similar to: CallerID Error causes problems for Polycom phones

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: *

Contact me off list, I have a sample extensions.conf file that I can share. It has Paging (one to one and One to Many) Ivr includes, time of da routing and it is geared towards Polycoms. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Thomas Johnson > Sent: Friday, January 20, 2006

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

I know the Polycoms work with NAT, but you have to specify the public IP. Is there anyway for it to discover the external IP automatically? I like the phones (been playing with a 301) but for some of our clients who have a dynamic IP (and no hope of getting a static ie cable or residential DSL) I'd be afraid to use them since you have to specify the IP. What about the Cisco phones?

Just got one of these. Horrible to program. Trying to key in the FTP server. Won't even remember the info after rebooting. Anybody know the proper way to beat on this stupid beast so it will work?

Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

Guys, this is a weird question but has anybody disabled the blind button that appears on polycoms or know if you can disable the use of blind transfers on polycoms to make any transfer attended? Thx!

Get the new firmware - it's supposed to have changed the callerid display presentation to include name and number. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Coulthurst Sent: Friday, May 20, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Displayed CallerID on

Hi all, I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says "Device" is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of

Hi all, I'm having a problem with a polycom 301 not picking up a ZAP call. Below is the CLI output of the call. I have: TDM400 with 2 FXO lines Asterisk 1.2.14 Polycom 301 When I dial the first ZAP line, I choose an extension that rings the polycom, polycom rings and I can pick it up and the call is bridged. When I call my second zap line, the polycom rings, but I cannot pickup the

I'm looking to send calls to a phone with no callerid data whatsoever shown on the Polycom as far as missed call. The specific application for this is that I have a 50 phone install with some being used for paging. Paging works perfectly, but the problem is that for every page there is a "missed call" shown on the screen. I have access to the Polycom phone.cfg file, and

Hello, We have updated the Wiki page for Polycom phones: http://www.voip-info.org/tiki-index.php?page=Polycom+Phones We posted several configuration specs as well as a link to an admin guide for the phone. We also posted a link on there to two firmware versions for download. The official Asterisk-Polycom support website should be up and live sometime in January. If anyone has anything to add

Does anyone know how to change the display format of caller id on the screen of a polycom 300/500/600? When I call FROM my 'shop phone 203' TO my 'office phone 201', a Polycom 500, it only says 'Shop' as the calling party. More specifically, the two lines look like this: Incoming call from: Shop I'm looking for a way to make it use both lines for caller id,

Andrei, Do you have X-Windows running on the linux box? I had a similar issue that was eliminated when I stopped this process and samba from running. Now samba is only allowed to come up during non-business hours, for changing BG music. Also, make sure your registration period in either (polycom) ipmd.cfg or sip.cfg is set to be at least the default 3600 time period. I also removed the

Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left

I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now

Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones....

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power