

(note: 192.168.0.200 = IP of my 3CX server)

Then, i went to my other 3CX hard phones and dialed *9, then the extension and INTERCOM works! Well, it rings once then auto-answers with 2-way audio. So this would technically be 'ring-answer' where the phone rings once then goes to intercom/auto-answer.

I am not happy though... because I would like to know how exactly provisioning accomplished intercom and why I could not accomplish it editing the configuration CFG files. On my cisco 7940s/7960s you setup the number of SIP lines you want the phone to register and then you use the phone's menu to select which SIP line will auto-answer... fairly easy and straight-forward. BUT on these polycoms... theres no simple setting for this. Yes, the provisioning works but there's 2 problems

1) what exactly does provisioning 'do' and how does it accomplish 'intercom/auto-answer' features on the polycom and why can't I edit the CFG files to accomplish it?

2) Because I am relying on 3CX to accomplish the intercom/auto-answering you have to dial *9 and then the extension...which means in a mixed IP phone enirnment you would have to remember which phones are cisco(dont require *9+extension) and which are polycoms(require *9 and extension) to get intercom functions.... users would not be able to remember. And if your asking why would you have 2 types of IP phones... well my managers use cisco ip phones for the extended features/multiple lines while the regular employees get a basic SIP phone with 1 SIP line...

 

[ciscotech2007]

dont get me wrong, I am happy I got intercom working... but its much more important to understand whats going on behind the scenes... I want to get to know 3CX in and out and knowing a product and how it works is very important to me

Plus in this situation, learning whats going on here will help me in the future when working with other IP phones and troubleshooting issues...

 

[Kevin Attard Compagno]

Hi there

The *9 simply causes 3CX PhoneSystem to send the SIP INVITE message to the phone REQUESTING Intercom functionality by adding the "answer-after" parameter in the "Call-info" Header-Line to the SIP INVITE:

Call-Info: <<sip:[email protected]>;answer-after=0>

Any phone which recognizes the command will "Auto-Answer". Any phone which does not will ignore the command and ring as normal.

Hope this clarifies...

Regards

Kevin

 

[Discovery Technology]

We have just completed a successful setup of a Polycom soundstation IP 7000 with 3CX and we found the following area on the Polycom website incredibly useful... (sorry about all the reading you're about to do though - at least there's plenty of info available here)

You can download the latest firmware at the top of the link, and down the bottom there are downloadable administrator guides with command reference info for your phone config files.

http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip330_320.html?product_type=2&category=%2Fsupport%2Fvoice%2Fsoundpoint_ip%2Fsoundpoint_ip330_320.html%23document

This should have everything you're after...

 

[danielnm]

Can you please post a config file or at least the key settings that made a difference.

I have recently purchased 2 IP7000 phones and I am trying to set them up as remote extensions.
All works fine except that when the Polycom rings, the call can not be answered.

Daniel