similar to: Asterisk to Email

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

This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply

Hello Some SOHO prospects only have a cellphone and I was wondering if someone had investigate running Asterisk on a smartphone, to perform tasks such as IVR, CID rewriting, voice-mail, notifications through e-mails, etc.? Thank you.

So I think I am able to reproduce the dropped call scenario. Here is what I do to get a dropped call: 1. Call 1-800-tmobile 2. Go true their IVR and get connected to the customer service IVR 3. Enter my number and SSN 4. press 0 5. Then the audio please hold starts. After about 2-4 seconds the call gets dropped. (fast busy tone) The time on my phone will stop running (call time) and I will get

Hi I've got a cheap card from x100p.com for my pots line. I haven't found a definate answer if it is possible to get danish(DTMF without signaling before it). I have had a look at bug #9 but that is written longtime ago. I am running zaptel 1.4.7 and asterisk 1.4.14 BRIstuffed 0.4.0-test4 both from the xorcom debian packages. I believe they should contain some kind functionality for DTMF

I am trying to write a script to attempt to make a call on a Zap channel, and if it fails, send an alarm. I can generate the call, but because the Zap channel accepts the call, even though the other end never answers, it sees it as a successful call, which it isn't. Anyone have any ideas on this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there are 3 BT lines connected directly to these ports. One of the lines has BT FeatureLine Compact and this is the line I am having problems with, the other 2 lines are working perfectly, detecting CID, answering incoming calls and placing external calls via SIP devices. I am receiving a error log entry: chan_zap.c:

Hi All, I am getting these events in asterisk message log: NOTICE[16647] chan_zap.c: Got event 4 (Alarm)... NOTICE[16647] chan_zap.c: Alarm cleared on channel 1 after that asterisk exits silently until I restart it. Sometimes zapata drivers also get in a state where I need to physically restart the machine. Does anyone have any suggestions how to troubleshoot these alarm events? Roberts

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

asterisk-users at lists.digium.com has now been added to the filters white list! Anyway, 100ohm termination isn't required for ptp - but is required for ptmp. I know the DAHDI package(s) no longer include make b410p - hence the reason it is included in the docs.

We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time

I have a friend traveling overseas. I want to allow him to call a number which will give him a busy signal (so no charge), but will then send me an email that he has called. I know how to use a call file to trigger a call (I created a callback system for myself when I traveled overseas a few months ago), but I don't know how to trigger an email. All I want is a simple message like

The DAHDI docs actually include the 'steps' that used to be in Zaptel's 'make b410p'. These steps involve downloading and compiling mISDN. Why re-invent the wheel? Just a thought :-). As for the 100ohm termination bit - it's simply changing a couple of dip switches on the b410p card (as described in the manual). Andy -----Original Message----- From:

Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI: uniqueid => 17 customer_id => 0 context => techmast mailbox => 14 password => 1234 fullname => Sales and Service email => user1@domain.com email =>

Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows

All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy "fixed" the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) After doing this, I recompiled ztdummy and it worked. Note

Hello List, Im just dipping my feet into the asterisk world, and im having major fxo problems Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons (from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a Debian Etch box, with 1gb ram, running all of the services for my home server (web / db / music server etc), and i would like to run my PSTN line from Kingston Comms,

Hi, We recently dumped a Xorcom box that was no end of trouble and replaced with a Digium G100.  PRI came right up, and we have been using it fairly flawlessly for several months now, with one caveat.  Calls that arrive from the PRI are sent to the asterisk instance (13.23.1, chan_sip), then routed by the dialplan to various other gateways or upstream providers.  When the call finally lands

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as the

Hi all, After reading great things about the OSLEC Echo Canceller (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of people who have tried it on a recent Trixbox thread (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems), it sounds like it is the "bees knees" for sorting out echo problems with cards like the x100p. Has anyone