search for: dericed

Using Asterisk 1.6.1.14 and dahdi 2.2.0.2+2.2.0. We're placing outbound calls over an analog line. Some of these calls are going to cell phones that play music rather than providing a standard ring. As a result, the Dial command sometimes returns a DIALSTATUS of CHANUNAVAIL and sometimes it returns BUSY. The problem is that this is happening on calls that are being answered. Has anyone else

...lease has been made (available at https://github.com/privatezero/flac_markdown/releases) and the draft document has been uploaded to the IETF datatracker ( https://datatracker.ietf.org/doc/draft-xiph-cellar-flac/) All the best! Andrew Weaver On Mon, May 22, 2017 at 11:06 AM, Dave Rice <dave at dericed.com> wrote: > > On May 12, 2017, at 1:05 PM, Dave Rice <dave at dericed.com> wrote: > > Hi all, > And cc'ing flac-dev. > > On May 10, 2017, at 12:15 PM, Dave Rice <dave at dericed.com> wrote: > > Hi Andrew, > > On May 10, 2017, at 11:19 AM, Andre...

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. Thanks, Deric.Page at nisc.coop -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, And cc'ing flac-dev. > On May 10, 2017, at 12:15 PM, Dave Rice <dave at dericed.com> wrote: > > Hi Andrew, > >> On May 10, 2017, at 11:19 AM, Andrew James Weaver <weevz at uw.edu <mailto:weevz at uw.edu>> wrote: >> >> Hello all! >> >> In a previous discussions on this list about people interested in working on the FLAC...

I've set up an outbound .call system for customer callbacks and the like. Calls are going out over analog lines and I'm trying to use the WaitForSilence routine to make sure the phone has stopped ringing before starting message playback. The problem is that if I set the first argument of WaitForSilence to anything other than 1, WaitForSilence never exits. Some general info on my setup:

> On May 12, 2017, at 1:05 PM, Dave Rice <dave at dericed.com> wrote: > > Hi all, > And cc'ing flac-dev. > >> On May 10, 2017, at 12:15 PM, Dave Rice <dave at dericed.com <mailto:dave at dericed.com>> wrote: >> >> Hi Andrew, >> >>> On May 10, 2017, at 11:19 AM, Andrew James Weaver <wee...

Is it possible to place multiple phone numbers in a single outbound .call file? If I try doing this, only the last phone number in the file is called. However, if I use 1 file per phone number, then Asterisk attempts to process all generated CALL files at once, incrementing the retry count for each that cannot be called because the designated channel is busy. For example if I have a list of 10

https://bugzilla.samba.org/show_bug.cgi?id=10074 Summary: rsync reorders --from-files alphabetically Product: rsync Version: 3.0.6 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P5 Component: core AssignedTo: wayned at samba.org ReportedBy: nicholas.man at

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

Asterisk 11.8.1 Today our Asterisk server locked up. Looking through the asterisk full log, I see the following over and over again and the log just continues to grow until we reboot Asterisk. Has anyone seen anything like this before or have any idea what may cause it? [Aug 19 14:41:26] DEBUG[7362] manager.c: Examining event: Event: HangupRequest Privilege: call,all Channel:

Hi all, > On Jun 5, 2017, at 11:52 PM, Andrew James Weaver <weevz at uw.edu> wrote: > > Hello all! > (cc-ing the flac-dev list) > > I would like to give an update as to the recent CELLAR work on the FLAC specification. > > • Work has been done to make internal and external links more accurate and reliable. > • 'Rice Coding' has been clarified as

Hi Ruud van Asseldonk, > On Jun 11, 2017, at 7:24 AM, Ruud van Asseldonk <dev at veniogames.com> wrote: > >>> I'm jumping in on this thread to make a few remarks about the spec. I >>> implemented a FLAC decoder by only looking at the spec, and I have a few >>> notes that would have saved me a lot of time if the spec had mentioned >>> them. They