similar to: Problems Solved, two left

On 5/24/23 08:03, Steve Matzura wrote: > > ***  extensions.conf  *** > > > [general] > > [globals] > > ; Make sure to include inbound prior to outbound because the > _NXXNXXXXXX handler will match the incoming call and create a loop > include => voipms-inbound > include => voipms-outbound > > [voipms-outbound] > exten =>

Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb Steve Matzura: > > On 5/24/2023 7:49 AM, Stefan Tichy wrote: > > Am Tue, May 23, 2023 at 07:22:22PM -0400 schrieb Steve Matzura: > > > > > 1. Still can't register my phone > > > The username and password are correct. I don't know what else to try. > > You can start a sip trace from the asterisk

And I think they're both small. Solved: tcpdump showed no packets coming in, so I went to my DID provider's Website to discover to my intense embarrassment that the DID number had been set up forwarded to their voicemail. I got egg on my face for this one. I changed that setting to SIP/IAX and packets now arrive and go where they should. Two problems remain. 1. Still can't

This was supposed to go to the list. I am now thoroughly confused. In the [voipms] stanza where endpoint is defined (type=endpoint), everything points to voipms. But in the [yealink] stanzas, I tried pointing everything to Steve, one item at a time, then both of them, and nothing changed. On 5/24/2023 10:00 AM, Stefan Tichy wrote: block quote Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb

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

OK, this is how I thought it's supposed to work. It just confounded me why the book would say the Playback() and Background() syntax were the same, then in the very next paragraph give an example that belied that claim. On 6/17/2023 1:46 PM, Doug Lytle wrote: > On 6/17/23 08:47, Steve Matzura wrote: >> >> Both Background()  and WaitExten()  allow the caller to enter DTMF

2013/2/12 Doug Lytle <support at drdos.info> > >> non-standard locations such as /usr/local/sbin > > If compiling from source, it'd normally be specified by the --prefix > option: > > ./configure --prefix=/usr/local > > Doug > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary >

On 6/16/23 20:29, Steve Matzura wrote: > As always, thanks in advance for a kick in the right direction. For both capabilities, you can use Background() instead of Playback() for audio prompts.  Background() allows for interrupting the prompts and continue on with your dialplan. Doug -------------- next part -------------- An HTML attachment was scrubbed... URL:

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? > > >

On 6/17/23 08:47, Steve Matzura wrote: > > Both Background() and WaitExten()  allow the caller to enter DTMF > digits. Asterisk then attempts to find an extension in the current > context that matches the digits that the caller entered. If Asterisk > finds a match, it will send the call to that extension. > > > My question then is, is "*" a valid exension, as

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

Hello, I found the problem and also the workaround. Clearly, since it was working with chan_sip it should not be dialplan problem, but sip stack problem. I have line=yes set up. After asterisk restart the old registration is not unregistered and new one is registered with different line value. Then incoming invites and qualify requests are sent to all the registrations and there the problem

Ah no, I'm asking what code I put inside of parkedcalls? This example works? [ramais] include => parkedcalls [parkedcalls] exten => 700,1,ParkedCall(701) exten => 702,1,ParkedCall(702) exten => 703,1,ParkedCall(703) exten => 704,1,ParkedCall(704) This exten works? 2016-02-03 17:27 GMT-02:00, Doug Lytle <support at drdos.info>: >>>> On Feb 3, 2016, at

Humm, so this context parkedcalls is inside on features.conf? 2016-02-03 17:42 GMT-02:00, Doug Lytle <support at drdos.info>: >>>> On Feb 3, 2016, at 2:32 PM, Vitor Mazuco vitor.mazuco at gmail.com wrote: > >>>> Ah no, I'm asking what code I put inside of parkedcalls? > > Nothing, > > The context parkedcalls is generated by features.conf, you just

Thanks for trying, what asterisk version do you use? вт, 25 июн. 2019 г. в 17:50, Doug Lytle <support at drdos.info>: > We have Polycom phones (I'm using a VVX601, the destination is a VVX301). > We're also on Asterisk 13. > > I forwarded my call to the VVX301 and then dialed my phones DID. The > forwarded call showed my cell phone number, so I cannot reproduce.

On 7/5/19 10:50 AM, Doug Lytle wrote: > On 7/4/19 6:40 PM, hw wrote: >> This has again, and for no reason, ceased to work again after >> restarting asterisk.  No matter what I try, I can't create a >> certificate asterisk >> would verify. > > Have you considered using LetsEncrypt for a valid certificate? > > Doug > > What would be the point

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."

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

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