Re: [asterisk-users] TON values

2021-03-12 Thread Doug Lytle
Mike, The below link turned up for me in a Google Search https://www.voip-info.org/asterisk-config-chandahdiconf/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] STIR/SHAKEN

2021-03-07 Thread Doug Lytle
On 3/7/21 1:43 PM, Greg Troxel wrote: So I wonder if your asterisk instance is connecting to the PSTN as a top-level carrier, or, more likely, I am confused in some way. Greg, I think this is the case for quite alot of those here. For me though, I just manage the on premise PBX and my

Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.

2021-01-04 Thread Doug Lytle
>>> OK, both combination worked but still silence until the all numbers are >>> dialed. I have never used the U option on the dial command to call a sub-routine, Doug -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.

2021-01-04 Thread Doug Lytle
>>> Gosub(check-number-forwarding,Dial(SIP/718xxpstn-5665,20,U(atb-sub)) >>> Is ARG1 = atb-sub ? No. My complete line exten => _45XX,1,Set(_ARG1=${EXTEN} same => n,Gosub(check-number-forwarding,s,1(${ARG1})) So, if someone were to dial a 4 digit number starting with 45 (i.e.

Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.

2021-01-04 Thread Doug Lytle
>>> How do you enable the phone speaker on the Gosub? >>> I had: >>> Dial(SIP/718x@pstn-5665,20,m(default)M(atb)) You can provide variables to your gosub routine, for an example Gosub(check-number-forwarding,s,1(${ARG1})) Doug --

Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.

2021-01-04 Thread Doug Lytle
>>> app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is >>> not available. Macros are no longer built by default in Asterisk 16. This was documented in the UPGRADE.txt file app_macro: - The app_macro module is now deprecated and by default it is no longer built.

Re: [asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel

2020-12-23 Thread Doug Lytle
Review your features.conf file in /etc/asterisk Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk?

Re: [asterisk-users] I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones

2020-12-18 Thread Doug Lytle
>>> Also, you will need a TFTP server working on your Asterisk box My suggestion would be to get a refurbished Polycom VVX 301 phone (With power brick if no POE is avaiable) for around $27 US. Doug -- _ -- Bandwidth and

Re: [asterisk-users] Timing source for Asterisk

2020-12-09 Thread Doug Lytle
The wiki page has some information on timing and troubleshooting https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

Re: [asterisk-users] which linux for asterisk?

2020-12-09 Thread Doug Lytle
On 12/9/20 2:03 AM, Dmitry Melekhov wrote: But because Centos is declared dead, what is best choice ? Oracle? Ubuntu? And for those that have no idea as to what he is referring to (I didn't), here is the Register article https://www.theregister.com/2020/12/09/centos_red_hat/ Doug --

Re: [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction

2020-08-08 Thread Doug Lytle
On 8/8/20 8:35 AM, Jerry Geis wrote: The VM is Intel box (host) and the physical box is a celeron. So something is not right there. What would be a good ./configure option that asterisk can compile with on the VM image so this illegal instruction does on occur ? Jerry, Under Compiler Flags

Re: [asterisk-users] Log queue threshold (1000) exceeded. Discarding new messages.

2020-06-26 Thread Doug Lytle
On 6/26/20 4:16 PM, Antony Stone wrote: Where can I set this threshold? /etc/asterisk/logger.conf ; All log messages go to a queue serviced by a single thread ; which does all the IO.  This setting controls how big that ; queue can get (and therefore how much memory is allocated) ; before new

Re: [asterisk-users] ODBC connection failure - can it be fatal?

2020-06-23 Thread Doug Lytle
>>> Is there any way I can tell Asterisk that an ODBC connection problem is a >>> fatal error Your be best bet would be to do that check in the script that starts up Asterisk and maybe a CRON job that periodically tests connectivity. Doug --

Re: [asterisk-users] Controlling Asterisk from within the dialplan

2020-06-23 Thread Doug Lytle
>>> other than using the System() command? Not that I am aware of, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/

Re: [asterisk-users] Mail2Fax

2020-06-19 Thread Doug Lytle
On 6/19/20 4:23 AM, basti wrote: Fax is not send. No Sip stuff is show in log. I don't know what is wrong here. Best regards Basti, This really belongs on the iaxmodem mailing list or the HylaFAX+ Mailing list.  Lee Howard is the author of both packages and very responsive. Doug --

Re: [asterisk-users] CDR mysql: timeout when remote database unavailable

2020-06-10 Thread Doug Lytle
>>> Instead, the call still terminates if mysql cannot be reached. I just tested this, I'm using cdr_odbc, by shutting down mysql and I did not experience the call being dropped. The console logged the mysql failure, but the call continued. You may want to consider moving to cdr_odbc instead.

