similar to: Chan_Capi 0.3.4

Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between

for today we only have experience with BRI applications together with asterisk. is the following scenario possible and stable enough for production? FYI : We want to build a unified messaging application integrated with SIP. We have an E1 connection in Belgium with 100 msn's We would think about having 2 servers : Server A : Asterisk PRI card (Digium TE410P) Server B : Fax

I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming,

hello! I'd like to use my asterisk-box on a p2p-line from my alcatel pbx. isdn4linux and capi works fine on a s0-line from the pbx, but i need more then 8 msns, cause i'd like to implement a voice-mail system and an extension to our "old" pbx with voip-phones. had anyone success using isdn4linux on a t0 (p2p) line (of course with an inexpensive passive card like my avm

Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric jacksch@tenebris.ca

<P>Hi List,<BR>is it possible to connect an existing telefon system with a stanard AVM ISDN card to the telco?</P> <P>&nbsp;</P> <P>[ISDN from Telecom supplier] ---- [Aterisk Box] ----- [ existing telephon installation]</P> <P>&nbsp;</P> <P>kind regards,<BR>Patrick</P>

All, When calling an invalid number using, I expect to hear: "dooh-deeh-daah We're sorry you have reached a number which has been disconnected ..." And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I continually hear ringing, and then the call gets hungup. Any ideas

Hi, I've got an important question: I use an E100P directly connected to PSTN, but it does not *really* work as it should be: exten => 1000,1,Dial(Zap/1/1234) BUT: It does NOT dial "1234" but it says in debug mode: -- Called 1/72976451 Apr 26 00:53:00 WARNING[10251]: chan_zap.c:5979 zt_pri_error: PRI: !! Facility message shorter than 14 bytes -- Channel 1, span 1 got

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making

hi all talking to this friend if mine just now, I realised I had no 'micky mouse' sound as eaerlier with chan_capi and AVM C2. We talked in 1 hour and 18 minutes without encountering any problems whatsoever. This was him calling me, but I don't know if that's important. roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356

I have been using chan_capi and a Fritz AVM (passive) card for a little while now.. So far so good.. There are still a couple of things I am not totally sure about.. First.. What is the difference between "softdtmf=0" and "softdtmf=1"?? Which is the preffered/recommended option?? Second.. What is Early B3 all about?? Should I be using it in my dial string?? Which is the

Hi, I'm using a ZaptelBRI card. It works fine. But I have a small problem with call logs. The leading zeroes of the external calling party are not stored (e.g. : 0140302010 will be stored as 140302010). Same for international numbers for which "00" will be stripped out. I would not mind if the cdr record would give me an indication of the call's origin (national or

After apt-get update && apt-get upgrade -y wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz tar xfvz chan_capi.02.5c.tar.gz cd chan_capi-0.2.5c make && make install shutdown -r now asterisk seg faults upon calling in via ISDN. Any ideas are greatly appreciated. rgds pos

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello together, I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine

Hi, I tested the capi_chan with latest cvs of * and I have problems with Early-B3. The following dialstring works for me (without Early B3): exten => _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30) But if I add the 'b' for using Early-B3 exten => _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30) nothing changes (no dialtone). If in this example the called party discards the call, there is no

hi all is it hard/possible to move the following from chan_capi_pvt.h into a setting (preferably global) in capi.conf? #define AST_CAPI_NATIONAL_PREF "0" #define AST_CAPI_INTERNAT_PREF "00" and ... Is it hard to move or copy the txgain and rxgain to [global], either as a given 'default' if nothing's set in the interfaces, or as a overall

Advance apologies for the length of this mail; I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works) and all my SIP extensions (which all work) along with

Hello, I've been using * for quite a while using AVM C4 card and a few Grandstream Budgetone 102 hard phones plus couple of HandyTone 286. Echo is a big problem. I am getting used to it, but some users complain. Anyone has experience w/ regards to echo comparing the AVM C4 with two HFC-cards? Before shelling out time and money, maybe someone else already has done so, and could tell me if

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

Hello all, I'm trying to get CAPI to work with two B channels (AVM B1 PCMCIA) on a P4 2GHz (linux kernel 2.4.21) system. All are ok with just one B channel. But when I open a second B chan, the sound is choppy, with too long gaps, and the CPU load is too high (~50%). On the Asterisk's console I get these messages: -- Executing Dial("H323:4478",