similar to: PRI Down but zaptel lets calls through

Hi All I am testing Asterisk 12 and got most things working, but cannot get a trunk setup working. I am using the new pjsip channel driver. The provider provides IP security so no registering or credentials are required. This is version 1.8 that works [maintrunk] type=peer host=1.2.3.4 disallow=all allow=g729,alaw,ulaw and use Dial(SIP/maintrunk/${ARG1}) My version12 [udpnonat]

Hi all, I'm working on a setup for a small office. I'd like to use SIP/iconnecthere most of the time, because they're cheap. But they only allow a single call. When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse instead: exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN} Well,

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

Hello List: Recently our shorewall FW server went dead (PS failure) & brought the entire system down. Luckily we are testing the FW and other servers, so we did not loose anything. Now we have decided to setup two Shorewall FW servers with a primary & another fallover FW server. I have done some research cruised the Internet and found that a product ''UCARP''

Halo, (First time posting, please forgive English is not native) Change from Courier to Dovecot 2.1.16 Having two server. Having mysql on thiz two server, one master, one slave. What we wish is slave Dovecot only ask slave mysql, unless slave mysql not work when then ask master, we have Postfix do thiz fallover good, but Dovecot talk to slave and master no mater what, we think thiz defeat

Good morning, On Mon, 2023-06-26 at 16:29 +0200, Michel van Dop wrote: > Hello, > > I use/test now a log wrapper. > He checks which log has already been sent. In this way it only sends > the new data it is fast and almost realtime. > > This was the link that not work about log rotation in old mailing i > found: > https://trac.xiph.org/ticket/2173 This is most

I may have found a bug... I am not sure. I am new to Icecast, so this might be how things are meant to be, I have a Debian install with Icecast2 install from APT - Icecast 2.3.2 I have three mount points (all 128kbp mp3) /EDH.mp3 (public = 0 and hidden = 1) /live.mp3 (fallback-mount = /EDH.mp3 and fallback-override = 1) /dj1.mp3 (fallback-mount = /EDH.mp3, fallback-override = 1, public = 0, and\

Hi Philipp, Thank you, thats good news that the log on date rotation is in the new version 2.5. And can you indicate what you are waiting for before making it the stable 2.5 release? And in which version of openssl does this problem no longer occur? Sorry for straying from the original question :-) Best regards, Michel > Op 26-06-2023 17:11 CEST schreef Philipp Schafft <phschafft at

Hi all, i am trying to setup my first t1 in asterisk, i have been using asterisk for several years but ahve never needed a t1 line before. I have a sangoma card already in the server with 4fxo ports. Now i ordered two single port t1 line cards from sangoma for the two servers i am connecting with the point 2 point t1. I am currently at the location trying to set things up but have some

Hi All It''s about time to upgrade my shorewall routers again so thinking of possible changes. In our main office have managed switches and 5 xen servers. I''m thinking of running the firewall/router under xen. Don''t have all the details figured out but this is roughly what I''m thinking of: Set up separate vlan for the two isps and plug isps into the switch.

Good morning, On Tue, 2023-06-27 at 09:34 +0200, Michel van Dop wrote: > Hi Philipp, > > Thank you, thats good news that the log on date rotation is in the > new version 2.5. keep in mind that the ticket you linked is about dumpfile rotation. log file rotation is the same as it always has been. As this is a very common thing being asked if I may just include: When you want to

Hi, I'm facing a strange issue while doing a migration from an hypervisor to another one. The migration takes for ever to start moving the memory. The VM had no workload what so ever, just a basic ubuntu image. The versions on the hypervisors are: libvirt 1.2.21, qemu 1.2.3 Command to launche the migration: virsh migrate --verbose --live --abort-on-error --tunnelled --p2p --auto-converge

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Hi Philipp, Okay that's also a nice solution rename the log file and reload icecast. Only it may be that the file size has been reached, turning off archive then is this option is the best way! Thanks The wrapper works well so far. And also retaining the original log files. I check the the Linux machine for the ssl problem for 2.4.4 and 2.5 beta. It is Debian GNU/Linux 10 (buster) OpenSSL

On Wed, Jun 01, 2016 at 03:55:37PM +0200, Peter Krempa wrote: > On Wed, Jun 01, 2016 at 11:59:29 +0200, Marc-Aurèle Brothier - Exoscale wrote: > > Hi, > > > > I'm facing a strange issue while doing a migration from an hypervisor to another one. The migration takes for ever to start moving the memory. > > The VM had no workload what so ever, just a basic ubuntu

I just was looking at the CVS nightly build of the icecast2\src\source.c and see that the mountpoint fallback is now implemented (compared to the 2.0.0 release's source.c) I was hoping to use this feature to support a multi-home radio station that has a master mountpoint stream and offsite "guest" mountpoints. These guest mountpoints would attach on a scheduled based and,

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up) exten =>

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically