Re: [asterisk-users] Asterisk 13 / chan_sip / registration after reject

Hello Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon ) a écrit : Hi List We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure: auth_rejection

Re: [asterisk-users] Any thoughts on Asterisk 16.17.0 outputting FRACK refcount related messages

I don't *think* it would purely volume related. We have 16.17 deployments with very large loads running without issue, and we also run 16.17 against load simulators without issue. In each case you have to traceback to find the cause of the problem. For example, a bad SBC which does not fully

Re: [asterisk-users] Asterisk bring in RTP audio

Turn you 16 RTP port device into a SIP UA. Use one of the open source SIP phones as starting point, setup as autoanswer, and start streaming the RTP. High level answer for high level question…but that should point you In the right direction From: asterisk-users [mailto:asterisk-users-boun

Re: [asterisk-users] recording not working to NFS

Just adding my 2c I don't think permissions which cause one process to see the mounted file system and another to see the directory underneath. I think using automount could cause this but there is still some other factor contributing to the problem. -Original Message- From: aster

Re: [asterisk-users] recording not working to NFS

Asterisk Users Mailing List - Non-Commercial Discussion Cc: Telium Technical Support Subject: Re: [asterisk-users] recording not working to NFS I did not explain myself well, for this I apologize. The files never appear on the NFS mount, only in the local drive. Restarting Asterisk with the mount on

Re: [asterisk-users] recording not working to NFS

If unmounting makes your files appear on the NFS mount, then there may be some caching going on, or files not being closed (by Asterisk). Unmounting will force files to close and could make them appear. Try restarting Asterisk (with NFS still mounted). Do the files then appear? -Origina

Re: [asterisk-users] SIP Source Port

I don’t think I’ve seen that requirement before, so someone else may have to answer if there is a PJSIP specific setting However, if not then it may be simple to achieve the same result by using your firewall NAT rules. From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com]

Re: [asterisk-users] Hook Flash

Since this function is handled by the ATA, you would have to look there (or post details) for something ATA specific. In general I don’t think so, hook flash just puts one channel on hold a creates/answers another. But, you may be able to script the functionality you need it in the Ast dialpla

Re: [asterisk-users] Server loses sip registrations after converting to vm to mysql storage.

How about starting a console with verbose turned up. After a loss of registrations review the console output to see if there is some event. From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Tuesday, April 20, 2021 3:54 PM To: Asterisk Users Mai

Re: [asterisk-users] STIR/SHAKEN

If you operate a small PBX for a business your approach is fine. If you operate a large PBX, or just have lots of high toll rate calls, the price difference between carriers can add up to a lot money every day. These operators will route their calls to whomever offers the best rate for that

Re: [asterisk-users] Failed to authenticate device message

You didn’t post the Asterisk version, but if this is an OLD asterisk version then the source IP may be missing from messages/logs. If you have low traffic in general then using something like Wireshark may help you examine any suspicious SIP packet on the PBX. For higher volumes it’s like d

Re: [asterisk-users] Stir Shaken

This sounds like the kind of business I can trust with my calls, and am eager to buy from. Oozing with professionalism. Well done sir! :) From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of d...@donkelly.biz Sent: Tuesday, July 14, 2020 4:48 PM To: 'A

Re: [asterisk-users] Voice broken during calls (again...)

Still lots of detail missing, butlikely causes include: 1. Egress latency (does your router/firewall support QoS, are you leaving headroom ) 2. Ingress latency - does your ITSP support it 3. Router/firewall latency - can it keep up with the traffic and packet size. Do you have way too many

Re: [asterisk-users] Voice broken during calls (again...)

