similar to: Polycom 301 line key display

This is not an Asterisk specific question but doesn't anyone know if you can automatically prepend a 9 on the call lists so clients can return dial without having to repunch in the number? If you go to directories now it just shows the number without a 9 (obviously). Maybe on the Asterisk side?? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! Is it possible to assign any of the soft keys on the Polycom IP series handsets to a specific function in the feature menu? I'd like to assign one of the keys below the LCD to function as a Do Not Disturb button but I have not been able to find a helpful guide or proper documentation that is understandable for the task. Any help is greatly appreciated. Thank you! Tim Nelson

Hi folks, A user here has asked if we can display the current voicemail message's envelope (date/time/caller id of message) in the LCD of the Polycom phones we use (430 & 501). I realize this is somewhat like the many caller-id-after-the-fact threads, but I figured maybe someone had solved this a different way. Has anyone been able to do this, via caller ID, messaging, the mini-browser

Hello, After receiving and finally being able to configure my new Polycom IP 600 phone Here are my initial experiences: GOOD STUFF: - The sound is wonderful from both handset and speaker (711U) - The large multi-shade LCD screen is beautiful (for a phone) - The phone has an integrated Ethernet switch so you can connect a PC to it as a passthru - The AC adapter was included (small thing, but

> Is your Asterisk server listening on port 5061? If not, just change > the > entry to 5060. Also, I'm not sure how your sip.conf is set up for asterisk, but if you've set it up like: [203] type=friend username=blah secret=blah etc... Your Polycom config file will generally look like this. <PHONE_CONFIG> <OVERRIDES

I have read that 1.4 has shared line appearances, which I assume will work with Polycom phones. Has anyone configured this and verified it working? I was going to start playing around with it but wanted to see if anyone else has tackled it yet. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

Was messing with a polycom 501 and changed the IP from dhcp to static. Working with a user remotely. Now, the user says the phone does not show anything on the LCD and does not respond to any buttons. When rebooting, there is text shown as it proceeds. ?? Is there a way to reset this to a default? Does not respond to ping on the address we set. joe a.

Hello; I am using vicidial which is using asterisk 1.8, mean while when the extension has voicemail, I always see the red light on the Polycom and hear the beep sound (toot toot) in period time. Also, I can see at the LCD an option to select it for accessing the voicemail ?but I am facing the following problems: 1) The red light and the beep: How I can let the Phone only have the red light

2011/11/30 Mike <list at net-wall.com> > Hi Olivier,**** > > ** ** > > It if occurs only on the sidecar, I would imagine this is either a > defective sidecar/Polycom phone, or a defective PoE switch not giving > enough power. Changing PoE port would eliminate of confirm the PoE port > being the issue, but I?m betting on a Polycom defect.**** > > ** ** > >

I have PolyCom phones in one office working perfectly, but in another office with a new subnet, new server, new everything, the time does not work. Everything else about the phones seems fine, but the time. If you look at the internal webpage in the phone, it shows "clock". Our server, which is configured to allow others on the net to get their time from it, and it in turn gets its

A follow up (late better than never) Finally had time to sit down and look at this sip.cfg <keys key.scrolling.timeout="1" key.IP_500.31.function.prim="BuddyStatus"/> This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! Bill ________________________________ From: Bill Gibbs

Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog '

I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on

Hello, I have finally received the details from Polycom to get into the backend configuration of their SoundPoint IP 600 SIP VOIP phone. The phone is quite nice looking but the configs are very sparse, not even a place for a secret(password) field in their SIP registration section. If anyone else has one of these and needs the passwords to get into the back end configurations, just send me an

I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

This is not necessarily Asterisk specific but if I have Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend a 9 - can I do this via the polycom config? I can't find anything in the docs. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

192.168.1.107 199 6bd3fb49505 00102/00000 ulaw No Tx: ACK 192.168.0.100 110 5c5a4953-65 00101/00005 ulaw Yes Rx: ACK Those channels are stuck talking to each other. The phones are disconnected yet that connection remains. I can clear w/ a restart obviously, but is there any way to tear down a call like that from the CLI? Bill -------------- next

I've been playing with Asterisk for a few weeks and it's working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html What are people using for receptionist phones that show all the extensions in

Hi all, I'm having an odd problem with my polycom 301. I am initiating a call to it with AMI Originate() function: Action: Originate Channel: local/112 at Management Context: to_meetme Exten: s Priority: 1 Variable: dropped_conf=111 The "to_meetme" context is very simple: [to_meetme] exten=>s,1,MeetMe(${dropped_conf},id) If I specify every other device I have to test: *