Re: [asterisk-users] call replicating

2020-06-05 Thread Doug Lytle
On 6/5/20 12:24 PM, Marek Greško wrote: How can this behavior been overriden? I do not expect this is problem on provider side, since it was working normally using chan_sip. Console output and dial plan snippets are always useful when diagnosing, Doug --

Re: [asterisk-users] Notification when on the phone

2020-05-29 Thread Doug Lytle
On 5/29/20 2:28 AM, Administrator wrote: You could also use DEVICE_STATE I am using DEVICE_STATE to identify when a phone is in use: exten => s,n,GosubIf($["${DEVICE_STATE(SIP/${ARG1})}" = "INUSE"]?SHOWBUSY,s,1(${ARG1})) I'm trying to figure out the best way to display that information to

Re: [asterisk-users] Notification when on the phone

2020-05-28 Thread Doug Lytle
>>> But if you've already got the caller on the phone, then you might consider >>> the CONNECTEDLINE function in Asterisk... And that we don't. It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions

[asterisk-users] Notification when on the phone

2020-05-28 Thread Doug Lytle
Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old

Re: [asterisk-users] Asterisk : CDR Analyzer Updated

2020-05-25 Thread Doug Lytle
On 5/25/20 5:56 AM, Mitul Limbani wrote: Maybe you can have it uploaded on GitHub.com as a repository ? With a README.md file on how to install it for PHP7 ? Anybody that would like to do this would be most welcome. I have no plans on supporting it. Basic instructions and attachment will

[asterisk-users] Asterisk : CDR Analyzer Updated

2020-05-25 Thread Doug Lytle
Everybody, I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a dozen years, it was easy to configure and didn't requite installing 'connectors' on anything or adding tables on the DB server. It's based off of PHP5 and the only reason I still keep around a Debian 7

Re: [asterisk-users] Meaning of RTT in channelstats

2020-05-16 Thread Doug Lytle
On 5/16/20 9:57 AM, Michael Maier wrote: 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

Re: [asterisk-users] Meaning of RTT in channelstats

2020-05-15 Thread Doug Lytle
Google says Round Trip Time https://www.voip-info.org/asterisk-rtcp/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] Mute conference participants

2020-04-26 Thread Doug Lytle
On 4/26/20 10:48 AM, Dovid Bender wrote: Hi, Looking at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there is an option for admin_toggle_mute_participants however the non admin users can still toggle toggle_mute. Is there any option for the admin to disallow non

Re: [asterisk-users] Troubleshooting load issues

2020-04-22 Thread Doug Lytle
>>> All the calls are using ulaw. The files that I am playing are gsm. I >>> suppose doing a file convert with sox to .ulaw may help but it should be >>> able to do 500 calls without an issue. Can it possibly be a bug? if not how >>> do >>> I profile which call(s) can be causing the spike?

Re: [asterisk-users] audio problem with asterisk and meetme conference

2020-03-26 Thread Doug Lytle
>>> Can I adjust the talk or listen volume for another user? I've never used the volume controls, but it would appear. https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration Doug -- _ -- Bandwidth and Colocation

Re: [asterisk-users] audio problem with asterisk and meetme conference

2020-03-26 Thread Doug Lytle
>>> I never moved to confbridge because they don't have an option for >>> controlling the volume of other >>> participants audio I have menu options in my confbridge configs that has increase and decrease conference volume. I'd still configure a small confbridge and test if you still have the

Re: [asterisk-users] audio problem with asterisk and meetme conference

2020-03-25 Thread Doug Lytle
>>> he problem is that there is some sort of distortion in the audio Has been been going on for a while or is this a new setup? Do you have a timing source? I bit the bullet around a year ago and moved to CONFBRIDGE; it wasn't as horrible as I thought it would be to setup. Doug --

Re: [asterisk-users] DAHDI not loading

2020-03-16 Thread Doug Lytle
>>> How do I do that? If you are using your package manager to install Asterisk & Dahdi, then I would not suggest that you compile. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

Re: [asterisk-users] DAHDI not loading

2020-03-16 Thread Doug Lytle
>>> I saw something about needing to SIGN the dahdi modules. How do I do that ? >>> If that is the solution. Just a guess, Recompile Dadhi. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] No CID between Asterisk using IAX trunk

2020-03-02 Thread Doug Lytle
My Asterisk 13 IAX2 trunk posted below: type=friend trunk=yes allowcallerid=yes disallow=all allow=alaw allow=ulaw allow=gsm host=my.super.duper.host username=my.super.duper.username secret=my.super.duper.secret context=sip qualify=500 qualifysmoothing=yes requirecalltoken=no trunk=yes

Re: [asterisk-users] No CID between Asterisk using IAX trunk

2020-03-02 Thread Doug Lytle
>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>> trunk between them. Carlos, Had caller-id ever worked between these two systems? Doug -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] reviewboard.asterisk.org SSL Trust Failure

2020-02-18 Thread Doug Lytle
>>> Reviewboard is a legacy site and will likely be shutdown. Is there a reason >>> you are wanting to visit it? After seeing Olivier's post about his recent failures on compile and it referencing NBS (Network Broadcast Sound), which I had never heard of, I was googling to find out more and

[asterisk-users] reviewboard.asterisk.org SSL Trust Failure

2020-02-18 Thread Doug Lytle
Under Firefox, browsing to https://reviewboard.asterisk.org I get Warning: Potential Security Risk Ahead Firefox detected a potential security threat and did not continue to reviewboard.asterisk.org. If you visit this site, attackers could try to steal information like your passwords, emails,

Re: [asterisk-users] Call from an extension

2020-01-28 Thread Doug Lytle
I understood that part, I was hoping to understand why. In the past, I've used the PSTN lines to connect two Asterisk systems for extension to extension calls and was able to route source and destination extensions via the dial-plan, just by parsing the assigned CID. Was thinking that may be

Re: [asterisk-users] Call from an extension

2020-01-28 Thread Doug Lytle
>>> I desire to make a call from my system looking like it comes from 4452 and >>> call the outside number If you have control over your CID with your provider, you can use Set(CALLERID(number)=4452) Otherwise, you cannot. If you would provide us with what you are trying to accomplish, maybe

Re: [asterisk-users] Call from an extension

2020-01-28 Thread Doug Lytle
>>> I can make calls over a SIP trunk as SIP//number >>> I am trying to make calls over an extension thought using the same format >>> SIP/4452/number - its not working No, Extension to extension calls would be: Dial(SIP/${EXTEN]) My extension to extension dial line is exten =>

Re: [asterisk-users] From the CLI, how can I hangup a channel name that includes a space character?

2020-01-16 Thread Doug Lytle
>>> Is there some control character(s) for the CLI to interpret everything in >>> between as a single argument? I think you can typically use tab completion when working with spaces or you can escape the space with a back slash For example Doug Lytle would be

Re: [asterisk-users] res_calendar & LetsEncrypt

2019-12-24 Thread Doug Lytle
On 12/24/19 10:34 AM, Sean Bright wrote: On 12/24/2019 9:02 AM, Doug Lytle wrote: [Dec 24 07:48:46] WARNING[10679] res_calendar_caldav.c: Unknown response to CalDAV calendar calendar.name.here, request REPORT to /dav/username/Calendar: Server certificate changed: connection intercepted

[asterisk-users] res_calendar & LetsEncrypt

2019-12-24 Thread Doug Lytle
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

Re: [asterisk-users] Block Spam Calls

2019-12-13 Thread Doug Lytle
On 12/13/19 11:48 AM, Julian Beach wrote: Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: This is exactly what I do - “press 1 for a human” Works great I do this as well, but I also do a database lookup to see if the number is on our speeddial list and if so, pass the call

Re: [asterisk-users] Block Spam Calls

2019-12-13 Thread Doug Lytle
On 12/12/19 6:55 PM, Adam Goldberg wrote: This is exactly what I do - “press 1 for a human” Works great I do this as well, but I also do a database lookup to see if the number is on our speeddial list and if so, pass the call directly on without the IVR prompts. Doug --

Re: [asterisk-users] Simple, fast single-word offline free speech recognition in Asterisk (or as an AGI)?

2019-11-26 Thread Doug Lytle
On 11/26/19 12:31 AM, Jonathan H wrote: Yes, I know I post similar back in January, but there was no response back then and I was hoping things might have changed :) I'm using IBM's Watson for voicemail transcriptions, they allow 500 minutes per month for speech to text on the Free/Lite plan. 

Re: [asterisk-users] email notification on missed call

2019-10-30 Thread Doug Lytle
On 10/30/19 12:10 AM, Fourhundred Thecat wrote: Does asterisk not have some internal function to send email ? It does so for voicemail. Is there perhaps a better way to this than described above ? As far as I am aware, Asterisk has no built-in dialplan function to allow sending of email.

Re: [asterisk-users] clarification on gosub, macros and AEL

2019-10-15 Thread Doug Lytle
>>> Nobody has any information or opinions on any of this? Personally, I don't think MACROS are going anywhere any time soon, so I have not bothered looking into a substitution. As for ael; I've never used it. Doug -- _ --

Re: [asterisk-users] setting up ODBC for cdr logging into MariaDB

2019-10-12 Thread Doug Lytle
On 10/12/19 8:15 AM, Fourhundred Thecat wrote: did you compile libmyodbc yourself ? No, If I recall correctly, after a lot of searching, I ran into the apt source below and created the myodbc.list and put it into /etc/apt/sources.list.d cat myodbc.list deb http://ftp.de.debian.org/debian

Re: [asterisk-users] setting up ODBC for cdr logging into MariaDB

2019-10-12 Thread Doug Lytle
On 10/11/19 10:12 PM, Fourhundred Thecat wrote: Hello, I am trying to set up cdr logging into MariaDB through ODBC. I have installed unixodbc unixodbc-dev and now I am struggling with configuring /etc/odbcinst.ini All the examples online use non-existent libraries, ie: On my Debian Buster

Re: [asterisk-users] Amazon AWS question

2019-08-21 Thread Doug Lytle
Dan, I don't run Asterisk on AWS, but I do on ESXi. Are you running a version of Asterisk before 13? Newer versions Asterisk handle timing better that don't require a hardware timing source. I'm running Asterisk 13 on a small 60 phone system without issues under ESXi 6.0 Doug --

Re: [asterisk-users] Lightweight ODBC DB

2019-08-01 Thread Doug Lytle
On 8/1/19 5:08 PM, Dovid Bender wrote: Glenn, I can't use MySQL as each node currently has MySQL however there is a lot of data that is stored locally on each box. I may have to take this route if I can't find something else but that would mean syncing all sorts of data that does not need to

Re: [asterisk-users] svnview.digium.com down?

2019-07-24 Thread Doug Lytle
>>> I have updated the wiki. The script can be found within the >>> contrib/scripts/sip_to_pjsip subdirectory of an unpacked download of >>> Asterisk 13 and forward. Got it! Thanks, Doug -- _ -- Bandwidth and Colocation

[asterisk-users] svnview.digium.com down?

2019-07-24 Thread Doug Lytle
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

Re: [asterisk-users] Better audio in than just 8k

2019-07-11 Thread Doug Lytle
Maybe streaming will be helpful, https://www.agix.com.au/streaming-internet-music-for-asterisk-10-on-hold-music/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] Asterisk and pulseaudio Console/dsp

2019-07-10 Thread Doug Lytle
>>> I setup and extension to connect me with Console/Dsp. I am hearing the >>> audio but its warbly or does not sound right. Any thoughts on what I need >>> to do for that ? I had that issue at a previous employer and got around it by using ALSA instead. Doug --

Re: [asterisk-users] Asterisk and Linphone

2019-07-05 Thread Doug Lytle
My self-compiled Asterisk also shows that speex dependencies are not installed Speex Coder/Decoder Depends on: speex(E), speex_preprocess(E) Can use: speexdsp(E) You'll need to installed the dependencies and re-compile. Doug --

Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?

