search for: lytles

I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So far, the only issue that I've encounted is. I have a scheduled CRON job that runs at 3am every Sunday, that issues a: asterisk -rx 'restart when convenient' The first Sunday that it ran, Asterisk never restarted. The CRON logs show that it issued the command successfully. This Sunday, it ran but never

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

I have a customer who named their endpoint to include a space (example, 1003 a) >From the CLI, I want to hangup a channel on this endpoint >From core show channels concise, I see the channel name includes the space PJSIP/1003 a-00000002 I realize the space is interpreted as an argument separator, so my first attempt below doesn't work. I have tried the following and all fail. hangup

Thanks Doug. Turns out if using hangup request does not work with the escaped character CLI> hangup request PJSIP/1003\ a-00000007 Usage: channel request hangup <channel>|<all> Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel. If 'all' is specified instead of a channel name, all

Anybody having issues with ztdummy under the current 2.6 RC7? I get the following errors when trying to modprobe ztdummy: "Unable to register zaptel rtc driver" Doing a Google on the error shows reference to a message from 2004 that said you might not have RTC compiled into the kernel. Checking via: cd /usr/src/linux-2.6.13-rc7 grep -i rtc .config shows: CONFIG_APM_RTC_IS_GMT=y

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

Is there a way to limit outbound calls when feeding files to the outgoing directory in asterisk? I several thousand files i need to feed asterisk, hoping to copy it to the outgoing directory all at 1 time.

Hey everybody, I have a situation where we have 2 Asterisk (CVS as of 08/25/2005) connected via IAX. On the corporate side, we have 1 TE110P connecting to a Definity G3R and it's connecting to a TN464F card, giving a 23 channel connection. I have echocancel=yes, echotraining=yes and echocancelwhenbridged=yes. One the remote office side, they a Adit 600 channel bank for 10 outside

We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to

Everybody, I have a fall though context that, among other things, tests to see if someone it trying to pick up a non-existent parked call (Defined from 90 to 99). I have the following: [not-in-service] exten => _X.,1,Wait(1) exten => _X.,n,ResetCDR() ; ************************************************** ; Check to see if the mis-dialed number was a parking ; slot. If so, jump to the

I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco <-cA-> Machine1 <-cB-> Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1

>>Your membership in the mailing list asterisk-users has been disabled >>due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Last month I moved a 1.4.x Asterisk install to Asterisk 11.5.1. Everything is working well, until I noticed that Caller ID between facilities are showing properly, on the phone display, until the handset is picked up, then it's displaying NAME NOT FOUND. I do database lookup against extension number (Of remote PBX) and use that as the MySQL key to pull name. Snippet below: -- Executing

On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > Doug, > > I tried that as well. Even with my dialplan looking like this: > > > > Ordering by includes works for me under Asterisk 11 and 13 > > What does the output of the below show? > > dialplan show from-external > >

> exten => _13XXXXXXX,1,dial(${OPERATOR},20) Hello "SIP/2.0 401 Unauthorized" Unfortunately the negative. An asterisk indicates a 404 error. On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle <support at drdos.info> wrote: > > On 3/5/19 2:46 AM, Gokan Atmaca wrote: > > Asterisk can send calls, but I don't get a call. What could be the problem? > > >

I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip and I'm trying to access the script that is provided to help with conversion. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip It would appear that said server hosting the script is no responding or the link is no longer valid. Doug

Everybody, For a while now, I've had a small home Asterisk setup to connect to my Zimbra mail server's calendar.  Making an entry on the calendar would cause Asterisk to schedule a wakeup call at the time of the calendar entry. The Zimbra mail server uses LetsEncrypt for the SSL Certs and renews every 60 days.  On the Asterisk side of things, if I do not restart the Asterisk process,

On 15.05.20 at 14:31 Doug Lytle wrote: > Google says Round Trip Time > > https://www.voip-info.org/asterisk-rtcp/ That doesn't answer my question (I know the abbreviation RTT). Therefore I'm trying again: I'm just wondering what the RTT *exactly* means. Where are the exact measuring points located? => How are the RTT values exactly calculated? Which values are actually