similar to: Extra digit needed for outbound call

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

Hi in order to keep the dial tone after pressing 9 for 'outside line' I have this in my extensions.conf [localpstn] ignorepat => 9 exten => _9[123456789]XXXXXXX,1,Dial,${PSTN}/${EXTEN:1} exten => _9[123456789]XXXXXXX,2,Congestion this is properly included in the handsets' context but the dial tone disappears after pressing 9. am I missing something? thanks in advance

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

Hi, I have two asterisks connected via IAX2 trunk. The first * use dial prefix 2XX, the second one 3XX. Calls routing works OK. But I don't know how to get dialtone of remote asterisk pbx. I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone of asterisk #1 after dialing 2. I know something about DISA but I'm not sure if it is a right way. Can you give me advice?

HI, I'm using netmeeting to connect to an asterisk server and dial out. my extension looks like this exten => s,1,Dial,Zap/1/ Unfortunatelly the number that I have dialed in Netmeeting is lost ;-( If I hardcode the number on the line above, like ... exten => s,1,Dial,Zap/1/6642794 ... everything works fine What am I missing?

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is

Hi to all, I'm trying to make outside call in this way : ignorepat => 0 exten => _0.,1,Dial(CAPI/xxxxxxx:b${exten}) But the first number 0 is not ignored. I'm doing something wrong ? Bye

I have the following setup.. [extensions] ; all extensions defined here. exten => 1234,.... exten => 1235,.... [dial-out] ; PSTN dialout config ignorepat = 9 exten => _9,.... exten => h,.... [local] ; phone context in sip.conf is here.. include => extensions include => dialout The question is where will the "h" extension be active?? it appears to run for ALL,

hi, I am new to * and setting up a test system. here my setup : - debian (from knoppix 3.3) - Asterisk 0.7.1 (from the debian package) - AVM Fritz card used with i4l - softphone I use for testing SJphone on windows - I can make great softphone - softphone calls - I can call from an outside line * and get connected to a softphone here my problem: I can not make outbound calls. I place a call

How can I dial *67 on a Sipura 3000 if I dial from a SIP phone connected on an asterisk server. I always get a message saying that authentication failed for INVITE for sip221@192.168.1.6. If I dial a number without doing *67 it's working fine... sip 221 being the extension of my Cisco phone and 192.168.1.6 being the IP of my asterisk server... I have my outgoing context configure

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

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

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

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

I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",

I've just started consulting for a SME client based in Reston Virginia. They don't know it yet but they are going to need a hosted asterisk service and some DID's. Email me if you are able to provide 10 DID's in Reston (must be able to be ported away!!) and hosted Asterisk with end user configurable IVR etc. Probably only 5-8 users at the moment BUT... they'll be

Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl >From: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs >Date: Tue, 04 May 2004 13:32:00 -0500 > >Send

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

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