similar to: iax busy / unavailable - not registered

If I have an IAX client (Firefly softphone in this example), and the client is not registered at the moment because they are not connected to the network and someone dial that extension, they get the user's "I'm on the phone at the moment" message vs. the "I'm unavailable" message. Is this by design? Here's the extension in question's dialplan:

Hi Asterisk-ians! Need all of your help. I am stuck with this issue for last few days. I have one X100P installed in my system. My Asterisk is registered with another Asterisk Server/SIP provider as client and the call is successfully received by my Asterisk server (named as starwars). Now, the extentions.conf file states, the incoming INTO * should go out to fxo as below: exten =>

I trying to receive a SIP call and have ring a analog phone. I have a TDM11B card with FXS(green) module in line 1. I have Sip server "SER" setup to accept a SIP call, add a 970 extension to uri and set to asterisk SIP server on port 5065. When I send a SIP call from "kphone a soft SIP phone" running to sip://wally.world@cci.net "SER" picks call ok and changes uri

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",

I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press

Hi - We're moving up in the world to a PRI (Verizon), and I'm having some problems with it. I'm new to this PRI thing, though, so maybe I've just screwed up a simple config detail. I've got a TE410P on a Dell PE1600SC (ServerWorks Chipset). The card itself has a green light for the PRI, and Zttool shows Span1 as "OK". All I'm getting when I call into any

Hi, I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is: [from-sip] exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN}) and my zaptel.conf is: fxsks=1 loadzone=us defaultzone=us and my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes

Greetings folks; I guess I must be missing something, because for the life of me I can't seem to make this work. I have remote clients connecting to Asterisk using IAX2, these clients have changing IPs so we're using the useful register tool. The client can call out successfully, that's not an issue at all. Calling coming from the server to the client, however, do not appear to go

asterisk-users Any help will be appreciated. This card did not connect with E1 line how to loop E400P card to test ? now I loop the card. span 1 ---span2 RJ45 pins 1--4 2--5 but show : When calling ,showing error: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' Asterisk Ready. *CLI> -- Registered SIP '2002' at 192.168.139.59 port 3289 expires 120

Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

The setup: Inc SIP Call -> Asterisk 1 -- IAX --> Asterisk 2 -- SIP --> phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto("SIP/XX.XX.XX.XX-0819f590", "cytel-internal|3044|1") in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial("SIP/XX.XX.XX.XX-0819f590",

Hi, I've created a test at "extensions.conf" like this with 3 steps: ; When dial 5555, get the first available channel and dial do 482343400 exten => 5555,1,Dial(Zap/g1/482343400,5,rt) ; When dial 5555, get the channel 20 and dial do 482343400 exten => 5555,2,Dial(Zap/20/482343400) ; Go to Voicemail 1234 exten => 5555,3,Voicemail(u1234) I've tried using just the

Hello everyone, I bought 2 HFC-ISDN Cards and want to run the first card in NT-Mode an the second one in TE-Mode. Everything looks ok under SuSE 9.1, but I can't dial out. I removed one card, for testing purposes and want to run this one card in TE-Mode. I only want to make a call with my Grandstream BT-101 over Asterisk via ISDN. When I try to make a call I get: ---------------------

Hi list I get this error message when I try to call to another city via IAX, in my iax.conf I have the codec iLBC but until now it works well, how can I solve this problem? -- Executing Dial("Zap/2-1", "IAX2/tuxtla/111110@default|60|Ttr") in new stack Feb 15 13:37:05 WARNING[360466]: chan_iax2.c:6266 iax2_request: Unable to create translator path for UNKN to GSM on

I have a "Cologne Chip Designs GmbH ISDN network controller" in my asterisk box, and an ISDN line connected up to it. I downloaded the bri-stuff 0.1.0-RC4a package from junghanns.net and used that to compile asterisk using these instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc%20install When I try to call my asterisk box, my sip phone rings twice and then the

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows:

I have a SIP phone that is registered. i can make calls out from the phone. I can't make calls to the phone. What does the error message mean? How can I fix it? Thanks! 8 headers, 0 lines Destroying call '6b9fb03c4677b9266e1263fb0c7ea304@127.0.0.1' Jul 15 22:10:49 NOTICE[262159]: rtp.c:285 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible == CDR updated

I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: ----------------- fxsks=1-2 loadzone=us defaultzone=us zapata.conf: ------------------ context=inbound-analog signalling=fxs_ks group=1 channel => 1 channel => 2 I created a couple of test extensions: ; test extensions

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]: