similar to: Migrating home POTS VM to Asterisk VM

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

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

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

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

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,

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

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

Does anybody out there run * on an AT&T broadband phone line? I'm not seeing any callerid and I can't tell if its AT&T doing something funky or if its my setup. I do see CID on my normal phones.... Thanks, Chris -- The face of a child can say it all, especially the mouth part of the face. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!

Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay.

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

Hey all..in trying to futher troubleshoot my caller id problem I'm looking at some past troubleshooting tips and this struck me as strange: If I cat /dev/zap/1 I *always* see data...no matter if the line is in use or not...is that typical? Just curious... Thanks, Chris -- "People are not homeless if they're sleeping in the streets of their own hometowns." - Dan Quayle

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS