similar to: Reservation call on busy

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

Hi everybody, I'm receiving the message "res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!" in asterisk console when I try to put a call on hold. I don't the reason and I'm sure the relative module is loaded. In musiconhold.conf I put these lines, trying something I found in some previous post: ; ; Music on hold class definitions ; [classes]

Here is the output. These are Panasonic KX-TG2564's. Does something need to be set for the phones? I can call out fine, but all of the extensions seem to be busy. Starting simple switch on 'Zap/5-1' -- Executing Macro("Zap/5-1", "exten-vm|200@default|200") in new stack -- Executing SetVar("Zap/5-1", "FROMCONTEXT=exten-vm") in new stack

Greetings list, We are having a puzzle with * (asteriskathome 0.5) and SIP phones (SPA2000 ATA's). If callwaiting is enabled, everything (including call waiting) is normal. If callwaiting is turned off, the phone will not accept incoming calls and the call goes straight to whatever is programmed for the busy voicemail response. It doesn't matter whether reinvite is on or off, or

This is the last remaining function that doesn't use the reservation lock completely to fence off access to a buffer. --- drivers/gpu/drm/ttm/ttm_bo.c | 25 ++++++++++++------------- 1 file changed, 12 insertions(+), 13 deletions(-) diff --git a/drivers/gpu/drm/ttm/ttm_bo.c b/drivers/gpu/drm/ttm/ttm_bo.c index 4ab9f7171c4f..d7d34336f108 100644 --- a/drivers/gpu/drm/ttm/ttm_bo.c +++

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn

Hi list, is it possible to get in the running dialplan the status of (SIP) lines without using AGI or anything like that? What I want is a stepwise calling: I have several SIP lines (let's say they are three) which I want to dial to alternatingly. But I do not want to dial to a already busy line and catch the busy. Instead I do not want to dial to that peer but to the next one. I want to have

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot)

Hello all, I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary changes to the * makefile, so the compilation went well. The first thing I did was configuring two extensions from AMP, namely 200 and 201. Then I installed X-lite on two PC's and configured them with one of the extensions: System settings - SIP proxy - Default: Username: 200 Authorisation user:

Hi all, Hopefully, I just need a second set of eyes on this one, but I just can't figure out what I'm doing wrong. I'm using an agi script to dial a number, check the dial result, and act accordingly. The problem is that I'm not getting anything back from DIALSTATUS, or HANGUPCAUSE. Here is the relevant perl code: ===============================================================

Hi, > It turns out that the bochs and vbox drivers automatically reserved and > unreserved the BO from within their pin and unpin functions. The other > drivers; ast, hibmc and mgag200; performed reservation explicitly. With the > GEM VRAM conversion, automatic BO reservation within pin and unpin functions > accidentally got lost. So for bochs and vbox, ttm_bo_validate() worked on

Hi, On Tue, Dec 01, 2015 at 02:30:05PM -0800, Dmitry Torokhov wrote: > Hi Sinclair, > > On Tue, Dec 1, 2015 at 2:18 PM, Sinclair Yeh <syeh at vmware.com> wrote: > > Port reservation is not required. > > You need to expand on why we do not need to reserve port. Thomas gave me this input earlier, too, so I added the one liner. There was a long discussion on accessing

On Mon, Mar 4, 2019 at 11:29 AM Gianluca Cecchi <gianluca.cecchi at gmail.com> wrote: > > > On Mon, Mar 4, 2019 at 11:28 AM Gianluca Cecchi <gianluca.cecchi at gmail.com> > wrote: > >> An older thread on CentOS pointed to this resource: >> http://dries.eu/rpms/perl-Net-SCP-Expect/perl-Net-SCP-Expect >> >> HIH, >> Gianluca >>

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

I noticed we were getting lots of warnings with xfstest 83 because we have reservations outstanding. This is because we moved the orphan add outside of the truncate, but we don''t actually cleanup our reservation if something fails. This fixes the problem and I no longer see warnings. Thanks, Signed-off-by: Josef Bacik <jbacik@fusionio.com> --- fs/btrfs/inode.c | 2 ++ 1

What is a use case of setting a reservation on the base pool object? Say I have a pool of 3 100GB drives dynamic striped (pool size of 300GB), and I set the reservation to 200GB. I don''t see any commands that let me ever reduce a pool''s size, so how is the 200GB reservation used? Related question: is there a plan in the future to allow me to replace the 3 100GB drives with 2

Hi, I'm trying to achieve the following: Incoming call for user A (97), user A make a blind transfer to user B's phone (96). User B's phone rings and since there is no one to take the call, it returns the call to User A with an AGI script. The dialplan looks like this: [local] .... exten => 96,1,Dial(SIP/user4,10,tr) exten => 96,2,AGI(transfer.php) exten =>

Hey all. We have a DS3 circuit with GBLX split off into 7 systems with a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P cards in the other 5. GBLX numbers their spans from 0 to 3 instead of 1-4 and we have a NFAS configuration with the d-channel on chan 96. All of our systems are running 1.0.7 for stability reasons (and no good time for maintaince, the entire platform

I'm having some trouble getting the channel status with an AGI script. #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->channel_status('Zap/1-1'); I am now stuck, and don't know how to get the return codes: -1 There is no channel that matches the given <channelname> 0 Channel is down and available 1 Channel