similar to: X100P no sound problem

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 =>

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 =>

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

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

Hi Vic, I hit that same problem! My SIP phones would sound okay when I made changes to indications.conf but incoming calls in to my TE410P had their own thing going on! Have a look at the zaptel source files, there's one called zonedata.c. You'll see the au settings... replace what's there with this: { 1, "au", "Australia", { 400, 200, 400, 2000 }, { {

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 =

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

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]

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]

I'm new to all this so I don't know where to look, some tips would be most appreciated. I've enabled sip debugging and everything looks fine on the client and server side. Using Linphone on the client side. GSM playback from the server console is fine. I've used Linphone to connect to a vegastream VoIP system so I know if that installed and working. I'm basically just trying

I'm trying to get the X100P card working in AU. So far I have managed to get it to handle incoming calls from the PSTN and have managed to eliminate pretty much most of the echo. My big problem is getting the outbound calls to work. When I get ZAP to dial out it won't connect and I get what I think is the Congestion signal - like a busy signal but with what appears to be a 10db

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

OK, I've spent way more time than I wanted to on getting an x100p clone to work in Australia. I'm happy to consider other (more functional) options. Does anyone have an opinion on both the Sipura 3000 and other Digium cards (like the TDM400P)? I need something that works with no much fuzz. I know the Sipura 3000 is cheaper the the TDM400P card. All I need is to channel my POTS line

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

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:

I'm trying to find a way to generate an 'internal extensions' tonelist but I can't seem to find anything on how to do this. My idea was to start a Playtones(intercom) tonelist and not indicate ringing to the line (dead air). But then, somehow StopPlayTones needs to be run once the ringing telephone picks up. This seems like a dirty way to do this. I envision an option to the

Hi everybody, I had an ISDN card with winbond chipset and isdn4linux, but there was a lot of echo when call from SIP to ISDN, so I buy an HFC chipset card (Conceptronic c128i). I downloaded and compiled bristuff-0.2.0-RC5 and everything is going fine. The sound quality is excellent, there is no echo, but i have a strange problem. Sometimes calls made from SIP phones to ISDN just finished

Dear plotting wizards, when plotting in R, the actual lengths of the axes are slightly greater than the ranges of the x/y variables or xlim/ylim values. how do I control the amount by which the axes are enlarged? Is there a way to enforce that the lengths of the axes equal xlim/ylim? example: plot(0:100,0:100,pch="+") it can be observed, that the x- and y-axis join at approx.

http://hg.dovecot.org/dovecot-dbox-redesign/ Looks like multi-dbox scales pretty nicely. Even after 100k messages the peak saved msgs/sec is the same as the initial saved msgs/sec, even if the average slows down somewhat. I tested this by first deleting mailbox, then running "imaptest" for a second to get saving to start writing several fields to dovecot.index.cache file. Then ran