similar to: Polycom SIP Phone config files

Hello, My Cisco phone has software: Boot Load: PC030300 Ver: 3.2(7.0) And I want to upgrade it to SIP 6.0 Is it possible or I have to upgrade to ealier then 6.0 and then to 6.0 ? bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031121/7331dff5/attachment.htm

Hi all, As promise, the new prerelease (0.9.2) is now available for download from the followiing locations: http://www.laser.com/dante or http://www.geocities.com/tdanro A detailed help file is available online and in the application package as a chm file, accessible from the app help menu too. Unfortunately the IAX2 support is not ready yet, but I work on it now (next on my list). The DLL used

Hi all, A first basic version of DIAX as an ActiveX can be downloaded from: http://www.laser.com/dante/diax/activediax.zip There is only one small file (diax.ocx) and a readme.txt with the usage instructions. For the moment you can only place authenticated (or not) calls and there is no feedback (ring, messages, etc) Put this simple thing on your web page and you will be able to dial from any

I'm looking to acquire another FXS module -- either the TDM400P daughter module or USB S100U is fine. I'll happily buy from Digium, but thought I would give the folks here an opportunity to recycle any extra hardware :-). Please contact me if you have one to sell. Thanks! -- David

Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk

What happened to "sip show peers" in 1.4.13? Jerry

Hi all, i never understood that why is there 2 branches of asterisk going on parallel. asterisk 1.2.* and asterisk 1.4.*, i also heard about beginning of another branch which will be 1.6.*. so whats the difference between these 2 or 3 versions, can anybody plz tel me? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to test the call parking, but It doesn't fonction :(these are my config files.extensions.conf:include=>parkedcallsexten => 4000,1,Dial(SIP/4000,60,tT)exten => 4001,1,Dial(SIP/4001,60,tT)exten => 4002,1,Dial(SIP/4002,60,tT)In features.conf:[general]parkext => 700 parkpos => 701-720 context => parkedcalls [featuremap]blindxfer => # disconnect => *0

Sorry - this is a bit off topic, but there is almost certainly someone here who will know the answer... Perhaps even a snom employee :) In recent snom firmware releases, the following sequence always causes a call to be sent from line 'n' Receive call on Line 'n' (where n > 1) Press Hold Dial new call The new call originates on Line 'n' In older firmware

Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the

I have 2 sip phones and an analog phone attached to a Digium USB fxs device. I would like the analog phone to ring when transfers are made to it, but I don't want it to ring when a call comes in from outside, although I would like the person at that phone to be able to pick up the phone and answer the incoming call. Is that possible? Thanks Sean Garland, MCP+I, A+ Siskiyou Technology

Hello, Today we setted up a server that needs to use MeetMe but doesn't have any Zap hardware. So we need to use ztdummy (at least, this was our idea). Rarely: zttest is not working at all (100% bad, using zttest -v doesn't give anything, etc.). Of course, after load ztdummy, there isn't any background or anything. It is the same kernel (Debian Etch default kernel, 2.6.18) than

We have an installation with around 50 sip phones but only 5 of those are hardware. There are three Grandstream phones, one Snom and one PAP2T. We are running Asterisk 1.2.8 with an E1 (R2). Only the hard phones are having problems which are either echo or distortion. The softphones all work fine and no one is reporting any problems. They are using 3Com switches which are fairly new. I

Hello, I have finally received the details from Polycom to get into the backend configuration of their SoundPoint IP 600 SIP VOIP phone. The phone is quite nice looking but the configs are very sparse, not even a place for a secret(password) field in their SIP registration section. If anyone else has one of these and needs the passwords to get into the back end configurations, just send me an

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com

I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work?

I have a soundpoint 501 phone that has locked up twice now. You can make a call but when a call is sent to it, it responds with sip busy messages. You get the same message when the phone is in do not disturb. I reset to defaults the first time and it worked for a week or so and then stopped. The incoming calls are ringing three phones ( dial(sip/1&sip/2&sip/3 ), often two of them are in do