search for: karlfif

Does anyone have a Dial script or a hint on how I can dial 10000 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me.

QUESTION: Who's in the wrong: I recently saw an example of a u-law file with a metadata header on the file. The asterisk playback function 'PLAYED' the ascii header values as if they were audio data, creating an audible 'click'. After realizing the click was coming from metadata (and fixing it), I became curious: Which is 'correct? In other words: 1. Is it

Is there a digium twitter 'user' to follow that only tweets important announcements and release information? If there is not, I think there should be. It would be highly utilitarian to get an SMS when there is an update to Asterisk, Dahdi, ADA etc, but I don't want to be bothered real-time with asteriskpbx tweets like: "Anyone trying anything cool with Asterisk over the

Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send

Does anyone know the mechanism by which companies like YouMail (and MNO's using their own voicemail system) are able to redirect ALL calls from a ALL subscribers to *just one* voicemail DID, yet determine WHICH subscriber did the redirection? I had always assumed this was simply done using RDNIS. In other words, the original calling party's CallerID is passed with the redirected

Allison Smith just created a hysterical parody music on hold Parody. Whatever you were doing, stop, and dial this number to listen to it: 360-519-5689. 2 minutes. I just gave her a few ideas, but she took it and ran with it--she chose the audio and did the mix-down and everything. Really funny!! -Karl -------------- next part -------------- An HTML attachment was scrubbed... URL:

In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'. "Use the NEW ulaw/alaw codec's (slower, but cleaner)" By slower does this mean more 'expensive', or does it instead mean that there will be more algorithmic latency? Both? Can anyone speak to the relative increases? With regard to accuracy, can anyone speak to what kind of situation might

Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via

...1) = 0xb7360000 close(12) = 0 --- SIGSEGV (Segmentation fault) @ 0 (0) --- +++ killed by SIGSEGV +++ Process 921 detached Anyone else seen this? sean ------------------------------ Message: 12 Date: Wed, 20 May 2009 19:28:19 -0500 From: "Karl Fife" <karlfife at gmail.com> Subject: [asterisk-users] Voicemail playback NEWEST first vs. OLDEST first To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <BC16DF1437384D7C88BE5DA5ECBF8155 at kfife2> Content-Type: text/plain;...

Somebody requested a path to listen without termination charges. Here's a SIP URI: (a SIP What??) karlonhold at sip.kfife.com or 3605195689 at 74.92.179.65 Thanks -Karl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090102/33b3bf7b/attachment.htm

In what appears to be the most current documentation of FFA I'm directed to http://www.digium.com/en/docs/FAX/faa-download.php However the download selector utility found there "ain't working". Specifically, none of the drop lists appear to be populated (in any browser, on any platform), thereby preventing any actual 'selection'. Has anyone else noticed this? Is there

Should SipPhone support ISN routing for their 747 ITAD? Cast a vote: http://forums.gizmo5.com/viewtopic.php?t=10197 Meanwhile if you're interested, you can use the Nerd Vittles 'bandit' ITAD #1089 to call a SipPhone/Gizmo5 subscriber via ISN, which I think is clever (Karl tips his hat to Ward Mundy) and it's also really, really funny.

From: http://www.usac.org/_res/documents/hc/pdf/training-2009/USAC-USF-overview.pdf -The FCC now requires all VoIP telecom providers terminating and originating to the PSTN to charge Universal Services Fund tax to end user customers. -USF is 8.4% of revenue for VoIP Telco's. -Last year USAC collected and dispersed seven BILLION dollars for libraries, rural health care centers, high cost

The 'Make sounds' routine into Makefile doesn't seem to "pre-fetch" all of the audio tarballs. Is this an oversight or is there a strategic reason for it? Specifically it doesn't seem to fetch the MOH tracks for selected codec's. For example, during the most recent 1.6.1 update, the g.722 (asterisk-moh-opsound-g722.tar.gz) wasn't downloaded until "make

Does anyone know the rationale behind why deny/permit values can not be specified in 'general' setting of sip.conf & iax.conf In other words, if I want to deny everyone, then allow selectively permit specific hosts or subnets, I can't do so without first deny'ing all in EVERY user/peer definition. Too verbose. Naturally I can accomplish this using templates, but it DOES

Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-way calling, recording etc? It ALWAYS works perfectly with ALL of our Soundpoint IP 5/6xx phones, but NEVER for our IP7000s. I just want to know it's POSSIBLE before I keep slogging away at this. Is there a 'bastard_phone=yes'

Calls that come in on DAHDI FXO ports are routed to [context], extension 's' INSTEAD, I would like to route specific ports to specific extensions, For example: I want DAHDI/1-1 to go to 1234 I want DAHDI/1-2 to go to 2345 I want DAHDI/1-3 to go to 3456 ...etc What is the CLEANEST way to do this? Yes, I can create a private context for each DAHDI channel but that seems messy and

On the landing page of the Polycom web site there's a "We're listening" nanosurvey, asking what is the one thing Polycom can do to improve their products. The link points here: http://polycom.zuberance.com/survey.htm I wrote a sentence about tweaking the user interface on the IP Soundpoint series phones, so that one can escape any level of any menu with repeated pressing of the

What is the current best practice to avoid transcoding on an outgoing call to a party whose codec preference is not known in advance? In other words, incoming calls are easy since codecs are negotiated from least-known (the remote party) to most-known (my endpoint) and my codecs can simply be preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate

Question: Naturally there are times when need to I reformat an extension in a context as such: ;Reformat add CC1 exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) -or- ;Reformat 011 with with +CC exten => _011X. ,1,Goto(+${EXTEN:3},1) It's a helpful trick, BUT there are times when I want to send the call to another context in its original un-reformatted state. Naturally the ${EXTEN}