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

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:

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

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 does not fix

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?

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

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

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

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

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:

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

...) Am 22.06.2020 um 17:01 schrieb Telium Technical Support: > 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 > supp

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/optimized

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

[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

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,

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

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

[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] 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:

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

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

, at 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 Asteri

[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

[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] 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

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

Telium Technical Support wrote: > 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). >

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

:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> 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 <supp...@telium.ca <

[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

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

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

ot; filter, as 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 Tech

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

me if I'm wrong, but I would 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 <supp...@telium.ca

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

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:

Re: [asterisk-users] iaxModem pickup problem

ea 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 <supp...@telium.ca> Cc: asterisk-users@lists.digium.com Subject: RE: [asterisk-users]

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=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

Re: [asterisk-users] PBX selection

Have a look at xCally from Xenialabs too – they are particularly popular with call centers (and still asterisk based). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roamer2998 Sent: Tuesday, April 18, 2017 11:00 PM To: Asterisk

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

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

Discussion <asterisk-users@lists.digium.com> 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 Techni

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

[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

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

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

n...@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-users] Having problem getting Asterisk to work on CentOS 7 The

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

Re: [asterisk-users] fail2ban Asterisk 13.13.1

If this is a small site, I recommend you download the free version of SecAst (www.telium.ca ) and replace fail2ban. SecAst does NOT use the log file, or regexes, to match etc.instead it talks to Asterisk through the AMI to extract security information. Messing with regexes

[asterisk-users] Replacing PBX during a call in progress

This was asked many years ago but I thought I would check to see if things have changed. Is it possible to take over a call in progress - using a replacement Asterisk server? In other words, if 2 user agents are connected through an Asterisk PBX, and I tracked the call ID, IP of each UA

Re: [asterisk-users] Issue command to force SIP client to re-register

Thanks - I actually found the SIP notify command before, but the options seem to force checking for new config (in which case reboot), or cold/warm restart. I was hoping to just force a re-register, not reboot. (Which in this particular case is a long interval which I cannot change, so need

[asterisk-users] Issue command to force SIP client to re-register

Is there a way to force a SIP client to re-register using a SIP command (or an AMI command)? If not, is there some other standard way to do so - or would I have to post/get to a web GUI of the phone (unique to each model) to force a reset, etc. -Raj- --

[asterisk-users] AMI version in Asterisk 14

I noticed that Asterisk 14 has changed the output format for some commands (eg: "Output: "). However, the AMI reports version 2.8.0 which is the same as Asterisk 13 Is that considered a bug? Since the AMI output format has changed, shouldn't the AMI version be incremented? (Makes is hard

Re: [asterisk-users] RS485 Audio device

Forget RS485 at that distance (your throughput will be too low). I would suggest you pull a fiber and just create an LAN connection on the end. I’m sure you would have had fun getting some of the old IP over Serial drivers working J From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] RS485 Audio device

This one caught my interest too...more out of curiosity! Keep in mind that RS485 max speed drops to 100kbps after a relatively short distance. And, 100kbps is RAW speed. If you encapsulated your audio stream in that you'd lose another 10%. So why are you doing this? If you are running a 100m

[asterisk-users] SIP show peers content

When I issue a 'sip show peers' command the left most column is titled 'name/username'. Some lines show one item in the column like 123, others show bob/123. Can someone explain the difference? (What does does name vs username mean) And why does 'sip show users' not show a name column title?

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

/bindaddr on reload Hummm, but why It is with that problem? I use UCARP, maybe is this the problem? 2016-08-29 12:17 GMT-03:00, Telium Technical Support <supp...@telium.ca>: > Oh! In that case ignore it. > > Asterisk won't rebind the adapter if you've only changed parameters.

