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Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Hi, all I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc.

Dial(UniCall/g1|300|) Where is the number you want to reach? I'd expect to see Dial(Unicall/g1/1234567890|300) To reach number 1234567890 - Mois=E9s Silva On Jan 30, 2008 1:21 PM, Roger C. Beraldi Martins <rogerberaldi at gmail.com> wrote: > Dears, > > After weeks trying to contact support of my telecom about 'Seize Ack' > because that is not returned, was a

I've a call center using IAX softphones provided by a third party. We've observed problems where the IAX phones seem unable to use our PRI trunks. A sample anonymized call is provided below with the PRI debug calls embedded. Any thoughts, comments or suggestions would be welcome. In anonymizing it, I preseved the format and number of digits sent. -- Accepting AUTHENTICATED

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I live in Sweden and I am having problems getting asterisk to properly detect when a caller hangs up. And yes, I DO have disconnect-supervision on my line. Also asterisk sometimes misinterprets the disconnect-signal as another incoming call. This usually happens if I hang up first and then when the caller hangs up, asterisk treats it as a new

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is

Below is my extensions.conf for the fax part [incoming_28345474] ; ;******************************************************************** ; BEGIN - Inbound call handlers ;******************************************************************** ; exten => 8862100,1,NoOp(${CALLERID(num)}) exten => 8862100,2,Background(if-u-know-ext-dial) exten =>

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

Hello, Seeing a strange thing with dovecot 1.0 b7 pop3. If a user gets a particular _very_ short spam message (not sure what virus makes these..), with NO Subject and NO To: headers and NO body, (example available upon request), the user is unable to download the mail. The log entry is: May 10 10:51:35 mail dovecot: POP3(user): Disconnected top=0/0, retr=1/1344, del=0/75, size=16383381

[0004f2xxxxxx](poly650) defaultuser=0004f2xxxxxx callerid="Front Desk" <1600> mailbox=1600 *setvar=callidnum=1234561600* and from extensions.conf: [outgoing] ; Outbound unrestricted domestic calls exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.) *exten =>

as being live and only reports that the SIP device has become unreachable (in full log). Is this something that is fixed in an update? (Currently running 1.2) It seems when Asterisk detects the SIP device has become unresponsive, it would auto-disconnect any calls bridged to that device...though its not. Thus creating what I like to call 'Phantom Calls'. I can arrive in the morning and

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

> Hmmm... I have this aweful feeling that I'm choosing the > exact wrong time to ask a "newbie question" :) Oh well, here > it goes. > > The quick question is : "How do I dial an extension?" > (answer is probably - "you don't" in which case:) "How do I > dial my asterisk box?" - I have no outside line, I just want >

Good morning gentlemen, is my first post in the list, now I'm starting asterisk wanted to have your help for some questions. Well the first function is as follow me. Here I will demonstrate how this configuration follow me on my extensions.conf but it is not working, and do not know why, but something is missing? You must set up followme.conf ? What I want is that the follow-me is

Came upon a problem today that I thought I'd see if it's by design, if I'm missing an option somewhere, or if my fix is the way to fix it. We setup a remote location with a server, same as we've done with others, but for some reason when they would transfer an outside call anywhere it would pause for a few seconds and hang up the line. Well, after spending most of the day on

I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed