similar to: New feature: Asterisk Manager Interface commands for DeviceState

Hi, I have read that DAHDI automagically turns of echo cansellation when it sees that it is a FAX. So I checked this out. I have a fax call into asterisk which is immediately called out to an external fax machine via DAHDI again.. For example, the result is: DAHDI/1-1 = incoming call, DAHDI/2-1 outgoing call. Now, with the help of dahdi show channel, if I check channel 2: echo cancellation is

Hi, I'm now thinking of always dialing out to Local/xxx at outbound/n on all my queue members. The reason for this, is both to be able to limit the number of calls to one agent, and to have fail-over-lines on the agents. (for example, if dahdi fails, go sip) But for a few years ago, I did some testing with Local/ channels, and they seemed somewhat unstable in large quantity. Are they more

Hi, Is it possible to have MOH in early audio, while waiting for someone to pick up a Dial() call? (When using zap channels, I have early audio working with playback) H?kon Nessj?en Loopback Systems AS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/c901ff90/attachment.htm

I want to create a script for IVR that compiles responses, aggregates them to a total number. Then, run an equation based on the result. Press 1 for X (X is a positive number 500) Press 2 for Y (Y is a positive number 200) Press 3 for Z (Z is a positive number 300) Press 20 to calculate the results = 500+200+300 =1000 then, exten => s,n,Read(NUMBER,,1000) exten => s,n,SayDigits(${NUMBER})

Hi, Is it possible to have early audio while waiting for answer in a Dial()? Say that I want to do this: 1,Progress() // Establish early audio possibillites 2,Dial(SIP/user,20,z(repeated-musicfile)) Where z would be like a function for playing early audio. Or z would just start MOH without Answer()'ing first. H?kon Nessj?en Loopback Systems AS -------------- next part

Hi, I've just uploaded a patch here: https://issues.asterisk.org/view.php?id=16925 This patch introduces a new parameter; "congestion" to both RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to 1 if the call failed to go through because of technical difficulties. And it also is more verbose than app_queue has been earlier, since app_queue usually silently

Hi, This is the output from queue show 28: 47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet Why is the devicestate "Ringing" when no channels is calling this number, and the queue says "has taken no calls yet"? Is it picking up the general state of a random channel on g0 in dahdi? Or what is happening? It only seems to happen with this particular

System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing

Hi all, we do have a callcenter system running with 1.4.21.1 - the agents are connected used sip phones. SIP accounts are configured using realtime (sip buddies) - and are configured with call-limit=1. It is operating just fine - but from time to time it does happen that an agent with an active call (inbound or outbound) does start to get a second call offered. I have taken a look at the

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

I need to reliably boot a server from a USB device. Since USB device ordering can be unpredictable, I wrote a simple early-userspace "init" program to find the root FS on the correct USB device & partition. I have "labeled" the root ext2 FS on the USB drive, and the program searches for the label by examining the bytes at a specific offset from the beginning of the

I installed wine version 0.9.24 and winetools on my computer (OpenSuse 10.2) but I can't get it to configure at all. I first tried winetools and it said that I didn't have a configuration so run with the command wt. So I typed wt and nothing happened. Heres where I think the main problem lies though. When I attempt to do winecfg I get a ton of errors. Here is the output:

On Sunday 25 October 2020 at 16:27:00, Antony Stone wrote: > Hi. > > I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and > for some reason it's simply not doing it. I've made a bit of progress - I can now get it to authenticate, although it's still not dialling on to the correct number. > I've even resorted to reading the source code

Hi. I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1 at least). According to the documentation in channels/chan_sip.c the Dial() string syntax is: * SIP/devicename * or SIP/username at domain (SIP uri) * or SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] * or SIP/devicename/extension * or SIP/devicename/extension/IPorHost * or

Hello, after a fresh install, wine 0.9.10 wont start while giving the following errors: flo@HAL2000 ~ $ wine can't create key_t for shm: No such file or directory wineserver: chdir /home/flo/.wine : No such file or directory winecfg also wont start: flo@HAL2000 ~ $ winecfg wine: creating configuration directory '/home/flo/.wine'... Can't open configuration file

Hi, I would like to be able to configure VMs running off dm-crypt devices that were unlocked in the host. Unlocked dm-crypt devices show up in /dev/mapper/devicename, with devicename being the second parameter given to cryptsetup luksOpen. The LVM storage pool type insists on searching in /dev/vgname and cannot be tricked into reading /dev/mapper by giving it a fake VG named mapper; the LVM

match will expand to guestfs___match, rename the local variable to avoid clash. Signed-off-by: Olaf Hering <olaf at aepfle.de> diff --git a/src/inspect-fs-unix.c b/src/inspect-fs-unix.c index 06ff96d..c30ad5a 100644 --- a/src/inspect-fs-unix.c +++ b/src/inspect-fs-unix.c @@ -1128,14 +1128,14 @@ map_md_devices(guestfs_h *g, Hash_table **map)

Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or

Hi, I'm new to the list, and would like to make a contribution to nut. Nut does a superb job of monitoring the UPS(s) that power servers, but your server may want to monitor other UPSs as well. You may have a SAN attached disk array and want to monitor its UPS(s) in order to shutdown yourself before the disk dies and you lose data. There may be other devices as well that would cause you

Ishfaq Malik <ish at pack-net.co.uk> schrieb: Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default