similar to: Ignorepat with capi

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

Hi, I successfully installed 2 avm card in my asterisk box but I'm unable to make call. My capi.conf is: msn=0721111,07211115 incomingmsn=* controller=1,2 softdtmf=1 context=default echocancel=yes callgroup=1 devices=2,2 my capi info : Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. my extensions.conf : exten =>

Hi, I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I would like to use the atxfer function but is not included in the stable asterisk. Is there a way to include it in my version of asterisk: I did no used the last cvs because I can't compile the chan_capi .in it. :( Bye

Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate("unknown", "unknown") does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa'

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

Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone

Hi, I finally got Asterisk installed using the German installation CD from http://www.asterisk.de.ms. I got two SIP phones working (SIPPS) asterisk*CLI> sip show inuse Username incoming Limit outgoing Limit 5678 0 N/A 0 N/A 1234 0 N/A 0 N/A

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

One of my buddies signed up with GV yesterday with a view to using them on his * server. Problem is that the settings they gave him don't work with asterisk. They do however work with X-Lite. Any ideas? He's using the settings outlined on my web page. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

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, after having read so much about Asterisk, I went on and tried out to create a little sample-setup. I'm using a Fritz Card USB with the AVM Capi Driver and two X-Lite Softphones. Dialing between the softphones makes no problem. Calling the MSN fron an external phone also works. I'm getting to the asterisk demo-voicebox which works flawlessly. Now may next step has been to enable

Hi, I'm trying to make work chan_capi with last asterisk CVS. I installed last zaptel,libpri,last cvs ana patched chan_capi 0.35 with the patch kindly suggested me by Jason Williams: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 First I received error 127 that I resolved commenting the line CC=gcc-2.95 but now I have this error: chan_capi.c: In function `load_module':

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

> I have placed "ignorepat => 9" in just about every context I > can think of in my extensions.conf, but yet when I dial 9 > from my sip devices, the dialtone is broken. I even tried a > nearly untouched version of samples, and it stil doesn't > work. Is there something somewhere else that needs to be set > to make this work properly, like may in the sip

Hi All, Could somebody please help me to understand the following: - We have 8 msn's 383590, 383591 etc. What I would like to do is for the person at extension 1 dial out on 383 590, the person at extension 2 dial out on 383 591 etc. I have got myself so confused that I need major help!!! If you could give me a simplistic example, including which files I put the coding in (i.e.

Hello list When i'm trying to dial into our pstn the following errors occure: -- Executing Dial("SIP/snomsip-dbd0", "/2100") in new stack Sep 9 10:02:22 WARNING[59409]: channel.c:1901 ast_request: No channel type registered for '' Sep 9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to create channel of type '' == Everyone is busy/congested

Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. Ciao Mauro

Hi, I'm trying to configure a natted budgetone phone to a asterisk server as described in wiki using port forwarding. I successfully make call from the client but it seems it does not register the client ip address and when I try to recall it is not reacheable. Asterisk can manage natted sip client with dynamic ip address ? Bye -------------- next part -------------- An HTML attachment

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

Hi, I can't make outgoing calls with CAPI (passive ISDN Fritz card). See Asterisk error below. Incoming calls and SIP to SIP calls do work. It looks like a msn mismatch in extensions.conf and capi.conf, but I can't find it. Can anyone help me find the problem? Thanks, Rob *CLI> -- Executing Dial("SIP/8112-1be9", "CAPI/356666666:BYEXTENSION") in new stack