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http://www.wunderground.com/weather/api/ -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of dk at donkelly.biz Sent: Wednesday, December 16, 2015 9:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] weather.agi -----Original Message----- From:

HOUR_SELECTED is going to be 1-12 On Thu, Feb 13, 2020 at 2:05 PM Don Kelly <dk at donkelly.biz> wrote: > Is HOUR_SELECTED a floating-point number (e.g. 11.99999999999)? If so, you > need to account for that in your comparison. > > > > --Don > > > > > > *From:* asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] *On > Behalf Of

Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is "free"), and I do a

Matt, At some point you need to consider how much is too much... I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely

Could this possibly mean that any person who has CF set should never be available as CF Destination. Simple db entry/check can have this done. On Tue, Jun 2, 2015 at 5:34 PM, <dk at donkelly.biz> wrote: > > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *Kevin Larsen > *Sent:* Tuesday, June

Hi, i am using elastix 2.3 and created some dahdi extensions,now i dialing between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4 second before it ring the destination. so cany anyone know how fix it so that after dialing the digits the destination should ring . without any delay after dialing. regards Upendra. -------------- next part -------------- An HTML attachment was

Asterisk as a phone system makes perfect sense in a condo. You can get all the DID's you want and eliminate costs for the owners. You can offer standard FXO for people who don't care and IP sets for people who want to "upgrade" to feature sets. Your door openner is a piece of cake. 1. Create an option in your dialplan only in the "from-access-door" context that reads

Did you have a look at the phone it self already? Is call forwarding activated or something and can you call the phone/extension from externally? I have seen this in the past where an employee enabled call forwarding on the phone and once at home he or family called the phone which forwarded the call to abroad. Good luck. Michel. Op 29-01-15 om 12:51 schreef dk at donkelly.biz: > It's

My, how embarrassing. I of course meant that as a personal message to Don. But if anyone else knows the answer, I'm interested! lol Cheers, j On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote: > > Hey Don, > > How are you? I may be heading your way in the next month or so. Have > to meet with a guy in Eden Prairie, and stop off at my > brother/sisterm-in-law's as

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

I need to point out the this is factually misleading and materially false: "I think this, being the basis of your whole argument, is the fallacy. S/S is forcing people to take responsibility, for sure, but carriers won't just let their customers leave because they don't want to sign calls. It will force them to make sure they know who their customers are, and make it impossible for

I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that

Here is a funny story. We mostly do hotels in the Caribbean, and one of our first clients (going on ten years now) used the sample "weather.agi" that used to be shipped with... asterisk at home? Trixbox? Can't even recall where we originally got it from. This perl script uses festival to speak a brief weather forecast to the caller. We told our hotels this was a feature for

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Larsen Sent: Tuesday, June 2, 2015 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Forward loop protection... > Ia had a server overload today because someone did a call forward > to their own extension. To do a

hello all, i have problem in using g729 codec. my asterisk version is 1.8.22. when i run "core show codecs" in asterisk, there is a g729 codec in the list so i assume that i can use it for my channels. but connection can not be set when i use it for my h323 channel. i read somewhere that codec g729 is a commercial codec and i should buy its license in order to use it. is it true? if

Told you it was too complex of an issue :-) I figured to do this in zapata.conf and all is fine now: transfer=no That was the magic two letter which was sending a request for RLT feature on the line. Set transfer to "no" and all worries gone. Thanks for the input everyone. -Bruce On Mon, Apr 12, 2010 at 10:10 PM, bruce bruce <bruceb444 at gmail.com> wrote: > Futher check

Hi, I have asterisk installed on centos with phpagi. Also I have PRI card connect to it. it's possible to show the sip number when calling from sip number to external number thru the PRI, instead of showing the PRI number show the sip number ? Regards -Hadi.Salem

What I'm trying to achieve is that a voicemail message should be at least 3 seconds long for it to be saved, but *after that* a prolonged silence (e.g. 10 seconds) should terminate the call and recording. My current settings (Asterisk 10.7.0 and 11.2.1) are: ; Minimum length of a voicemail message in seconds for the message to be kept ; The default is no minimum. minsecs=3 ;

> Hmm the calls are made during the day (and sometimes very early in the > morning). Right now it looks like someone actually made these calls. If > that is the case it's somewhat comforting to know the system wasn't > compromised. However, the $25,000 phone bill still remains. Yikes. $6.25 > per minute to Cambodia seems quite steep to me. Since the Mitel had a default

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough.