similar to: What's with Indications/SetLanguage/Zaptel/RingBack ?

Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back

I once (for a brief period) had dialtone, but I do know why :) Otherwise I get a boooop-booop-booop sequence. I cannot tell if this is the D-Link doing this, or asterisk... Who should be giving solid US dialtone? My indication.conf says: [general] country=us ... [us] description = United States / North America ringcadance = 2000,4000

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

Greetings, fellow Asterisk users! I'm having problems with Asterisk and Digium's X100P (using kewlstart) in Finland - Asterisk doesn't detect POTS hangup. I've modified indications.conf according to the Finnish specifications (well, I think I got the specs correctly :-), but still no go.. Has anybody used * in Finland? Does your * detect POTS hangup (which plays the

Many thanks in advance for anyone that can offer help on the following questions: Asterisk Box Using Asterisk@Home build and updated Asterisk to v2.1 P4, 400 Mhz, 384Mb RAM, 40Gb HD 4 OEM X100P Cards Phones Grandstream GXP-2000 2 * Grandstream BT-100 HandyTone 486 Sipura SPA-3000 Questions 1) When someone calls in to one of the FXO lines, there is a 3-4 second delay before the configured

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

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

Aqu? tienes, [es] description = Spain ringcadence = 1500,3000 dial = 425 busy = 425/200,0/200 ring = 425/1500,0/3000 congestion = 425/200,0/200,425/200,0/200,425/200,0/600 callwaiting = 425/175,0/175,425/175,0/3500 dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 record = 1400/500,0/15000 info = 950/330,0/1000 dialout = 500 Sergio Serrano Revuelto Avanzada 7 Original Message:

Hi there, I have a new asterisk working in New Zeland and everything is working well except when an incoming call to the PSTN hangs up, asterisk wont hang up the zap trunk (X100P). I have found this information: http://bugs.digium.com/bug_view_page.php?bug_id=0001474 Which discusses my problem and i have made sure that i have the latest info in the indications.conf as follows: [general]

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

Hello everybody, I have installed Asterisk from CVS (18/07/2003) and although everything works fine, SetLanguage application doesn't seem to work. As it used to work with previous version I wonder if I am missing something here. The relevant line in extensions.conf is: exten => 3,1,SetLanguage(gr) In the directory where Asterisk sounf files reside, I have installed files of the type

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi all, I'm having some trouble. I've got an IVR with 4 different languages using the SETLANGUAGE command and I'm getting a 6 second delay when I make the first selection, after that all is fine. There's nothing in my dial plan that I can find that would be causing it. And the delay is driving me nuts! I have an IAX connection from a provider coming in. Could this be the cause?

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

Hi, Not sure where to submit this to so I'll try here. Below is the toneset for Iceland. Hopefully this can be added into the asterisk package. [is] description = Iceland ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/5000 congestion = 425+250/250,0/250 callwaiting = 600/100,0/100,600/100,0/9000 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0

Hey All, Having a bit of a problem with a Wildcard X100P card. When I try to make an outbound call using the card, it picks the line up and then only dials a single digit. I've confirmed it's only dialling a single digit by listening on a phone plugged into a parallel socket. Incoming calls work fine, it's only outbound calls I'm having a problem with. I've tried

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>

Hi, can't get X100P (fully zapata compatible clone) to work (I'm in Australia). * recognises the card and the channel (1) but has definetely some problems talking to the pots line. I set up this simple dialplan for ZAP ("incoming" context, as setup in zapata.conf, for channel 1) [incoming] exten => s,1,Answer exten => s,2,Playback(somefile) exten =>