similar to: Netowrk addresses contactor

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:

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)

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

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 am copying the Master.csv file to another server and importing to mysql. I am looking for a simple billing application that will produce a bill for a give account code for a give period, based on a rate table. Is this available? -- 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:

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

For all my PBX installations I want to have Fail Over on the main incoming PSTN line so that a power outage does not leave the offices stranded. Is there any commercial solution to this? I would rather a finished product than a home soldering project. Chris Mason masonc@masonc.com Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483

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

I have installed Asterisk on a CentOS4 box and then installed Asterisk from CVS. I installed a Sangoma A101 and connected it to a Adtran 600 using a T1 Crossover cable. The 600 has 12 x FXS, 12 x FXO interfaces. I ran through the wanpipe install instructions and configured it, now I can run [root@altpbx asterisk]# wanrouter hwprobe ------------------------------- | Wanpipe Hardware Probe Info

I am having a mail server problem where the mail to one domain is being sent to the wrong server. When I use "dig masonc.com mx", I get ;; ANSWER SECTION: masonc.com. 86400 IN MX 5 mail.masonc.com. ;; AUTHORITY SECTION: masonc.com. 86400 IN NS ns1.picadoo.com. masonc.com. 86400 IN NS ns2.picadoo.com. ;;

I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp("IAX2/netconcepts@nnn.nnn.nnn.nnn:4569-5", ""Dial 710"") in new stack -- Executing

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)

I have tidied up the script and added some help text, feel free to download and maybe improve. http://www.masonc.com/phoneconf Usage: Usage: ./phoneconf [config|help] phonemacaddress extention username context ./phoneconf help will print syntax info ./phoneconf 0004f201aa11 500 MyPhone defaulot will configure the phone with that mac with one extension (500), and add the phone to sip.conf.

I am using AstLinux with Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by root@ls on a i686 running Linux On this box I am registered to two different providers for long distance and international. If there is an internet outage of more than a few minutes, I'm not sure how long it takes to make this happen, the registration times out and when internet comes back the pbx never

I have a server running Centos4. I found I was seeing some network problems so I tried "route" and found there was no routing. I thought reboot would solve it but it didn't. I had to manually enter a route to the default gateway, there is still no route to the loopback interface. How is this possible? I thought there was default routes built into the kernel, taken from

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 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 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

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