I don't know if there was a prior email with more details, but Latency is as important as speed. Have you checked latency between your device and pop? What about QoS at your location, and does your ITSP support/respect QoS? Could problem be inside your network? Have you tested/opti

Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

Just run ‘core show calls’ as a command from the AMI, and parse the results. I don’t think there is an equivalent pure AMI command. From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan H Sent: Sunday, June 14, 2020 5:45 PM To: Asterisk Users Mailing Li

[asterisk-users] Send message to AMI from dialplan

Is it possible to simply send a message to appear as an AMI message/event, from the dialplan? For example exten =>123,1,ami(myEvent, param1, param2) and in the AMI a message appears like: Event: myEvent Privilege: call,all Channel: PJSIP/misspiggy-0001 Uniqueid: 1368479157.3 Ch

Re: [asterisk-users] CLI color prompt

That means that Asterisk is not echoing the escape character (27) to your terminal. Try different escape formats (octal, slash prefix, etc) -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fourhundred Thecat Sent: Sunday, May 31, 2020

Re: [asterisk-users] CLI color prompt

Have you tried adding ANSI color escape codes? There's lots of documentation for BASH prompt color using escape codes. Give those a try. (I haven't tried it, but would make sense) -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fou

Re: [asterisk-users] Troubleshooting load issues

be causing the spike? On Wed, Apr 22, 2020 at 2:21 PM Telium Technical Support mailto:supp...@telium.io> > wrote: Could some calls be arriving with a different codec? (Is transcoding causing the spikes)? Are you limiting codecs to match your audio files? From: asterisk

Re: [asterisk-users] Troubleshooting load issues

Could some calls be arriving with a different codec? (Is transcoding causing the spikes)? Are you limiting codecs to match your audio files? From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Wednesday, April 22, 2020 2:01 PM To: Asterisk U

[asterisk-users] Compile Asterisk without CPU specific extensions/optimizations

I'm compiling an Asterisk system on a ESXi VM with recent CPU, but will deploy onto an old ESXi VM with older CPU. Is it possible to configure Asterisk to NOT use CPU specific instructions/optimizations so that the executable is portable? Thanks Dan (in learning mode) -- _

Re: [asterisk-users] Load issues using AGI

solid and really doesn’t need much maintaining. I’ll probably write the author and see if I can keep it updated for him. Regards; John From: asterisk-users On Behalf Of Tech Support Sent: Friday, September 20, 2019 1:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discu

Re: [asterisk-users] Load issues using AGI

as simple as optimizing a single subroutine. Then you will know exactly what the problem is. (2)See which version of Perl are you running and see if upgrading it solves your problems. The easiest way is to download the newest Perl they support from ActiveState.com. It creates a completely

Re: [asterisk-users] Lightweight ODBC DB

IF you use the HAAst or PBXSync solution, you can include/exclude at the table and database levels. You can also use SQLite if the data is suitable (and these products can sync SQLite too). If you want a non-commercial solution, MySQL’s log rolling may be most suitable. From: aster

Re: [asterisk-users] Lightweight ODBC DB

Have you looked at PBXSync (or HAAst) from Telium? (https://telium.io) These products will sync MySQL, SQLite, plus files, directories, etc. intelligently. (Differentials only) between PBX’s, reload configurations on the fly, etc. No need roll logs or recover from a base in case they get t

Re: [asterisk-users] Does Asterisk cache AstDB?

3:34 PM, Telium Technical Support wrote: > Is the Asterisk internal database cached by Asterisk? Or is it always > reading/writing to the SQLite database? (If I read from the SQLite DB > is it sure to match what Asterisk is using) There is no additional caching built into Asterisk itse

[asterisk-users] Does Asterisk cache AstDB?