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

> > 2016-08-29 11:30 GMT-03:00, Telium Technical Support <supp...@telium.ca>: >> This shows that asterisk's IAX is already bound to all adapters - so >> that's >> good. Symptomatically does your IAX stop working? Or do you just see a >> warning? >> >

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

isk/asterisk.ctl unix 2 [ ] DGRAM116862050/asterisk 2016-08-26 19:21 GMT-03:00, Telium Technical Support <supp...@telium.ca>: > Could you post the result of "ip addr" command, and "netstat -anp | grep > ast" after t

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

Could you post the result of "ip addr" command, and "netstat -anp | grep ast" after the reload? I suspect something else is going on here... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join

[asterisk-users] How does Ast use IP vs FQDN for SIP header fields

We are working with an ISP that needs Asterisk to place a FQDN name in the SIP 'FROM' and 'INVITE' fields - where Asterisk is currently using an IP address. A SIP trace shows the following from my Asterisk box: INVITE sip:62351155@1.1.1.1 SIP/2.0 Via: SIP/2.0/UDP

Re: [asterisk-users] SPA112 flapping

I would guess conflicting IP addresses. It comes back up at regular intervals, detects the conflict, and shuts down.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Friday, June 17, 2016 5:57

Re: [asterisk-users] asterisk admin interface

I think only PJSIP and MWI support Sorcery – so that likely won’t do what’s being asked for… And reading/writing a flat file should be even easier than learning the ARI -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] asterisk admin interface

>Thanks Raj >You are correct. Is there any open source application in that? Not that I know of – I think it’s getting too simplistic J We created some C++ functions for our High Availability for Asterisk product (HAAst) which modify config files and extensions files, but it’s more work to

Re: [asterisk-users] asterisk admin interface

You don't mention a configuration generator (like Elastix/FreePBX) so I'll assume you are using a plain old vanilla Asterisk installation. In which case all user/endpoint information is kept in config (ini) files, and no user/endpoint manipulation is done through the CLI or AMI. In this case a

Re: [asterisk-users] AMI: check if the user has a Mailbox

I don't think the directory check method is reliable; a user can have a mailbox but if no messages have been left then the directory structure may not exist. Through the AMI you can show peer/user information and I believe it shows you the mailbox associated with user/peer. -Raj- -Original

Re: [asterisk-users] Asterisk load balancing for TCP/SIP and RTP?

Have a look at this page for HA ideas: http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Design There are a lot of tradeoffs in design, and easy to confuse load balancing with HA From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]

Re: [asterisk-users] Pass variable to voicemail script

>If you are talking about the 'externnotify' parameter in voicemail.conf, the variables are passed simply as @ARGV. I'm referring to the mailcmd= setting in voicemail.conf. Asterisk runs this when emailing a voicemail (with attachment) --

Re: [asterisk-users] FAX Detection.

Perhaps use T38 instead? Would make your life a lot easier. (And you can use a T38modem software). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Panic Button SMS Asterisk Integration

We integrated a variety of USB devices with Asterisk. A number of ways to do so…the device interface has a lot to do with the how. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Nighswonger Sent: Friday, February 05, 2016

Re: [asterisk-users] Failed to authenticate device 100

The details of the source IP are available in the asterisk security log (if you have that enabled) – but that particular attack hides its address from the messages file. It’s essential that you secure your PBX; there are options ranging from free to commercial. Have a look at:

Re: [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?

So here I am, asking everyone what router they use, and whether you're happy with the results when there's 100 simultaneous SIP calls in progress. I want to know what happens when the rubber hits the road. On 2015-11-20 14:22, Telium Technical Support wrote: > Well router and firewall are ver

Re: [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?

Well router and firewall are very different...it depends on what you are trying to accomplish. If you are trying to secure an Asterisk-based call center, get a real security product. Look here for details: http://www.voip-info.org/wiki/view/Asterisk+security This covers firewall, Asterisk

Re: [asterisk-users] Asterisk HA with heartbeat and systemd

If you’re still in the planning stage, there’s a lot more to think about. Your Asterisk failure detection will be very simplistic (is the process dead). Synchronization of data – without risking synchronization of corrupt data to a peer. Prevent a deteriorating/failing peer corruption from