similar to: Dialing from phonebook, and hiding the dialed number from the user.

On Monday 26 Jan 2015, Antonio G?mez Soto wrote: > Hi, > > does anyone have a recommendation for a SIP phone, which > allows dialing from a phonebook, and hiding the dialed number > from the end users? Also from the call history of course. > > It seems Mitel can do this, and I have a use case where this is > a requirement. If I have this right, you want to make sure the

Hi, does anyone know about a plugin that allows dialling a contact from the phonebook of evolution or T-bird? -- Alexander Topolanek http://www.topolanek.at

Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server? Amaury Rodr?guez http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 Go OpenSource And Be FREE!! --------------------------------- Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze.

Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Best, Stanley

Hello list, I got a couple of those "wouldn't it be great questions", as following: 1. Is it possible, with asterisk to hold a central phonebook directory of callers?, so that when this party calls a "textual" caller ID will be displayed on the phone display. 2. How can this be configured with Trixbox, I've looked at the configuration options - I assume it plays no

So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 Log for a call from mitel heading outbound: ------------------------- -- Accepting call from '' to

Hello, sorry for the bulk question, but I cannot find answer anywhere, I have model like Phonebook::Category, and i`m stuck on calling find() method on this model ?

Hello, I wrote a small AJAX phonebook targeted at Grandstream phones although the basic functionality doesn't require a SIP-phone. Asterisk integration (call history, incoming call info, click-to-dial) is not yet implemented, but on my ToDo list. Features * Licenced under the GPL v2 * Uses gettext. So translation should be easy. Includes German and English. * Uses xajax for a

Hello, I updated my AJAX GS/* phonebook. It now actually has Asterisk integration with incoming call-notification, a searchable call history and clickable BLFs. The BLFs can monitor specific extensions (SIP/phone - may work with other channels than SIP if the AMI generates the appropriate events), Asterisk hints or connectivity to external SIP-servers. Features * Clickable BLFs (Dial,

Hi list! I'm using Asterisk 1.8.30.0 on a OpenWRT device and it works perfectly. I configured a voicemail and I receive an E-Mail with some information about the call. Again, wonderful! Now my wish: I'd like to have Asterisk to search the caller in a list file and send me the name corresponding to the number in the E-Mail of voicemail. Is it possible? I currently use ${VM_CALLERID} in

Hi all, My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote an AGI script that does the following: 1) If it's a toll free number (800|888|877|866), set the CallerID name to "TollFree Caller" 2) Use curl to look up the number in Google phonebook 3) If a business listing, set the CallerID name to business name, as is. 4) If it's a residential

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110621/764a6fa9/attachment.htm>

Dear all I have configure asterisk with 100 SIP PHONE ( SNOM ) but now thing is that my boss need phonebook feature find extention number by Pbook so i have read about it there is a feature in asterisk but it is with voicemail now i have IP SIP phone of SNOM so how to fine phone number by SIP phone ?? how to asterisk directory work ? Rgd satish patel

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

I have a small government department that wants me to implement a Asterisk installation, however, they connect to the Government PBX, a Mitel SX200, and want to keep the ability to do that. I know there is no chance to connect the digital extension lines, but would it be possible to have the pbx admins send analogue extensions over and have those lines interface through an FXO interface? Or

I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was