Is the Asterisk internal database cached by Asterisk? Or is it always reading/writing to the SQLite database? (If I read from the SQLite DB is it sure to match what Asterisk is using) -- _ -- Bandwidth and Colocation Provided b

[asterisk-users] using CIDR for hosts entry in sip.conf

I am setting up a system with a large number of trusted trunks (by IP). I find that I have to make one entry sip.conf for each trunk becauses the host= line requires a single IP. Does asterisk support a CIDR or wildcard or multi-ip format for the host= line in sip.conf

Re: [asterisk-users] Sending SMS and SIM card

I use VoIP Innovations and ThinQ (formally SIPRoutes) and they both support SMS. That way it’s very easy to write it into the dial plan. Regards; John From: asterisk-users On Behalf Of bilal ghayyad Sent: Thursday, April 25, 2019 7:56 AM To: asterisk-users@lists.digium.com

Re: [asterisk-users] Dialplan reload from AMI

Does reloading pbx_config ONLY reload the dialplan? Or is something else reloaded too? This sounds like a preferable way to do it From: Ian McMaster [mailto:ian.mcmas...@gmail.com] Sent: Saturday, April 20, 2019 1:19 PM Subject: Dialplan reload from AMI Rather than Action: Command

[asterisk-users] Reload dialplan from AMI

I see there is a modulereload function available from the AMI, but none of the listed modules (on the wiki) seem to reload the dialplan. Is there a way to reload the dialplan through this function? Or do I have to use the 'command' action? --

Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us

This is usually a symptom of something on the call path mishandling the session setup. Check routers/firewall/SIP proxy, etc. Likely a firmware bug is causing it to use the wrong IP address and passing that to the other end. Even if you disabled these devices, REMOVE them from the call path

[asterisk-users] Problem with AudioCodes MP-114 ATA

All; I have an AudioCodes MP-114 four FXS ATA that recently stopped registering to my PBX. I'm pulling my hair out here trying to figure out the root cause without much success. Does anyone have a sample config file that I could use as a sample? Any insight at all would be greatly appreciated.

[asterisk-users] Disabling a trunk at runtime

I have an Asterisk system with 2 trunks (as shown below). I need to be able to disable a trunk at runtime. I may not change the dialplan but I can change sip.conf and reload. Any attempt to dial in the dialplan uses trunk A and trunk B in that order. Normally calls will route through trunk A,

[asterisk-users] Call Queue Data

All; A few years back, we put a heck of a lot of effort into developing a software package to analyze call queue data that we want to open source. It's a pretty good package and I would like to dust it off and resurrect it. What I need to do that is have sample call queue data to test with. If

Re: [asterisk-users] AGI timeout option

Hello; I’ve been using AGISpeedy for Perl for years. I just works, and it works really well. I’m a Perl programmer, not a PHP programmer, but it seems to me that you need the PHP equivalent of Perl’s ‘alarm’ command. Just a thought. Regards; John V. From: asterisk-users [mailto:asteri

[asterisk-users] Asterisk 16 AMI changes

Does anyone know if Asterisk 16 includes changes to the AMI? (syntax / commands / etc) I see a release candidate is forthcoming. Just curious -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astri

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

: [asterisk-users] getting invites to rtp ports ?? On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group mailto:supp...@telium.ca> > wrote: Depending on log trolling (Asterisk security log) misses a lot, and also depends on the SIP/PJSIP folks to not change message structure (whi

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

: Wednesday, August 29, 2018 6:33 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] getting invites to rtp ports ?? On 08/29/2018 11:59 AM, Telium Support Group wrote: > Block a single IP is the wrong approach (whack-a-mole). You should consider > a more comprehensive ap

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

Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki: https://www.voip-info.org/asterisk-security/ -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@

[asterisk-users] Passing arguments to the 'mailcmd' option in voicemail.conf

All; I'd like to change the default command that is used to send email when a person has a new voicemail. I believe that's set in voicemail.conf as the 'mailcmd' option. The default is to use the /usr/sbin/sendmail -t command. I wrote a quick test script to see what arguments are passed to the

[asterisk-users] Core show channels concise = deprecated

I want to get a list of all active channels from the AM. I've been using 'core show channels concise' (as a commands from the AMI) but I see in the documentation that the command is deprecated and will be removed. What's the best way to get the equivalent from the AMI? -- ___

Re: [asterisk-users] Pass through registration / proxy

ium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba Sent: Wednesday, April 11, 2018 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pass through registration / proxy On Tue, Apr 10, 2018 at 09:22:02PM -0400, Telium Techni