2019-07-05 Thread Doug Lytle
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 --

Re: [asterisk-users] Spectrum SIP trunks

2019-06-28 Thread Doug Lytle
>>> We've recently replaced an old Meridian phone system (Analog) with Asterisk >>> and signed up for Spectrum SIP trunks. Should have included that we're running Asterisk 13, under chan_sip Doug -- _ -- Bandwidth and

[asterisk-users] Spectrum SIP trunks

2019-06-28 Thread Doug Lytle
We've recently replaced an old Meridian phone system (Analog) with Asterisk and signed up for Spectrum SIP trunks. The service gets installed on July 8th and I was hoping somebody that may have already gone through the process could give me some hints. I've only ever dealt with PRIs or IAX2

Re: [asterisk-users] 302 moved temporally callerid behavior

2019-06-25 Thread Doug Lytle
core show version Asterisk 13.26.0 built by doug @ asterisk on a x86_64 running Linux on 2019-04-05 11:41:43 UTC Built from source, Douh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

Re: [asterisk-users] 302 moved temporally callerid behavior

2019-06-25 Thread Doug Lytle
>>> Surely that is "call forwarding", which is quite different from either a >>> blind or attended transfer? That would be correct. The forward button on the polycom phones just do a redirect to the destination extension or external phone number. Doug --

Re: [asterisk-users] 302 moved temporally callerid behavior

2019-06-25 Thread Doug Lytle
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. Doug --

Re: [asterisk-users] Res_Srtp

2019-03-31 Thread Doug Lytle
On 3/31/19 8:21 AM, Gokan Atmaca wrote: Hello The "res_srtp" module does not appear. How do I install it? Are you compiling or installing from packages? If compiling, you'll need to install the development library.  Under Debian it is libsrtp0-dev Doug --

Re: [asterisk-users] Paging systems?

2019-03-21 Thread Doug Lytle
>>> Does anyone have an (overhead) paging system that they like that works with >>> SIP? Our old phone system back ends into a Bogen AMP. I'm in the process of replacing that system (Meridian) with Asterisk and found that the snom PA1 works very well. If an AMP is involved, this might work.

Re: [asterisk-users] IVR Loop

2019-03-15 Thread Doug Lytle
Your IVR should only play audio prompts and only attempt to dial once a selection has been made, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] internal call record

2019-03-09 Thread Doug Lytle
On 3/9/19 9:56 AM, Gokan Atmaca wrote: a) work for recording incoming / outgoing calls b) do not work for recording internal calls then we might be able to give you a clue what's wrong. Hello For example: My phone number is 1000, the other's number is 1001. These numbers are in the same PBX

Re: [asterisk-users] asterisk 16.2.1 inbound route

2019-03-05 Thread Doug Lytle
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? [from-siptrunk] exten => 13XXX,1,dial(${OPERATOR},20) You are trying to match a pattern, so this needs to be exten => _13XXX,1,dial(${OPERATOR},20) Doug --

Re: [asterisk-users] MWI Delayed on Polycom VVX phones

2019-01-15 Thread Doug Lytle
>>> I'll try that patch later on today. I'm not using the mailboxes=##, but >>> will try the patch just the same. Patch applied and fixed my problem, Doug -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

2019-01-15 Thread Doug Lytle
>>> Carlos and Stefan (and other who have helped): Thomas, You stated that your virtual environment was Oracle, would that equate to VirtualBox? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] MWI Delayed on Polycom VVX phones

2019-01-15 Thread Doug Lytle
>>> https://github.com/astlinux-project/astlinux/commit/3bfd9f0400e990a42e1317f4aa2bad51a3ef9f17 >>> I am using "mailboxes=##@default" and had the issue as well (before). >>> Michael Thanks Michael! I'll try that patch later on today. I'm not using the mailboxes=##, but will try the patch

[asterisk-users] MWI Delayed on Polycom VVX phones

2019-01-15 Thread Doug Lytle
Hi all, When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes. After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing. Installing Asterisk

Re: [asterisk-users] Voiceprompt Organisation

2018-12-29 Thread Doug Lytle
On 12/28/18 8:24 AM, Cyril Alberts wrote: Maybe You have a chart like this for the print extensions, too? No sir, That's all I have. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

Re: [asterisk-users] Voice mail: MWI problem / pjsip (13.24.0)

2018-12-28 Thread Doug Lytle
>>> Before I'm opening an issue, I would like to prove my expectations - maybe >>> it isn't a problem at all or it's a problem of the phone. Michael, Just a side note. I've had reports of MWI not turning off after a message has been listened to under both 13.24.0 and 13.24.1. It will

Re: [asterisk-users] AMI not listening on secondary IP address?

2018-10-23 Thread Doug Lytle
>>> No, it's not a firewall problem; I've currently allowed connections to 5038 Antony, Do you have any deny/permit section in the manager.conf that would need to be adjusted? Doug -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Autoreply ( Autoreply (Re: getting invites to rtp ports ??))

2018-09-10 Thread Doug Lytle
And Google Translate: "Thank you for your message. Online4You is no longer operational since 1 August. We have informed you of the circumstances and your options by e-mail. Unfortunately we can no longer help you, your mail will not be forwarded and / or answered." Doug

Re: [asterisk-users] change dialing process on live call

2018-08-19 Thread Doug Lytle
On 08/19/2018 05:57 AM, Khalil Khamlichi wrote: Is there a way to add another extension to a live dial, for example Dial(PJSIP/1000,,) and after 20 secondes change it to Dial(PJSIP/1000/1001,,) This is a simple one.     exten => s,1,Dial(SIP/1000,20)     exten => s,n,Dial(SIP/1000/1001,20)

Re: [asterisk-users] G729

2018-07-26 Thread Doug Lytle
>>> And I'm not at all against Digium making money. But this whole 729 thing is >>> a mystery without any official feedback... Not an official statement from Digium, but at the top of that thread, it says it all: "david551 Sep '17 It still need copyright licensing. You will need to find an

Re: [asterisk-users] Asterisk not matching longest prefix with include

2018-06-26 Thread Doug Lytle
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 Doug --

Re: [asterisk-users] Asterisk not matching longest prefix with include

2018-06-26 Thread Doug Lytle
On 06/26/2018 06:57 PM, Dovid Bender wrote: Hi, My dialplan looks like this: [from-external] Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) Exten => _X.,n,Noop(CALLED NUMBER: ${EXTEN}) Exten => _X.,n,Ringing Exten =>

[asterisk-users] Voicemail Directory

2018-06-22 Thread Doug Lytle
I am currently using Asterisk 13.21.1 under Ubuntu (Compiled from source). The Dial-by-name directory option that I'm currently using: Directory(sip,sip,eb) That allows for first and last name matching. I've recently enabled forwarding voicemail with the directory by enabling

Re: [asterisk-users] Trying to add MoH to conference bridge

2018-05-24 Thread Doug Lytle
On 05/23/2018 05:23 PM, Mike Diehl wrote: However, my user isn't hearing anything.  MoH does work otherwise. The only difference between your setup and mine is that I'm having them wait for the marked user.  In that case, MOH does play. What does your console output look like? Doug --

[asterisk-users] IBM Watson Voicemail Transcription

2018-05-22 Thread Doug Lytle
to enable this on a user basis and not all or none. It appears that the mailcmd= line cannot be used on an individual voicemail box, at least my try failed 4155 => xxxx,Doug Lytle,dlytle@REDACTED,,mailcmd=/usr/local/bin/sendmailibm Any ideas on how to go about limiting it? D

Re: [asterisk-users] withheld caller id

2018-04-11 Thread Doug Lytle
>>> Thanks for the reply. So how do i alter my config to call with prefix 9+the >>> code to block caller id(#31#)+ the number? That's something I'll leave for you to investigate. As many have said, "Google is your friend" Doug --

Re: [asterisk-users] withheld caller id

2018-04-11 Thread Doug Lytle
On 04/10/2018 08:02 AM, Atux Atux wrote: so any ideas, please? On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux > wrote: after adding the ww: See of the D option of dial will do it: D([called][:calling[:progress]]): Send the specified DTMF

Re: [asterisk-users] withheld caller id

2018-04-10 Thread Doug Lytle
>>> My suggestion would be to add a pause or two before dialing the phone number Looks like using w for a pause is no longer supported. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

Re: [asterisk-users] withheld caller id

2018-04-10 Thread Doug Lytle
>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) My suggestion would be to add a pause or two before dialing the phone number exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) D(digits): After the called party answers, send digits as a DTMF stream, then connect

Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

2018-02-28 Thread Doug Lytle
The Polycom phones are using the standard FTP; It was only a shot in the dark. Doug On 02/28/2018 05:15 PM, Thomas Peters wrote: Doug: Not sure these phones understand TFTP. Docs say they don't. But I made changes like what you suggested. No luck; the phone still never bothers the

Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

2018-02-28 Thread Doug Lytle
>>> but unfortunately, my difficulty is much more basic. I think it has to do >>> with DHCP, specifically, what options I’m offering the phone via DHCP. I'm using ISC DHCP with the following: option tftp-server-name "10.30.100.109"; next-server 10.30.100.109; That provides enough

Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

2017-12-12 Thread Doug Lytle
>>> If you do not get an answer from the user list, you might try a post to >>> the dev list. >>> It is a bit more active and some of the people watching for dev news >>> might be able to help you. Surely, you mean the Biz List Doug --

Re: [asterisk-users] Can't park/unpark/re-park call

2017-11-14 Thread Doug Lytle
>>> The problem lies when I go to park the call again, and nothing happens. I’m >>> running Asterisk 11. Any insight at all would be greatly appreciated. Check your /etc/asterisk/features.conf Look for the option: parkedcallreparking Doug --

Re: [asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

2017-09-02 Thread Doug Lytle
On 09/02/2017 06:36 PM, O. Hartmann wrote: Is this question to "blunt" for this forum? No, But it is a holiday weekend in the States. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Doug Lytle
On 08/28/2017 06:00 PM, Joseph Smith wrote: I set no optimize and better backtrace through "make menuselect" and the output is now Please ignore the noise, I need to slow down when I read. Doug -- _ -- Bandwidth and

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Doug Lytle
On 08/28/2017 06:00 PM, Joseph Smith wrote: I set no optimize and better backtrace through "make menuselect" and the output is now menuselect => Compiler Flags => Better Backtraces Doug -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

2017-08-02 Thread Doug Lytle
>>> On Aug 2, 2017, at 6:45 AM, Richard Kenner ken...@gnat.com wrote: >>> I wouldn't believe it either. I'd be quite surprised if something won't >>> work with any ESXI version. *Perhaps* there's a configuration issue, but >>> I'd be surprised about that too. There are certain versions of the

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

2017-08-01 Thread Doug Lytle
>>> I am having a very tough time trying to replace an Elastix 2.X >>> install running as a virtual machine on ESXI 4 Licensed or free ESXI? I want to say your version is too old. I'm currently running ESXI 6.0 update 3 at home and Asterisk in a VM under debian without issue. Doug --

Re: [asterisk-users] OT: DMARC enabled domains on this list

2017-06-02 Thread Doug Lytle
>>> On Jun 2, 2017, at 4:19 PM, Daniel Tryba dan...@tryba.nl wrote: >>> Having enabled a strict DMARC setup I noticed everytime I send a message >>> here I get all these reports of messages which fail DMARC. Since I don't >>> want people to miss my wise thoughts maybe the maintainers of this

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

2017-05-10 Thread Doug Lytle
>>> I ask my SIP provider to include more headers to show the real ANI? What >>> would that service be >>> called? If it's anything like a PRI provider, I've been told they only way to get true CID, in those instances, would be to provide a 1-800 number (US) for them to call. Then you'd get

Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

2017-04-19 Thread Doug Lytle
On 04/19/2017 04:32 AM, Irfan PHEERUNGGEE wrote: asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Instructions at the bottom of each email Doug --

Re: [asterisk-users] E-911

2017-03-02 Thread Doug Lytle
>>> On Mar 2, 2017, at 12:33 PM, Jeff LaCoursiere j...@jeff.net wrote: >>> Can anyone point me in a direction to start implementation of E-911 >>> services? Is this just something my upstream should supply, or can I >>> connect to something on my own? For me, it was a pay for option from my

Re: [asterisk-users] Asterisk 13.13.1

2017-01-30 Thread Doug Lytle
>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote: >>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from >>> here: >>> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz >>> >>> I continue to see errors like this: >>>

Re: [asterisk-users] Dial() from the console?

2017-01-11 Thread Doug Lytle
>>> On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: >>> Can I dial directly from the asterisk console with the Dial() application? console dial number@context Doug -- _ -- Bandwidth and Colocation

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