similar to: dialing wrong numbers

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

Hello List! I finally got asterisk with capi working, and its already answering my call as well! :) Now i would like to call a number from my shoft phone (kphone). This is my extentions.conf: --- [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password

Hi all, this is a rather "newbie-oriented" question, so please bear with me... The system running Asterisk has been provided with an AVM FRITZ!Card PnP. SuSE Linux 9.0 recognizes it right after booting the system and it seems to be configured (MSN) correctly... The hwinfo looks like this: --- pbx:/etc/asterisk # hwinfo --isapnp 11: ISA(PnP) 01.0: 10300 ISDN Adapter [Created at

What I've got... Software: Linux: Slackware 9.1 Asterisk: out of CVS - so its new. isdn4k-utils: to test the ISDN Card Hardware: PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM 1 x ISDN BRI Card - DIVA EICON (Installed + working) 2 x Grandstream (Barbie?) BT100 SIP Phones. What Works.. I can call from one phone to the other... get read voicemail... I can

Hi, Apologies in advance for the lengthy email. I'm new to asterisk and have trouble with isdn4linux. The setup is very basic like this: winxp ------- asterisk -------- winxp x-lite | x-lite | pstn The hardware involved is: Compaq EVO with RH9/kernel 2.4.20-30.9. Fritz!Card PCI v2 Asterisk CVS-04/17/04-21:36:18 Basically

Hello all, This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio" problem of the previous release. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael.

We have setup an asterisk box to let everybody call into the university internal network, but I get unexpected hangups when doing an outbound call from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into the call. ----------the dial and the problem----------- -- Executing Dial("SIP/57966-a19d", "Modem/g1:96121||rt|") in new stack --

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

hi, i installed asterisk and works very well, the only problem is that when i try to call a direct number of a company that has a normal PBX i got this error: to 10.8.210.153:5060 == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro) -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack -- Goto (doisdn,00115601992,1) --

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

hi ---------modem.conf :---------- msn=240862922 incomingmsn=240866365,6365 device => /dev/ttyI2 group=1 device => /dev/ttyI1 ; ttyI3, ttyI4 ---------extensions.conf ;------- [sip] exten => _XXXXXXXXXX,1,Dial,Modem/g1:BYEXTENSION (Sjphpone) Call to : 024076xxxx result : --Executing Dial(Sip/roseau-6163","Modem/g1:BYEXTENION") in new stack -- Called g1:024076xxxx --

Hi List, we are using openh323 gatekeeper for voip telefony. We also have a voip over ss7 TELES Switch for voip into POSTN Network. Know we want to use Asterisk for converting SIP to h323. Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do we need openh323 GK for astrerisk, too?. And how can i tell asterisk to sent all none SIP-ip calls to the gatekeeper over h323? thx

Hi, I have been trying to workout how to use the call pickup. So Far, I have the following in zapata.conf [channels] signalling => fxo_ks context => local pickupgroup=1 callgroup=1 channel => 1-3 When I dial *8# all I hear is busy tone. What have I missed? thanks Jay.

I had unexpected hangups from my asterix box using the i4l driver. (SIP <-> SIP calls worked execellent, but SIP<->ISDN didn't.) Then I changed the i4l driver in modem.conf with the chan_capi from jungham. (http://www.junghanns.net/asterisk) I followed his instructions in the INSTALL file, and first encountered some errors compiling it. It help by deinstalling several

Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network