similar to: Asterisk and Eyebeam

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)

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:

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

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

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 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 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

Sorry, I don't remember who was asking about this, but it seems that if you record a video message that contains the send video start, it will actually fire up the remote receive window. I.E. Previously I was using the recording section of voicemail to create my video IVR's. This meant that when I arrived at the section to record the message, I had already clicked send video in Xten.

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 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 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 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

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

What does the Zombie mean in this line from the CLI? == Spawn extension (daymenu, s, 1) exited non-zero on 'SIP/700-fe13<ZOMBIE>' 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