similar to: SIP

Hello, For each incoming or outgoing call, sip hardphones I'm using, turn BLF on and off like this: the first call (after leaving idle status) turns 1st BLF on, the second one turns 2nd BLF and so on, when a call is hanged, its BLF is turn off. My first question is : do you think such behaviour is general ? My 2nd question is : using AMI, how can I tell for a given extension : 1. the number

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

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead

Hello List, I have been working on a PHP application in order to build a BLF style script. Until now everything is going Ok but something a little (in my oppinion) strange is going on with the 'ExtensionState' command; The problem is that it does not returns the 'Status' as it's suposed to, mentioned in the A.T.F.O.T book - version 2., where it sais something like:

Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for

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:

This is my first attempt to write software of any sort. What I am trying to is to use a .php page to query asterisk Manager and get the ExtensionState for each particular extension. Then when it has the answer it outputs an XML file for use as the directory on a Cisco 7960 phone. What I am thinking is that when the user hits the directory button to veiw the directory that is at this URL it will

Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/9123 at local B. AMI/FOP: 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123

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

Hi, 1. How do you then, synced then unread message presence with custom device status ? From an external program ? When a user leaves VoiceMailMan application ? Using externnotify ? 2. What is MWI:101 at default expression for (see [2] ? Cheers [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at

Hi, I'm banging my head over this. Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to enhance BLF with Directed Call Pickup : basically, SIP hardphone (here a Thomson ST2030S) is configured to send an INVITE message whenever a BLF is pressed while blinking. The INVITE is build with the extension number (attached to the BLF that was blinking and pressed)

Replying to myself, I've just read in 1.6.1 announcement that a new Incomplete dialplan application is the one that provides what I'm looking for ... 2008/10/3 Olivier <oza-4h07 at myamail.com> > Hi, > > If my memory serves me right, there was thread (in dev mailing list ?) > explaining how we could implement 2 stages dialing with SIP endpoints: > user dials 1234

Hello, I'm trying to supervise an existing Voicemail box with a BLF button on Debian's asterisk 13.14.1 system. I mostly found this [1] document. I added in a context a line like: exten = *7000,hint,MWI:31 at default With "core show hints", I can read this: *7000 at subs : MWI:31 at default State:Unavailable Presence:not_set Watchers 1 My questions

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

I have a function *1 that starts and stops recording in a call. I use a function so I can use MixMonitor. It works well, however I would like to make it a little more integrated for my users. We have GXP 200 hardphones. So far I've been able to configure a softkey using the speeddial option to dial *1 during a call. I also have setup another key to monitor the status of recording (on/off)

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, I need a hack to query current calls coming in and going out an Asterisk 1.6.1 system and send this list back as a custom UserEvent to other listening endpoints. For various reasons, I might need to write this hack in PHP though I'm more experienced with Asterisk Java. What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ? I'm referring here to

Hi, In ASTDB, I've got a rather long list of entries like: /FamilyA/Key1 Value1 /FamilyA/Key2 Value2 /FamilyA/Key3 Value3 ... Instead of sending several DBGet queries (and parsing every response), I'm wondering if a single "database show" or "database show family" query could be implemented. Alternative if to use ssh ("asterisk -rx "database show

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

I have two calls (626 and 542) coming into the same phone (524). SIP/524-000005b5!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!9!SIP/542-000005b4 SIP/542-000005b4!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!542!!3!9!SIP/524-000005b5 SIP/524-000005b3!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!40!SIP/526-000005b2