Re: [asterisk-users] Pass through registration / proxy

:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pass through registration / proxy Hi You could use kamailio +asterisk On Tue, Apr 10, 2018, 9:25 PM Telium Technical Support mailto:supp...@telium.ca> > wrote: I need to create a SIP

[asterisk-users] Pass through registration / proxy

I need to create a SIP proxy to be placed in front of a legacy PBX. When a phone registers with the proxy, I would like Asterisk to register with the PBX behind it. (To tell the PBX to send calls to the proxy and then to the SIP phone). Can I use Asterisk to create a proxy like this? Is ther

[asterisk-users] Looking for C library for the Asterisk AMI

All; We do a lot of programming and customizations for Asterisk and normally, we do everything in Perl. For that, we use the CPAN module Asterisk::AMI, and it works great. However, we have several programs that would benefit greatly if they were written in C. So my question is this: Does anyon

Re: [asterisk-users] Blacklist failed attempts

If this is a home system, try the free edition of SecAst (www.telium.ca/?secast ). If allows you to set thresholds for the number of attempts, and specify the period in which they occur. The Free edition of SecAst is a drop-in replacement for fail2ban (but with a

Re: [asterisk-users] Search for (multi tenant) fax to mail solution

s; John John V. John V., Tech Support VoIP Business Solutions 240-215-3479 x325 <mailto:f...@voipbusiness.us> supp...@voipbusiness.us -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael

[asterisk-users] Getting the number of parked calls in a parking lot

All; With Asterisk 11, it was trivial to get the number of parked calls in each parking lot simply by issuing the "parkedcalls show" command. However, with Asterisk 13, things are done very differently and the "parkedcalls show" command no longer exists. So my question is, how do I get that in

Re: [asterisk-users] AMD and fax detection

I think that there is a much easier way to detect if the far end is a fax. In your dialplan, include a section that uses the 'fax' extension. If the call jumps to that extension, then the far end is a fax machine. Regards; John V. -Original Message- From: asterisk-users-boun...@lists.digiu

[asterisk-users] Forwarding a call "off pbx"

All; I had someone ask me if they received an incoming phone call and it was forwarded "off pbx" to their cell phone, would the call be strictly between the caller and the cell phone, or would it between the caller, the pbx, and the cell phone where the VoIP minutes continue to be charged. Rat

Re: [asterisk-users] Can't compile Asterisk on Fedora server

t: Re: [asterisk-users] Can't compile Asterisk on Fedora server On Wed, Jan 10, 2018 at 10:28:42AM -0500, Tech Support wrote: > > > All; > > I have a Fedora 26 server that I am trying to compile > asterisk-certified-13.13-cert6 on. However, I'm getting the following

[asterisk-users] Can't compile Asterisk on Fedora server

All; I have a Fedora 26 server that I am trying to compile asterisk-certified-13.13-cert6 on. However, I'm getting the following errors. I'm also having a tough time trying to compile Dahdi. I'm not sure what I'm missing, but if anyone else is running Fedora, I'd really appreciate any help

Re: [asterisk-users] Showing CallerID on multiple phones

CallerID on multiple phones What kind of phones are you using? To some phones you can push text dialog over HTTP/XML. Investigate this features, maybe it is what are you looking for. TH Dne pondělí 11. prosince 2017 15:16:28 CET, Tech Support napsal(a): Hello; I certainly

Re: [asterisk-users] Showing CallerID on multiple phones

call. If no one picks it up by ring x, have it go to another phone or to voice mail. https://www.freepbx.org/ring-group-and-follow-me-ring-strategies-1-of-2/ might be useful. Ron On 08/12/2017 2:17 PM, Tech Support wrote: All; I have an interesting scenario where I have a small office

[asterisk-users] Showing CallerID on multiple phones

All; I have an interesting scenario where I have a small office with maybe half a dozen phones and several incoming lines. The calls are routed based on the DID that people call. What they would like is when a call comes in to a single phone to have all the phones show the CallerID. That way t

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

That was it. I can't thank you enough. Regards; John V. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, November 14, 2017 09:27 AM To: Asterisk Users Mailing List - Non-Commercial Disc

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

All; I'm having a problem with parking a call and I'm hoping that someone has seen this problem before. A call comes in and I park it. A few seconds later, I retrieve the call. So far, so good. The problem lies when I go to park the call again, and nothing happens. I'm running Asterisk 11. Any

Re: [asterisk-users] Looking for the carrier that owns a particular DID

users] Looking for the carrier that owns a particular DID On Thu, 2017-11-02 at 11:33 -0400, Tech Support wrote: > How do I find out which carrier owns the DID in question? Try here: https://www.twilio.com/lookup -- _ -- B

[asterisk-users] Looking for the carrier that owns a particular DID

his number is owned by another carrier that we do not have a business relationship with." So my question is this. How do I find out which carrier owns the DID in question? Thanks; John V. Tech Support Tech Support VoIP Business Solutions 240-215-3479 x325 <mailto:f...@voi

Re: [asterisk-users] What version of Linux?

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: Monday, August 28, 2017 03:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] What version of Linux? Hello Asterisk, I've been running CentOS si

Re: [asterisk-users] Detecting DoS attacks via SIP

s provided by a few of you guys. It was mentioned that this is a broad hammer, but I'm kinda looking for a broad hammer! ;^) Looks like I need to do some research, but I think I have what I need. Thanks again, Mike Diehl. On Sat, Aug 19, 2017 at 4:36 PM, Telium Technical Support ma

Re: [asterisk-users] Detecting DoS attacks via SIP

say this conversation you have posted is a bit outdated, now fail2ban can be used with asterisk security log https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger. On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support mailto:supp...@telium.ca> > wrote: Keep in mind that

Re: [asterisk-users] Detecting DoS attacks via SIP

Keep in mind that the attacks you are seeing in the log are ONLY the ones that Asterisk is detecting and rejecting. All other attacks aren't even showing up! There's a good discussion of how to secure your PBX here: https://www.voip-info.org/wiki/view/asterisk+security In general, don't let the

[asterisk-users] How are people billing for minutes?

All; We have always tried to avoid charging customers for minutes simply because we didn't want the hassle of doing the accounting. I was wondering what software packages or services people are using for this. Best Regards; John V. -- __

Re: [asterisk-users] Writing CDR's to two database servers

Virtual Server) HA machine in front of the two, so that writes can go to either server using only a single IP address configured in Asterisk. Then, if one fails, you can still write to (and read from) the other, repair the failed one, and restore replication. Antony > > On Jun 19, 2017

[asterisk-users] Writing CDR's to two database servers

All; I know that there are probably several solutions to this problem, but what I am trying to do is provide some redundancy for my customers CDR data. I know that doing simple backups of MySQL is probably the easiest way to go, but I'm thinking that there may be some benefit to simultaneously

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

Just a guess (without knowing about your network), but are the two ends points on public networks and visible to one another? If not the reinvite may be passing an internal (nat'ed) address to the other and the connection will fail...just a though -Original Message- From: asterisk-users-b

Re: [asterisk-users] Automatically dial a number, then an extension

: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Automatically dial a number, then an extension On Tuesday 23 May 2017 at 20:01:14, Tech Support wrote: > Ok, the purpose of the answering machine detection (AMD) is to > determine when the audio file

Re: [asterisk-users] Automatically dial a number, then an extension

users] Automatically dial a number, then an extension On Tuesday 23 May 2017 at 19:20:25, Tech Support wrote: Isn't it safe to assume that if you've been given an extension number to dial after the initial call is answered, then it wasn't answered by an answering machine

Re: [asterisk-users] Automatically dial a number, then an extension

for the extension to pick up. Simply placing the AMD command after the SendDTMF() wasn’t the answer I don’t know how to approach this problem. Thanks; John V. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support

Re: [asterisk-users] Automatically dial a number, then an extension

using AMI? Or AGI maybe? Or Call files? What Asterisk version do you have? El 15 may. 2017 12:35, "Tech Support" escribió: All; I have an application that dials a list of numbers and then plays a recorded message. My customer uses it to dial a list of customers to conf

[asterisk-users] Automatically dial a number, then an extension

All; I have an application that dials a list of numbers and then plays a recorded message. My customer uses it to dial a list of customers to confirm their appointment for the next day. No biggie, maybe 25 - 30 calls per day for customers who want the confirmation call. What they need now is a

Re: [asterisk-users] iaxModem pickup problem

good idea to keep a backup astdb on the PBX in case of corruption. -Original Message- From: James B. Byrne [mailto:byrn...@harte-lyne.ca] Sent: Thursday, May 4, 2017 12:29 PM To: Telium Technical Support Cc: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] iaxModem pickup pro

Re: [asterisk-users] iaxModem pickup problem

It depends a bit on your version of FreePBX, but here's a link to show you how: http://telium.ca/pages/forums/viewtopic.php?f=7&t=19 Hopefully option 1 works for you (quick and easy). If not, you'll have to try option 2. Ignore option 3 since that's only for users of High Availability for Aster

Re: [asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16

Sorry. I wasn’t clear in my post but I was talking about installing both certified 11 and 13. The reason I am trying to install 11 is for support for a couple of legacy systems. Any insight at all into the original question would be greatly appreciated. Thanks; John -Original Message

Re: [asterisk-users] softphone instead of desktop phones

I thought this was a non-commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Epshteyn Sent: Saturday, April 29, 2017 08:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subjec

[asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16

All; I'm trying to install certified asterisk 11.6 cert16 on a Ubuntu 16 server. However, when I try to compile it, I'm getting hundreds and hundreds of errors. Here is a sample of the output. make[1]: Leaving directory '/usr/src/asterisk-certified-11.6-cert16/menuselect' [LD] aelparse

Re: [asterisk-users] ** in extensions.conf

Is ** also defined in features.conf? Thanks; John -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, April 26, 2017 05:41 PM To: Asterisk Users Mailing List - Non-Commercial Disc

Re: [asterisk-users] Asterisk download stats

On a similar note, does anyone have any idea as to the total number of Asterisk installations out there? Thanks; John From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Klein Sent: Tuesday, April 25, 2017 10:00 AM To: asterisk-u

Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16 I don't - it just seems to.. work! Try a reboot - it always comes up OK for me. Are you doing "make install"? On 19 April 2017 at 14:19, Tech Support wrote

Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

how you get on. On 18 April 2017 at 13:41, Tech Support wrote: > All; > > I am trying to build and install certified Asterisk 13.13 cert3 on > a Ubuntu 16.04.2 LTS host without much success. I am getting the > following errors when I try to compile. > > > >

Re: [asterisk-users] PBX selection

recommended it to us) > > > > > > B: Commercial > > > > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now > > acquired by a HongKong company now > > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It > > also includes VoIP SDK, W

[asterisk-users] Can't compile Asterisk on Ubuntu 16

fied-13.13-cert3/Makefile.rules:149: recipe for target 'res_pjsip/config_transport.o' failed make[1]: *** [res_pjsip/config_transport.o] Error 1 Makefile:402: recipe for target 'res' failed make: *** [res] Error 2 Has anyone seen this error before? Any insight at all would be greatly

Re: [asterisk-users] AGI Exec Voicemail

Why not use an ALIAS and let sendmail send the email to a distribution group? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, April 12, 2017 1:09 PM To: Asterisk Users Mailing L

[asterisk-users] Tool to restart Asterisk

. Tech Support Tech Support VoIP Business Solutions 240-215-3479 x325 <mailto:f...@voipbusiness.us> supp...@voipbusiness.us -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check o

Re: [asterisk-users] Manager events showing in CLI

Discussion Subject: Re: [asterisk-users] Manager events showing in CLI Ok, Please, check your manager.conf and logger.conf for any clue about debugging options, into the Asterisk configuration directory. El 26 mar. 2017 14:52, "Telium Technical Support" mailto:supp...

