search for: busypatterns

Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn?t work and i was told that 1.6 can?t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best

Hey everybody, I've had issues with detecting call disconnects on the Centrex system and an Adit 600. I've discovered that if I set the channel bank's FXO to lscpd, even though the CO says they don't support, that Asterisk can see the hangups. The catch? Asterisk can no longer seize the line. I've used this to my advantage, since I have 13 lines to work with. (5) +1

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Thanks for your suggestion Steve. I have done as you advised and set busypattern=300,200 to match the sample I recorded. This hasn't worked though, asterisk doesn't seem to detect the busy signal. Does asterisk require a the signal to be in a certain power range? The signal I get is very quiet. Thanks for your help Regards Jonathan On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

OK, got a question on 911. Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't present a problem. However, the club wishes to ensure that 911 authorities are able to identify

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me

Hi Everyone... I am running Asterisk 1.2.22 on Debian "Etch". I installed it from sources. I have also installed tiff-v3.6.0, spandsp-0.0.3. and downloaded http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_rxfax.c http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_txfax.c and a Digium TDM card with 4 FXO ports When my dialplan

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Morning list, Was curious if it is possible to decrease the time asterisk takes to answer an incoming call to a zaptel interface. Example: [Jun 11 09:33:06] VERBOSE[4489] logger.c: -- Starting simple switch on 'Zap/2-1' [Jun 11 09:33:10] NOTICE[4489] chan_zap.c: Got event 18 (Ring Begin)... [Jun 11 09:33:12] NOTICE[4489] chan_zap.c: Got event 2 (Ring/Answered)... [Jun 11 09:33:12]

Hello, I have a problem with the Bristuffed version of Asterisk. I have tried Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have the same problem it seems: The setup: A machine with a single hfc-s PCI BRI adapter running Gentoo 2.6.15. Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1) installed and working perfectly. Grandstream gxp-2000 as a SIP phone, and a normal mobile

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

Hi everyone, Currectly I'm having some troubles to get correct status of my calls throug ISDN lines, when outbound calls don't get its destination I always receive NO ANSWER as ${DIALSTATUS} despite the fact I know the target number doesn't exists or is busy at that time. Maybe there is something I must change in my zaptel.conf or zapata.conf, current configs follows: ####

Dear,is any way to find silent channels , and disconnect them after 30 secs? best -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090824/7709c910/attachment.htm

Hello people, this is my first mail to this list. I'm new to asterisk and trying to set up an IVR. So far my dialplan works nice connecting with softphones, but I'm having problems to detect hangups on the analog line. Here are the details: -Clone pc with Ubuntu server 10.04 64 bits -Asterisk 1.6 -Openvox A400P, lspci -v says: 01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX

"from CHANGES in the trunk: ---------------------------------------------------------------------------= --- --- Functionality changes from Asterisk 1.8 to Asterisk 10 ----------------= --- ---------------------------------------------------------------------------= --- ... chan_dahdi -------------------------- * Busy tone patterns featuring 2 silence and 2 tone lengths can now be use= d