similar to: Voicemail systems- flow charts, digit/key cards, etc

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin

Thank you all for the voicemail cards you sent. If you have the following in PDF or laying around (scan): * AT&T/Cingular flow voicemail card * Verizon flow voicemail card * Sprint flow voicemail card * TMobile flow voicemail card * Alltel flow voicemail card * Avaya Nortel Octel flow voicemail card * Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one I will work on

I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to throw it off, so I still have a bit of work to do. Have your friends and significant others

Is the new NIN Ghosts music (free download) safe for MOH? Justin ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks for the feedback. I did some research and it looks like you were calling over international lines. It also appears that there was high than average static on the line, which is not normal for my system. It's true that I threw my recordings together quickly and the beep was supposed to be funny - it was actually me saying "beeeeeep". However, the static and noise you received

Does anyone have use for a broadcast/annouce app? I wrote SystemAnnounce which will play a specified file to all active channels (in an UP or bridged state). This was originally to tell users to get off the system, but there are several other uses... I also wrote a new CallPickup and CallPark app, both of which work more as expected (supply actual extension numbers, etc). Let me know if there

This system targets a different market... I like Olle's system. He did a good job. Olle's minivm is a great choice for those wishing to build customized voicemail systems, but as the name suggests, the systems are very basic. Large systems are difficult to maintain in the dial plan and some of the functionality we need would be difficult to implement with that approach. Justin -----

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

I have a problem of using call file to make an auto dial out call through FXO channel. I defined the channel in the call file as "Channel: DAHDI/1/8775203463" When I put the call file under the /var/spool/asterisk/outgoing dir it did not call out but came to the context I defined in extensions.conf as if the callee had answered the call. If I make a call from an extension to

Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? I have been diligently investigating converting our small (Ontario, Canada) office to Asterisk, and ditching our Avaya PBX. However, our head office (New Jersey, USA) maintains our AUDIX system, and a) have no intentions of leaving it and b) some users rely upon AUDIX's ability to transfer

Does anyone know if the ABIOSDSK service in XP would "find" and a HDD image pushed to ram by MEMDISK?? ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

How many of you are using Rails: 1. As the primary development tool at your Day Job 2. As a small part of your Day Job 3. Not using Rails in your Day Job, but are using Ruby 4. For OS/outside work/hobby -- sam http://www.magpiebrain.com/

Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like

Hi All; We have a telephony service provider that is asking what is new technology and services to be added with the telephony service that can be used for VoIP and PBX purposes. Any suggestion to be added that can really give new advantages and technologies specially in VoIP issues? Anyone interested? Regards Bilal

Hi all, I found similar sequences in the /var/auth.log files of freebsd boxes, I supervise.: Aug 13 13:56:08 www sshd[26091]: Illegal user test from 165.21.103.20 Aug 13 13:56:11 www sshd[26093]: Illegal user guest from 165.21.103.20 Aug 13 13:56:15 www sshd[26096]: Illegal user admin from 165.21.103.20 Aug 13 13:56:18 www sshd[26103]: Illegal user admin from 165.21.103.20 Aug 13 13:56:21 www

Anyone have an H.323 trunk tied between their Avaya and Asterisk box that works? I am having some issues trying to get the two systems to connect. I am using the ooh323 channel to try to make the connection between the two system. I have all my configs if anyone would like to look over them. If I do a trace on Avaya I get a denial event 1191: Network Failure. Thanks! -------------- next part

Hi guys, I'm interconnecting an Asterisk box with a Lucent Definity PBX by means of FXO/FXS ports on a TDM2400 card. Everything works well, except for one little thing. Every now and then somebody (from an Asterisk extension) will call another extension on the Lucent Definity PBX and they hit their voicemail. They caller leaves their message (or not) and hangup, BUT the Lucent sometimes

I''m starting a new project and I want to try writing the client first. I.e. I want to write my ''views'' first and I would like to mock out all the models in such a way that I can run the site and browse it with the mocked out models. Is it easy to use rspec''s mocking for this sort of thing and has anyone done it before? Does anyone know of a tutorial out there

Here's my attempt at a fix. Adding Jakob to make sure I did this right. On Fri, Jul 19, 2013 at 2:34 AM, Peter Newman <peter at uformia.com> wrote: > That does appear to have worked. All my tests are passing now. > > I'll hand this out to our other devs & testers and make sure it's working > for them as well (not just on my machine). > > Thank you,

Hello list, in a previous post I mentioned I had managed to make USB-Stick with 2048 byte sectors bootable. In fact I wiped out my Linux partitions bootrecord. This was with an old version of extlinux as still distributed by openSuSE. Interestingly enough the current version syslinux-3.61-pre2, spares my hard disk, but still seems to live under the impression it is the USB stick. The setup is