similar to: CRM integration (was RE: CallerID)

> > > >| -----Original Message----- > >| From: asterisk-users-bounces@lists.digium.com > >| [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > >| Michiel van Baak > >| Sent: Martes, 24 de Mayo de 2005 04:45 p.m. > >| To: asterisk-users@lists.digium.com > >| Subject: Re: [Asterisk-Users] CallerID > >| > >| On 23:44, Tue 24

Hello, We use astGUIclient, it does have server side apps that have to be installed on your Asterisk server, but it does have callerID popups that allow you to search a customizable web page when a call comes in. We are also releasing a new version of the astGUIclient app next week that is entirely web-based and easier to configure the client side. http://astguiclient.sf.net/ MATT---

hello everyone, I just had a thought on this subject why not create a daemon process on the Client PC That registers its self and What phone the user is connected. An AGI script could monitor the progress and when answered could send a push to the registered daemon which would push a link to the registered daemon on the telephone operators on the desk top. this would not waist resource as much as

I am also very interested in CRM integration. Anything I can do to help? One thing I don't understand is how is the browser being launched on the person's PC. Or is it not launched automatically? Anyone know of a simple app running on the desktop to do this? I looked into IPSwithcBoard and it appears like it should be able to do the job, but: a) It is pretty heavy - it does a lot of

The method we use for web popups on incoming calls in the astGUIclient client app that we are working on for release next week is to use AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't require any META refreshes. We've been using this internally for the last month and it works great. MATT--- -----Original Message----- From: Anton Krall

Hi all, One of the areas I am trying to research before I can confidently start deploying Asterisk is "Credit Card Machines". (PDQ / Streamline machines / any similar) I'm talking about the credit card swipe boxes at point of sale desks. I believe they dial out to the specific bank provider everytime a card is swiped. My question is: - Does anyone have any experience using

Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together. Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/e71c34a3/attachment.htm

FYI - We have a solution here provided by Lucent that allows us to play / review voicemails left on the Octel attached to the 5ESS switch... While this is a simple webpage - doing a refresh every 3 to 4 seconds, it does actually work. The only loss of course would be if someone hung up on first ring - the browser might not catch it in time. -----Original Message----- From:

Has anyone any experience of the above. Key feature for me is tracking incoming and outgoing emails and linking them to the contact record. Thanks, sorry for the OT ;-)

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hi all, I've been searching the wiki and google for a couple of days now but cannot find any reference to a timing source on OpenBSD. I have * CVS-v1-0-02/15/05-21:54:52 (I always do a cvs -q up -Pd before compiling) running like a charm on OpenBSD 3.6 Now I want to setup some IAX trunks to work and 3 friends and some meetme rooms but it looks like I need a zaptel timing source. Anyone can

Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I

Hi all, My BRIstuffed 0.2.0-RC7k is running fine on my debian box for voice calls over ISDN2. Now I want to implement receiving incoming faxes into my setup so I did a google and some reading on the wiki. I got the spandsp 0.0.2pre10 package compiled and installed, patched asterisk's apps makefile and compiled * again. This all worked out fine. When integrating the RxFax into my dialplan the

Hi all, I been searching on the wiki and google for ENUM in NL. All I could find were some docs from the Dutch Financial Department about taskforces and plans. But it all links to dead pages and no-longer-connected phone numbers. Is there anyone who knows some more about Dutch ENUM stuff ? If it's all dead, anyone interested in setting it up together with me ? Hope to get some answers,

MD -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michiel van Baak Sent: September 12, 2008 2:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] Setup speed dials on Cisco 7921 On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: > I've added lines like this: > > > >

Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

Hi all, I got a couple of Snom 190's through this week and after some initial foolishness I have them both setup no problems. But here comes the except. When there is voicemail waiting the softbutton appears but the phone always dials its own extension. No matter what I put into the "mailbox" parameter on the line settings. (Phone registers correctly with * and all standard

Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to

Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak michiel at vanbaak.eu http://michiel.vanbaak.eu GnuPG key:

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been