similar to: Asterisk billing from CDR database

Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip-info.org/wiki/view/Asterisk+cdr+pgsql What I want to know is how do the billing solutions (like Asterisk2Billing) work with Asterisk. The billing system just use the information that Asterisk puts in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command,

Does anyone have a script that will notice a Rumplestiltskin type spam attack (where they try every name possible) and drop the sending into a block list? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com -- This message has been scanned for viruses and

I have a Astlinux system that can't run mysql. I want to bill calls by extension but all I have is the raw CDR files. Is there any software that cal produce a simple call accounting from text CDR files? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com --

I have two feeds, one with a static IP and one with a dynamic IP. How can I configure a Multi-ISP setup with the dynamic IP, or can I? I don''t think the gateway will change, just the interface IP. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com --

I have placed the custom-cdr-V1.0.tar for download http://www.itsngroup.com/software/asterisk/downloads/ Thanks > Dear Ben, > I've also the problems as Chris Mason, Can you send us your own AGI CDR > logging application? > Best regards, > Jian Hong Guan > France > www.directcentrex.com > > >

We have a billing system that depends on the CDRs. We had a guest that made a one minute call to a local cellphone, this call went out Zap channel through our channel bank. The CDR recorded a 200 minute call, but I checked with the Telco's records and it had terminated after one minute. What can cause this and what can I do to prevent it? -- Chris Mason NetConcepts (264) 497-5670 Fax:

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband

I have two external interfaces in a Multi-ISP config. I allow access to port 81 for a webcam, but I only want that to work for one of the interfaces, and I want to limit the connections to it by maximum time for one user, or failing that, maximum connections, as people just leave it running on their desk all day (it''s a Caribbean beach so people sit and dream). ow do I do that as

A client wants to put phones in a semi-public place, using a Calling Card solution. What kind of hardware is suitable? I'm looking for a wall mounted booth, a SIP phone that can't easily be broken, but I might just use cheap analog phones and a channel bank. What do you suggest for the calling card software? This installation will be outside the US. -- Chris Mason NetConcepts (264)

Is it possible to filter HTTP signatures/headers with SHOREWALL ? or is there addon for it ? take care *º¤., ¸¸,.¤º*¨¨¨*¤ Stingray *º¤., ¸¸,.¤º*¨¨*¤ __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ------------------------------------------------------- This SF.Net email

If I use a call file (/var/spool/asterisk/outgoing) is it possible to have a video phone connect to a camera (linksys wvc200) and show the camera stream and hear the audio? How would I do that? Jerry

I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com

For our hotel application, we don't want to have to write 50 voicemail entries, is there a way to do a voicemail macro in the same way as a standard extension macro? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com

I have some IP 500s that I bought used, but the connectors are different than the new ones. There is a Modem/Power RJ11, a Line RJ11, and Handset and headset connector. Does anyone know how they work? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com

I have a sample of the Yoda VG400 and I am having a devil of a time trying to get all four channels to register to Asterisk. I have an Asterisk 1.2.1 server. I have tried adding one at a time and rebooting it, but it stops after the first. http://www.yoda.com.tw/model.php?type=Enterprise_VoIP&pname=VG400 Anyone had success with this? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264)

One might also conclude that during the outage the support people were focusing on getting the system back up and were not near phones. At least that is what I would bet on. Just a thought considering how most of the smaller ITSPs seem to work. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

I have two issues... First I am working with a hotel software vendor to include an automated way to turn vmail on and off while clearing it at the same time. The vendor is looking to interface via serial cable as they currently do with Mitel systems. i am willling to work with them on an IP interface but I am not so sure on how to implement it in asterisk. Does anyone know of a way that may be

Hi all, is there anyway i can set SIP_HEADER(To) to the value i like? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070409/528077f9/attachment.htm

What is the function of the snare daemon? I just found it on a machine and never heard of it before. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla at yahoo.com -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be

I am trying to use ChanIsAvail to detect the best route for a call. I am testing by dialing an extension that is then forwarded to the DID. Normally it will be an incoming PSTN call that is forwarded. When I try it, I get put on hold for a few seconds and miss the beginning of the recorded message. Any ideas what is going on? -- Executing ChanIsAvail("SIP/501-304d",