similar to: `hint` priority and Polycom 500

Hi all, I'm researching IP phones for a new office setup. We will need 30 phones. I have read the wiki and the polycom site for the phones, but I still had some questions about real world experience with these phones. -According to the documentation, the 500 series ( and 600, according to a polycom rep ) have built in hubs. Has anybody noticed performance issues in this setup, when

Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off

Hey all, First of all, thank you for the help I've gotten on this list in the past. Very helpful, and I apprecaite it. Now, what I'd like to do is send a message to my snom 320s. I'd like to have the message display regardless of what the phone is doing. I have been trying SMS, or the sipsak method on the wiki but I have had no luck thus far. Does anybody have this working,

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, after a little head banging, I got winbind chatting with the domain control ( win2k server ) by hard coding my realm info into krb5.conf. My question is this: Is there a cleaner, more fault tolerant method than this? I googled, and read about dns srv records being used ( intead of hardcoding ), which I understand win2k uses to begin with,

HI, I am trying to setup winbind on Samba 3.0.2 running on Red Hat AS 3.0. I have completed most of the steps of setting up winbind successfully but when it came for me to login in using the AD account username and password, it didn't allow me to login. the error message i am getting is incorrect password or check username. During the setup i tested the wbinfo -u command and i was

Hi folks, For the longest time, I've had a problem changing or modifying ACLs from my window clients. Whenever I tried, I'd get this in the logs: [2004/07/29 12:36:26, 0] smbd/posix_acls.c:create_canon_ace_lists(823) create_canon_ace_lists: unable to map SID S-1-5-21-1292428093-651377827-xxxxxxxxx-1333 to uid or gid. I could change the ACLs using getfacl/setfacl, btw. After a

Are phone numbers portable in other countries? Are the same rules and conditions that exist here in the States mirrored elsewhere? How does a person in Europe go fully VoIP and still keep the main number? Do they use call forwarding? Is their another way to use an origination carrier without loosing your number? Alex -------------- next part -------------- An HTML

Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena

Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get "Push message cannot be displayed" back from the Polycom phone, and all I am sending is the Polycom example : <PolycomIPPhone> <Data priority=?critical?> <h1> Fire Drill at 2pm </h1>

...even when I do the "factory reset" (4-6-8-* then 456). I tried using FTP and TFTP, but even though the phone uploads the log, I get these errors: 0927211350|app1 |3|00|Time has been set from 0.us.pool.ntp.org(69.60.124.59). 0927211350|cfg |4|00|Could not get all 512 bytes of the header. 0927211351|cfg |4|00|Could not get all 512 bytes of the header. 0927211422|app1 |4|00|Loaded

Hi all, Sorry if this has been asked, but I haven't had any luck with my searches, so I would assume it hasn't. I have joined my samba box to my AD domain ( win2k server ). I can do individual user lookups with wbinfo -a user%pass successfully. But I can't retrieve a domain user list with `wbinfo -u`. `wbinfo -g` works, sorta, but it only returns the BUILTIN accounts ( System

Hello, At one of our locations, we have started to see Polycom 501s (running 1.6.7 firmware) randomly reboot. We have taken packet traces of the phones to determine if there is something odd in the Layer 2 or 3 of the network that might cause it, and have not seen anything strange. There are no errors on the ports. This appears to be affecting POE powered as well as AC powered phones. The Polycom

Two * servers: *a and *b. Outside call comes in *b, and is automatically routed to *a. Someone on a sip phone connected to *a then decides to transfer the call to someone on a sip phone connected to *b. The transfer works. At this point, is *a still in the converstation? Or is * smart enough to see where the data stream is going/coming from? Thanks for any help in advanced, and sorry if

Hi all, I've got an old avaya partner acs <7.0 system here. I'd like to add a simple voip bridge so I can hook up our remote offices. From my research, it would seem the pre-7.0 series doesn't have a t1 port, so if I wanted to do this, I would have to feed the avaya system fxs ports from the asterisk box. Does that sound about right? Has anybody ever done this? Does

Astricon 2004 is the first Asterisk user's and developer's conference, to be held in Atlanta, Georgia in September. See http://www.astricon.net ** We will soon open for early bird registrations! ** To get a very low price, I recommend that you participate as a speaker. In fact, speakers doesn't pay any fee at all to participate in Astricon. **** Send us your speaker's proposal

We're now opening up registrations for the Astricon tutorials again. We've been able to move to new conference rooms within the same hotel. Register on line at http://www.astricon.net We're sorry for the inconvienience our recent closing of the tutorials may have caused you. You are welcome to contact us at info@astricon.net to change your reservation to include the tutorials. There

I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns

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