similar to: UK issue - Asterisk dialling 999... sort of

Hello Happy New Year to all!! I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving over configuration and scripts where needed. All went surprisingly well! Anyway, one lingering issue is that the function key "lamps" on our Snom phones have all stopped working! We're using a mix of Snom 290/320/360 phones and

Hi I have a situation where I want to be able to count how many times a caller goes round a loop of "Please hold...", "please continue to hold". I have found an example on voip-info but I can't get it to work. Not sure if I've got some syntax wrong somewhere? All that happens at the moment, is I hit is the playback of "som-debug" at 9999. Any ideas would

Hello Looking for some help with a rather odd problem. We have Asterisk 1.4.10 running on a Linux box, within our Windows domain. Our Domain Controller is a Windows 2003 server, providing the normal Windows domain functions, such as DHCP and DNS. When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal

Hello Does anyone have any experience with configuring BT (British Telecom) ISDN2e lines to work with Patton SmartNodes - and then Asterisk? I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e lines - and in turn connected to our internal LAN. I'm having huge issues configuring the SmartNode to successfully "see" the ISDN channels - and to be honest, I'm

Hello Does anyone have any experience with configuring BT (British Telecom) ISDN2e lines to work with Patton SmartNodes? I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e lines - and in turn connected to our internal LAN. I'm having huge issues configuring the SmartNode to successfully "see" the ISDN channels - and to be honest, I'm lost as to how to then

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've

Any word about this issue? Should I file a bug in an actual bug tracker or something? //Mike On Sat, 15 Aug, 2020 at 13:26, Michael Gratton <mike at vee.net> wrote: > Hi all, > > I'm seeing Dovecot include the string "MISSING_DOMAIN" in fetch > envelope requests when an mailbox's `addr-spec` part does not have a > `domain` part. > > For example:

On Tue, 1 Sep, 2020 at 09:59, Timo Sirainen <timo at sirainen.com> wrote: > On 1. Sep 2020, at 6.24, TACHIBANA Masashi <tachibana at qualitia.co.jp> > wrote: >> >> Hi, >> >> Is this expected or not? >> >> From: user1 at fuga.example.com <user1 at example.com> >> To: user2 at hoge.example.com <user2 at example.com> >>

Hi all, I'm seeing Dovecot include the string "MISSING_DOMAIN" in fetch envelope requests when an mailbox's `addr-spec` part does not have a `domain` part. For example: > C: a022 uid fetch 40 (envelope rfc822.header) > S: * 5 FETCH (UID 40 ENVELOPE ("Sat, 15 Aug 2020 12:53:05 +1000" > "test {{name}}" (("Michael Gratton" NIL

Hey all, I'm getting an error starting a libvirt managed qemu/kvm VM at physical host boot time, but manually starting it afterwards works fine. This is on a Ubuntu Wily i7-4790 box running Linux 4.2 and libvirt 1.2.16. There is a legacy (5V) PCI card being passed through to the VM, the error seems to relate to that. The error that always appears at boot in

Hey Mike, did you ever have any success figuring out this issue? I am having the same problem (on CentOS 7) with a custom data acquisition PCI card. It seems like the solution might have something to do with delaying the start of libvirtd in systemd until the PCI card (or vfio?) is ready, but I don’t know how to do that. Can anyone else offer assistance? My output: 2016-03-29T18:29:26.191010Z

This reverts the change made for RHBZ#1168144. The warning is now always displayed. It would be nice to make the warning actionable, but there is not a lot that end users can do since fstrim is such a complex topic interacting with all filesystem and storage layers. --- v2v/v2v.ml | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/v2v/v2v.ml b/v2v/v2v.ml index

On Fri, Aug 12, 2016 at 04:50:31PM +0200, Pino Toscano wrote: > On Friday, 12 August 2016 10:37:29 CEST Richard W.M. Jones wrote: > > This reverts the change made for RHBZ#1168144. The warning is now > > always displayed. > > > > It would be nice to make the warning actionable, but there is not a > > lot that end users can do since fstrim is such a complex topic

>>>>> Rich Shepard >>>>> on Mon, 22 Jan 2024 07:45:31 -0800 (PST) writes: > A statistical question, not specific to R. I'm asking for > a pointer for a source of definitive descriptions of what > types of data are best summarized by the arithmetic, > geometric, and harmonic means. In spite of off-topic: I think it is a good

On Mon, 22 Jan 2024, Martin Maechler wrote: > I think it is a good question, not really only about geo-chemistry, but > about statistics in applied sciences (and engineering for that matter). > John W Tukey (and several other of the grands of the time) had the log > transform among the "First aid transformations": > > If the data for a continuous variable must all be

Hello, I'm experimenting with Overlap Dialling in asterisk 1.6. I've enabled this in sip.conf and on the SNOM 300 phone. My problem is that asterisk dials out as soon as it matches an extension without waiting to see if the user is going to type in more digits. Is there a way to set a timeout per channel or globally? I'd like Asterisk to wait for a few seconds once its found a match

what is the format for an h323 entry in the dialplan? can I use chan_h323 without compiling anything else or should I compile oh323? basically what's the best way :) cheers Dave

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite