search for: 601s

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20060518/8...

Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the "latest publicly available" from Polycom's web site). I've noticed that some keys get "sticky" though. Soft buttons for example (i.e. "end call") need to be pressed 2-3 times for them to react. I've downgraded to

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

I posted a week ago to see if anyone could share a good, or great, dialplan for a small business who wants to make good use of some polycom 601s. I received a pointer to some the loligo dialplan, which is nice, but doesn't help me with the polycoms. And nothing else, except a number of "please share if you find one" requests. Is nobody willing to share? Thanks- Tom

If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all goes well until the second time I hit forward (to join the caller with the extension); then, the caller's MoH goes away (making them think they've been hung up on), and the server spits out: asterisk-cw*CLI> <-- SIP read from 10.20.2.16:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP

..._sip.c: SIP: Received packet with bad UDP checksum Any advice on what I can do to get more information? I see it occur when Asterisk doesn't think there are any active channels going on as well. There's a SIP device that is misbehaving somewhere. Most of the SIP phones are Polycom IP600/601s. --johann

I am having a problem with some Polycom 601 phones. If I dial without picking up the handset or selecting the speaker I can dial numbers that are any lenght. But if I pick up the handset or are using the speaker I can only dial numbers that are 8 digits. When I dial the 8th digit it dials immediately. Obviously this creates problems when I am dialing long distance numbers or anything that

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....

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?

I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I

For the last few months we have intermittently been experiencing some very strange registration problems with certain polycom phones. Here is some background information: I have about 150 Polycom Soundpoint IP 600s, 601s, and 650s spread between 8 servers at different locations. Each phone is on the same network (and subnet) as the server it connects to. There is no NAT or anything else strange that should be messing with the connection between the phones and asterisk. We are using Netgear FSM7328P and FSM7352 P...

I tried to add a couple of SIP phones (polycom 601s) to my existing asterisk installation. I can successfully make a call from the SIP phone to any other phone (inside or outside), but I can not make any calls to a SIP phone. Attached are the pertinent parts of sip.conf and extensions.conf. The log starts off normal with: Mar 26 09:51:15 DEBUG[...

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >