similar to: Which parameters of a live Asterisk server would you monitor ?

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090627/37b93684/attachment.htm

Hi, I feel I've read a thread about this previously but I couldn't find it. Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. So that, you could

Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/fce6307c/attachment.htm

Hi, I would like to tune rx/tx gains using dahdi-monitor for a system which will be connected to french PSTN. I'm not aware of any public phone number in France I could call to get a "normalized 1004Hz signal". My questions are : 1. Does such numbers exist ? Is there a directory somewhere listing some of them ? Do you think regulations could make providing such numbers mandatory

Hi, I need to replace digital handsets in offices where there cabling is appareantly not Ethernet-compliant. Today's usage is to press a key to toggle between private ou public line before issuing an outgoing call. Are you aware of a DECT handset (to overcome cabling limitations) that mimic this line-key behaviour ? For instance, acceptable behaviours would be to dial number string and press

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi,

Hi, Watching Snom 870's video (http://www.youtube.com/watch?v=9e8hPxX0oDU), you can see a new sidecar (phone extension) which seem very interesting. Has someone details on this extension ? Any release date or/and data sheet ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This probably wouldn't be useful for users in USA, Canada or Hong Kong as costs to call a mobile is

Hi, I'm using asterisk 1.6.1 and AEL2. I'm trying to find the best way to write my own custom functions ? At the moment, I'm using this pattern (extensions.ael) : context foo { 123 => { &myfunc(123456); NoOp(${GOSUB_RETVAL}); }; macro myfunc (arg) { Return (${arg}); } 1. First, I keep getting warnings like Warning: file /etc/asterisk/extensions.ael, line

Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081127/b41ca08b/attachment.htm

Hi, I would like to originate my first call from CLI. As I'm new to this, I'm wondering if it's possible. When I type "originate" from CLI, I've got this : " There are two ways to use this command. A call can be originated between a channel and a specific application, or between a channel and an extension in the dialplan. This is similar to call files or the

Hi, I'm bumping on this : cd /usr/src wget http://www.misdn.org/downloads/mISDN.tar.gz tar xvf mISDN.tar.gz cd mISDN-1_1_18 make <snip> In file included from /usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/core.h:9, from /usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/avm_fritz.c:20: /usr/src/mISDN-1_1_8/include/linux/mISDNif.h:791: error: field

Hi, Which is your favorite desktop wifi sip hardphone ? I'm looking for something like http://www.mitel.com/DocController?documentId=19401 which could be easily moved from one meeting room to another. (In this specific case, finding an electrical plug to power a large desktop phone is seen more relevant than finding an PoE Ethernet plug or using a mobile handset.) Which product would you