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Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a

Does anyone have a document showing what all functionality libpri supports? I am trying to get the configuration set with Adelphia/now Telcove before they come out and drop a line that won't do what I want. Quick run down of what I want to accomplish. We are getting a PRI installed to our colocation provider. We will have 21 phone number assigned to the group. We wish to use 1 number for

I find the example to predictive dialing or progresive dialing. I wish to do webpage used by extensions or operators.. where them says click ...and the asterisk dial to a customer and the extension same time. (If the call is answered). Thanks.

Hi All If I want to get my ADSI Phones (successfully connected off a Rhino Channel Bank and TE410P) to connect to Asterisk to get their config downloaded, is there something specific needed in extensions.conf for them to dial to get this? Thanks :)

I am curious. I understand that features can be pushed to an ADSI phone that make navigating your own voicemail easier, and for other internal things. But, does anyone push this data outside of their own phone network? Example: I am at home with my spiffy new ADSI compatible handset and dial up my bank, OneWorldBankingConglomerate. Would they be able to push me their menu? Button 1 for

Martin Pycko <martinp@digium.com> writes: > sure, use the 'n' option of the queue and put voicemail app as the next > priority Will that work? From my read of the code, the timeout parameter is only checked while the call is being sent to an agent's phone (inside the try_calling function). The timeout doesn't seem to be checked while the user is waiting to get to

What are the meaning of these Zap show channel output? Caller ID string: Owner: <None> Real: <None> Callwait: <None> Threeway: <None> Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0,

Can asteric be used with just a voice card. If so, how would I get this going? Also, what carrier would I use connect to? Ie would would be my carrier. Thanks Chip

Hi, Looking around I can't seem to find a way to show the number of agents currently logged into a queue and if possible who they are. Is there a way to do this ? Thanks -b -- ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the

I am trying to make outbound calls from my Asterisk client through a remote Asterisk server with IAX. In iax.conf on both sides [dar] context=trusted secret=xxxxxx type=friend host=192.168.1.1 in extensions.conf at the client making the call Exten=_1NXXNXXXXXX,1,Dial(IAX2/dar:xxxxxx@192.168.1.1/) What goes in extensions.conf at the remote server? What is needed for the

I'm working on migrating my home POTS phone system into an Asterisk PBX. Currently I have my FXO and FXS setup and working great. Zapateller is nice! I'm working on my voicemail now and currently have my 3 mailbox answering machine plugged into my FXS along with all my other analog phones. I'd like to slowly migrate away from the FXS answering machine in favor of Asterisk Voice Mail.

Hi all, I'm trying to install * on Mandrake 9.2/P4, but under asterisk - make clean;make install there is the following error: ---------------------------- [root@net asterisk]# make clean for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x clean || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' rm -f *.so *.o .depend make[1]: Leaving

Hi there, I'm having some trouble with getting Asterisk to make a call, I think it should be quite easy, but anyway... Using the following file contents: ## Channel: Zap/3/<TEL NUMBER HERE> MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: phones Extension: 502 Priority: 1 ## Extension 502 is simply one that plays a sound back. When I dump this file into

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Hello, I am very new to Asterisk, so please forgive me for my basic question. I couldn't find a good answer to this on the info boards. I'd like to know if the following is possible and reasonable to accomplish: I'd like to configure a voice recording system using Asterisk and a Tormenta2 Quad T1 card. A co-worker was able to create this system a while back with Bayonne and a

Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to

We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for

Sorry that got accidentally sent incompleted, here's the full post: OK, here is the long drawn out description of how I am using Zap Barge and Monitor: Zapbarge(listen in on live calls): Very simple actually I just added this to my dial plan(extensions.conf): ; barge monitoring extension exten => 8159,1,ZapBarge exten => 8159,2,Hangup Then when you dial 8159 on