similar to: Polycom 601 stops ringing

Any great disadvantage to using polycom 300/500/600 vs the 301/501/601? joe a.

My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a.

Hi All, I am a remote office that is connected to my office via openvpn on UDP. Voip has always worked well (after discovering g729). Initially I used a softphone, then an analog set on a sipura 2000, then a polycom IP500 (I still LOVE this phone). At that point, I started noticing that the polycom doesn't ring a lot of the time. Since I was desperate for a phone, I didn't upgrade

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin

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.

Getting "no service" display on aastra 480i. Sip debug shows an "unathorized" blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out.

I have a setup with a polycom 601 and an act p160s. All on local segment, no NAT. Can call the act p160s, from the polycom, rings, connects, and a conversation can take place. The reverse is not true, Dialing from the act to the polycom does not work. SIP debug shows, at the end, "Incoming call: got sip response 416 "unsupported URI Scheme" back from 192.168.0.xxx. Which is

Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone call to a specific number and make an announcement? I imagine the first part is the big question. joe a.

Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting. Using a TDM400p, with FXO and FXS modules. On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a "hook flash", to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can

I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works

This is a follow up to an earlier post. Looking for a means to "individualize" incoming FAX, so as to distribute them to the intended recipient. While the PBX is based on Asterisk, it is not possible for me to "enter" the box to modify things, to any great degree. I thank those who mentioned IAXMODEM, earlier, but that seems a no go. Currently, there is a dedicated T1 into

Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in

has anyone done it? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080528/ff622d28/attachment-0001.htm

Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a "Page". From what I understand, the "Page" is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller

Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying "Could not contact boot server, using existing configuration". I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. -------------- next part -------------- An HTML attachment

I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs) As I understand it, I must have faxdetect = incoming to enable detection of the fax tone. Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on. In my case, I ask it to answer immediately and do a distinctive ring (r3) to alert that is its a FAX call so no one picks

JR Richardson Engineering for the Masses> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of asterisk-users- > request at lists.digium.com > Sent: Saturday, March 01, 2008 12:00 PM > To: asterisk-users at lists.digium.com > Subject: asterisk-users Digest, Vol 44, Issue 1 > >

Hi List, I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times

Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio