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...I realize this is somewhat like the many caller-id-after-the-fact threads, but I figured maybe someone had solved this a different way. Has anyone been able to do this, via caller ID, messaging, the mini-browser in those phones, or some other way? Thanks! Martin Smith, Systems Developer martins@bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221

...#39;Zap/97-1' -- Executing Hangup("Zap/97-1", "") in new stack == Spawn extension (default, h, 1) exited non-zero on 'Zap/97-1' -- Hungup 'Zap/97-1' Any ideas as to why I can't hear anything? Thanks! Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality,

Hi, I'm trying to find a way of passing the actual call duration (something like ANSWEREDTIME) to an AGI script that runs periodically during a call. Any ideas? Thanks, Adi. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070531/115b0aee/attachment.htm

Hi, I just got a Polycom 330 and, of course, I don't have the firmware and sip.cfg files to provision it. Where can I get those? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070605/a14615c8/attachment.htm

Hi all, Does anybody know why my asterisk doesn't have a "show agi" command? Do I have to load any module for it? Thanks Ronaldo.

Dear Users ! I have recently installed asterisk 1.4 i got a warning message whenever i use reload or extensions reload. [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local'

...lly" or better yet, when dialing a dtmf digit, to prefix it with something to force asterisk to ignore it and pass it along? Am I stuck with absolutely no features that depend on # or * if I want users to use those digits on a remote PBX? Thanks :) Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221

...IP phone ChanSpy'ing them. We have ChanSpy set to only spy on bridged channels as well. Has anyone seen a bug like this? We often top 10,000 calls a month, and this only happens maybe twice a year. We're switching to 1.4 soon, but just curious :) Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221

Hey folks, Is anyone working on Asterisk (or other) presentation proposals for OSCON 2008 in Portland, OR? Here's the link, in case: http://en.oreilly.com/oscon2008/public/cfp/13 I'd love to see more Asterisk content there! Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

...party hangs up after a Dial(), redirect the called party to another location. I'm not sure how else to describe what the user wants to do, but I'm willing to try if people have questions :) Is there a simple way to do this without a meetme room? Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221

I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a

I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call

Hello there, I appreciate any help about this problem that I can't figure out... I need to record all my calls: this is pretty easy using Monitor() before the Dial(). eg: exten => 425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes

Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the

I know that I can record the contents of a call by calling Monitor() or MixMonitor() from the dialplan just before invoking Dial(). I have a potential customer who wants only the first minute of each call recorded (for identification purposes, without the storage overhead of keeping the complete call). Can anyone here think of the easiest way to do this? The only possibilities I can think of