Re: [asterisk-users] Manager events showing in CLI

I tried that but it had no effect. Still see things like: [2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining AMI event: Event: SuccessfulAuth Privilege: security,all EventTV: 2017-03-26T13:49:39.407-0400 Severity: Informational Service: SIP EventVersion: 1 Accoun

[asterisk-users] Manager events showing in CLI

I somehow cause AMI events to appear as output in the CLI, and I can't figure out how to turn them off. Can someone offer a command which will suppress AMI events/commands from showing in the CLI? Ron -- _ -- Bandwidth an

Re: [asterisk-users] Large astDB - millions of tuples - issues?

We wrote a call screening (and CID rewrite) app for an ITSP a few years ago. We had to use MySQL as the astDB could not keep up (* was choking – we did dig deeper we just switched to MySQL). I don’t think astDB is the right way to go. If you’re comfortable writing a * func then you might as w

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

boun...@lists.digium.com [ <mailto:asterisk-users-boun...@lists.digium.com> mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Telium Technical Support Sent: Wednesday, March 15, 2017 11:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-user

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

The history of the question is lost (in the mail thread) so I'll jump in based on what I could see in my recent mail and the subject line: -The ASTDB should have no impact on Asterisk service start (which I assume is the problem given the subject line) -If you disabled SElinux the

Re: [asterisk-users] fail2ban Asterisk 13.13.1

terisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support Sent: Wednesday, March 1, 2017 2:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] fail2ban Asterisk 13.13.1 It's poss

Re: [asterisk-users] fail2ban Asterisk 13.13.1

It's possible that you need to increase the value of 'findtime' to something greater than 300 secs. You also may want to set "timestamp = yes" in asterisk.conf so each line in the CLI will be time stamped. Time stamping it will be the definitive determination on whether or not the 'findtime' is

Re: [asterisk-users] Which tool to automatically restart Asterisk ?

Hello; Over time, we’ve built a huge enterprise level monitoring system for our internal and customer PBX’s. Using Nagios as the core, along with Grafana, Graphite, Carbon, Whisper, etc. so we can also create custom dynamic dashboards, we typically monitor over 1,000 different metrics for e

Re: [asterisk-users] Voicemail notification by email is missing CallerID info

LLERID}, on ${VM_DATE},\n${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?so:(originally sent by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n On 18 February 2017 at 16:35, Tech Support wrote: > All; >

[asterisk-users] Voicemail notification by email is missing CallerID info

All; I am running Asterisk 11.6-cert16 and I have voicemail setup so voicemail messages are sent as email attachments. That works fine. However, the body of the email contains the CallerID(name), but is missing the CallerID(num). For example, the email body looks like this: Just want

Re: [asterisk-users] Call List Campaign to an IVR

f you wanted you could leave it ringing for twenty minutes and it would still have the same effect. Kind regards, Matt On Feb 6, 2017, at 12:29 PM, Tech Support wrote: That's the basics, but you have to nail the timing just right. The timing is really important to do it the

Re: [asterisk-users] Call List Campaign to an IVR

ry 06, 2017 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call List Campaign to an IVR > On Mon, 6 Feb 2017, Tech Support wrote: > > We were able to develop a feature to send the call to voicemail about 90% of the time. That way, an

Re: [asterisk-users] Call List Campaign to an IVR

ubject: Re: [asterisk-users] Call List Campaign to an IVR On Mon, 6 Feb 2017, Tech Support wrote: > We were able to develop a feature to send the call to voicemail about > 90% of the time. That way, an end user could (1) not be bothered by > having to answer the call, (2) delete the message w

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