similar to: German Termination and DIDs

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

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

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

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

Just tried compiling chan_capi 0.3.4 under CVS Head and get the following errors. chan_capi.c:60: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:61: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:62:

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

<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>

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, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation. Thanks. Angel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best

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

Greetings all, I've been working on having Asterisk put a call through to two different numbers, and give the call to the first one that acknowledges by pressing the 1 key. I found an example on the wiki, but I can't get it working. When I answer the call I hear the message telling me to press 1 to connect, and as soon as the message is done, the call is connected. In other words, it

-----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

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

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,

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

Does anyone know how (if possible) to do three way calling on the UIP-200? There doesn't seem to be much info about this phone, but all the feature lists I've read says it can do conference calls. I can't seem to do it, though. Any help would be appreciated. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us