similar to: auto dialer

Group I'm having some trouble with asterisk and the page cmd. Any help would be great! This is what's in my extensions.conf exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0) exten => _**2,2,Page(SIP/36651)|d exten => _**2,3,Hangup CLI output ******************** Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid =

Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks [CC] app_cepstral.c -> app_cepstral.o In file included from /usr/src/asterisk/include/asterisk/linkedlists.h:23, from

I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give.

I search Google to find how to get the message light to flash on my Cisco 7960 running (Application Load ID POS3-06-3-00) (Boot Load ID PC03M030) (DSP Load ID PS03AT38) All I see is about the sip.conf file witch mine has the mailbox=XXXX but still no light. Also the messages button does not work. Any ideas?

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

I have created the following stuff: BRANDS: Brand Name Language Published Number DID Inc Markup (in 1/100 of 1%) Gamma en 5000 6 0 TRUNKS: Trunk Name Technology Peer/Trunk Gamma SIP 200.68.XXX.XXX ROUTES: Pattern Comment Trunks Connect Fee Inc. Seconds Cost per additional minute 63 Parana Gamma 0 60 500 CARDS: I have created some custom numbered cards. So, every call started with

I'm getting Number but when I look at the CDR database. I do see the name -----Original Message----- From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you

Group I'm using realtime for voicemail the it works great.. The only problem I have is I'm not able to use directory or vmail.cgi Does anyone have a solution for this problem? Asterisk CVS-HEAD-01/24/05-07:36:37 RedHat 9.0 Any help would be great!!!! Thanks

I'm looking to set up asterisk to call customer 3 days before the app and remind them we will be out to see them. I'm looking for any ideas on good ways to do this. Also I think it would be best to do some type of text to speech however I do not like the sound of the free one . Any ideas? Thanks!!! Eric Hall -------------- next part -------------- An HTML attachment was

D Not sure why this works exten => _3665[0-9],1,goto(test|${EXTEN}|1) but this does not. exten => _366[50-59],1,goto(test|${EXTEN}|1) I would like to route 36650 - 36700 to a Context 'test' however I'm only able to get 10 to work at a time. Any ideas? Any help would be great! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello group I just installed ASTCC and it was VERY easy to get running. I have a question about Pattern Via the web page I click the Routes link and everything makes sense to me but the pattern part. I tried _NXXNXXXXXX with the idea that everything would match this. Well it doesn't work... Does anyone have a good how-to? Thanks for all your help!!

Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI> -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1

Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

Hello All, I have a requirement to setup a predictive dialer for a customers call center. I am asking for pros and cons of the different dialers available for Asterisk. If you are going to send marketing material send it to my e-mail directly please and not to the list. I was hoping to get the opinions of any one using any of these dialers and what they liked and didn't like, ease of

I am trying to switch over to the new "style" of printing, but when I do, I loose the ability to print duplex. I have two servers. The old one (spike) is setup as a file server, etc. and the print services date back to the early 2.0 days. It is currently running 2.2.2 with the original printer setup. The printing works ok with this server. People can print using features of the

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

Hello, We would like to share our webinar about Auto-Dialer Business Model (Telemarketing). It is educational video which we made for our clients and now we are sharing it with you. http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/ Enjoy. NOTE: This is not attempt to sell you anything. No product or service is sold/marketed in the video. Regards, Kolmisoft Team

hi wich predicitive dialer are you using and wich one do you recomend? a link to the project/product and a link to a how to will be VERY apreciated. Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all- I have asterisk running on 2 systems, with four E1 spans each. Each system is connected to a (big) NT DMS-100 switch. For load testing an IVR system running on one of the asterisk systems, I'd like to use the other system to generate a lot of outbound calls under program control - on most or all of its channels simultaneously. All of the asterisk dialplan and agi programming