similar to: Zaphfc CallerID problem...

I can't see for the trees :) I can make calls out to my SIP provider but get an "unable to authenticate <calling no.> when I try to call in via the PSTN number they have supplied me (where <calling no.> = phone number trying to make the call) sip.conf [general] register => 4316568:xxxxxxxx@sipgate.de [sipgate] secret=xxxxxx username=4316568 fromuser=4316568

A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Iain bell] include => mailboxes include

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you,

Hello, I am trying to use a HFC-PCI (CCD/Billion/Asuscom 2BD0) card in TE mode to dial-in and out with ISDN. The problem is I can not get the card to dial out with a Zap channel. Dial-in is working. I am using bri-stuff 0.1.0-RC4 (but tried also RC3 and RC2k). I tried all combination of "immediate", "overlapdial", "pridialplan". I earlier also managed to dial out

Dear asterisk users! I want to control a hardware pbx with asterisk. What I need to do this is being able to press "hold" which can be done with capicommand(hold) and then send digits on a bri card which connects to my asterisk computer. So far I use Dial(CAPI/ISDN1/27:<<digits>>/bo,15) to do this. Are there better ways? Note that these are not dtmf, I'm afraid. I use

I finally got my ASUSCOM (Cologne chip) ISDNLink card working here in the US. When a call arrives, I get "Unknown IE 42 (Unknown Information Element)" and "Unknown IE 21 (Unknown Information Element)". IE 21 (0x15) is defined as Q931_CALL_STATE_SUSPEND_REQUEST. IE 42 (0x2a) is not defined in the code and I couldn't google it. Is this something perculiar with ISDN in

hi all, i have a probably very stupid question/problem. for testing purpose i am trying to get asterisk running with two isdn cards. I'd only like to here the demo sound when i call the number - but nothing works. The output of show channels is not showing any channel - should there be 4 channels ? - capi info shows my two cards perfectly. The ISDN Controller's are attached to an PTMP

I have a problem receving calls via the ISDN line, using the followin components Asterisk 1.0.9 with asterisk@home chan_capi-cm-0.6 AVM Fritz card datalink protocol = point to multimode I can make calls out with no problems so the issue is only incoming calls. When I make the call from an external line to the ISDN line connected to asterisk, I get a busy signal after about 5 seconds. I have

Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over ISDN/Capi but I don't succeed. My `capi.conf' is like show in many tutorial on the web. In `extensions.conf' I just added the following lines: [capi-in] exten =>

Hi! I am playing around with Asterisk and have a problem :-) (Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4) I have a sip-phone at my desk and an ISDN-phone (independent of the Asterisk-server) in my living room, when I'm not at my desk, the sip-phone is switched off. I would like to be able to accept calls at both phones (when available) and have Voicemail kick in if I don't

Hi, I like to forward an incoming call on an ISDN line to my mobile phone. Since ISDN offers two channels, I thought that this should work, but Asterisk tells me, that there is no channel available. There is no one else using this line, so guess I made a mistake in the configuration or it might not work for another reason. Here's the CLI output , the capi.conf and extensions.conf. 83086921 is

Anybody knows if it's possible to disable zap echo cancellor from dialplan only for certain outbound calls ?? I share the same phone lines for voice calls and faxes. Iaxmodem works fine for me only turning off the echo cancellor, but I need it for voice calls. Any ideas ? maxx

I am using rails edge. I am using gem "rspec-rails", "= 2.0.0.beta. 19" . I have following code at spec/models/user_spec.rb require ''spec_helper'' describe User do it { should validate_presence_of(:email) } it { should validate_presence_of(:name) } end Here is my gemfile group :development, :test do gem ''factory_girl_rails'',

Hi, I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens PABX. From a SIP phone, I can call other internal SIP phones, external numbers (to PSTN), but I can't call internal phones connected to the internal phone network. When I call 107, which is an internal phone, heres the logs from asterisk: -- Executing Dial("SIP/Greg-081f5a10",

[Something seems to have went wrong with my previous posting. It appears on the archive page in another thread. I did not receive anything myself. So I may give it another try:] Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over

I seem to be having a problem with B3 on my ISDN line, as you can see from the dial string I am having to have asterisk generate ringing else there is no progress indication. -- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/g1/142392203000/ b|40|r") in new stack -- Called g1/142392203000/b -- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",

Hello, I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32). I use a context [capi-in] for icoming ISDN calls: ====== [capi-in] ; Faxe fuer Ruben exten => 12345,1,Macro(faxin,ruben.roegels at jumping-frog.org,${EXTEN}) ====== My macro for the fax receiving looks like that: ====== [macro-faxin] ; Faxe ; ARG1 =

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute