similar to: Asterisk 1.2.18 and Polycom phones not forwarding anymore

Hey, all. I've got an office set up with Asterisk, and forwarding's got a bit of a glitch: When they forward, they listen for the remote phone to ring, then hang up. If the remote phone doesn't connect, it goes to the original phone's VM. Is this Polycom's "fault," or Asterisk's? I've been reading up on blind/supervised forwards, and, honestly, have

We are having an issue that I have been unable to figure out how to resolve. I think its related to the Polycom Phones and not the Asterisk configuration, but I'm not positive. We have several Polycom 500/501/601's on both a LAN and at employee homes. The problem we are having is if our internet connection goes down the Local LAN phones loose their connection to the Asterisk Server.

Hi, I know that if you have Polycom phones properly configured, you can use "sip notify polycom-check-cfg SIP_REGISTRATION_ID" to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike -------------- next part

Hello, A few days ago I've posted two questions about Polycom phones: How to access corporate phone directory from the phone and how to use a conference server with it. After I got zero responses I tried openning a support call in Polycom's site. Here are the replies I got from them: - Corporate directory: They are thinking about it, probably will use LDAP. I was asked to open a

i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/

> >> Jason Brown wrote: > >> | Anyone have experiece with polycom phones? > >> | > >> | I am experiencing a really weird problem. In an office > where I have > >> | the following extensions: > >> | On the Polycom phones, when I want to dial from extension > >> 100 to any > >> | extension 120 or above, or dial out, it

Hi All: I have experienced some big problems on an asterisk production server under 1.2.18: First of all, a very rare message like this... No application Macro ??? -- Saved useragent "Linksys/SPA922-5.1.7" for peer 1363 Jun 6 15:08:24 WARNING[406]: pbx.c:1720 pbx_extension_helper: No application 'Macro' for extension (pbx-incoming, 1133, 1) == Spawn extension

Here is what I had to change on the phone1.cfg file: I had this value in my 1.6.7 file, put in there following suggestions from the Wiki (http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501) : reg.1.server.1.expires="30" Now, this worked flawlessly with 1.6.7. But with 2.x, this seizes up the phone with a huge CPU load (approaching 100% at times) and makes it

Hi Floks, This is probably really dumb but here goes: I used to be able to place calls to my SIP phones from the CLI using the 'Dial' command for testing. I have installed asterisk on a new machine and copied over the .confs and started it up. It all works fine. But when I try to initiate a call from the CLI using 'Dial' it just says: *CLI> Dial No such command

Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? Mike _____ From: Mike [mailto:list@virtutel.ca] Sent: Wednesday, April 11, 2007 13:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Polycom 501 issue with latest

Hi, I`m having a problem I cannot explain. When dialing 555-555-5555 (for example), I get a ringing sound until the person answers. When I have my Polycom forwarded to 555-555-5555, I do not get the ringing, but it dials fine and eventually when the person answers everything works fine. Where could be the difference? Both are using the same context to dial out. Mike --------------

Greetings. Has anyone made this work with BootROM 2.6.2 and app 1.5.2? I've tried sending DHCP options 128, 144, 157 and 191 containing a single digit (the VLAN ID) with the phone's 'Fixed' setting for DHCP VLAN discovery. Different DHCP data types don't seem to help, as I've tested with raw bytes, ASCII and 16-bit unsigned ints to no avail. Setting the phone's

We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP

Hi - Just wondering if anyone has gotten a Polycom VVX phone to successfully do an Auto Answer with asterisk. I have an older generation of Polycom phones that do this just fine, but I can't seem to make the VVX phones work. I tried the guide here: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 And I have this in my diaplan:

I have meetme working with BLF on polycom phones however when meetme is not actually being used by anyone the 'status' of meetme becomes "idle". Which the Polycom phone sees and produces a clock symbol and FLASHING red LED. Are there any 'tricks' or work-arounds to change this status to something that does not blink the phone's LED making it look busy when meetme is

Hi, I'm in a hosted PBX context. I'd like to push some strings ("Hello World") from Asterisk to a Polycom phone's screen that is behind a NAT/firewall. Basically as part of the SIP if possible, to make it firewall friendly. Is this even possible? Part 2: Is this possible as part of a queue called (when I am not using the dial command directly)? Mike

Hi, Where should I go to get the Polycom`s latest official (non-beta) version? I am registered on the Polycom customer website but that doesn't seem accessible. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061102/284b67a7/attachment.htm

Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I know having Asterisk aware of Polycom "Do No Disturb" state wasn't working before (1.4), but is this working in any recent version? Is there any "custom" way of doing this? Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone