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.......and which one should i use ..the one connected to usb or the one connected to phone jack on the card ..? Any further help/guidance on IVR is gladly appreciated... thanks digvijay _________________________________________________________________ Get head-hunted by 10,000 recruiters. http://go.msnserver.com/IN/46246.asp Post your CV on naukri.com today.

...n=>h,..." is not suggested for the CDRs. What impact will the use of 'h' make on CDRs? Also, what is the advantage of using ResetCDR with exten=>h? Regards... Girish _________________________________________________________________ Easiest Money Transfer to India. http://go.msnserver.com/IN/41490.asp Send Money To 6000 Indian Towns.

...in case of an incoming call... I have tried rmmod and then reprobing but that does not help....i have the dialtone option in my phone.conf.... please advise thanks digvijay _________________________________________________________________ Looking for something? Cant find it anywhere? http://go.msnserver.com/IN/50756.asp Log onto baazee.com

Hi there, I would like to communicate H323 IP phones with SIP phones. My H323 phones are registered to a gnugk GK, and the SIP phones are registered to a asterisk SIP proxy. I could not create a dialplan that works. Inside my extensions.conf file I created the following two entrances: exten => 4,1,Dial(SIP/4) exten => 5,1,Dial(SIP/5) This allows SIP phones call each other.

...e in figuring out the problem? I admit that there are many bugs in my program and i beleive that the problems are because of these bugs only. Still wanted to hear from you... Warm Regards... Girish _________________________________________________________________ Gujarat Kite Fest at http://go.msnserver.com/IN/40247.asp www.gujaratkitefest.com

...* [subdirs] Error 1 Please help me. I had totaly upset to install the asterisk with cdr. Please help me because i am now helpless. With best regards. Dipak Kumar Paul Tryarc LLC _________________________________________________________________ Claim your Citibank Ready Cash today. http://go.msnserver.com/IN/54177.asp It’s fast, easy and affordable.

Can anyone please help, I can't tell why it will not connect. I do not know how to read this debug file to were it is wrong. Thanks Sip read: REGISTER sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:9082 From: <sip:2203@192.168.1.101>;tag=97442d5b-75b7-4e23-9021-b8605797eb56 To: <sip:2203@192.168.1.101> Call-ID: ea352d6f-a879-4db6-a361-365487a20d4a@192.168.1.100 CSeq: 1

...s'. exten => 12345, 1, SetVar,MYDIGITS=${EXTEN:2,3} ; MYDIGITS = 2345. Also, there is Substring application available with Asterisk, but it is deprecated i think... HTH, Girish _________________________________________________________________ Post Classifieds on MSN classifieds. http://go.msnserver.com/IN/44045.asp Buy and Sell on MSN Classifieds.

Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace

I ma new to Asterisk. I'd like to setup * as voicemail system for SER. Let's say I have an phone number registered in ser as 5554321. When somebody dial to ser for this number and nobody answer, the ser will forward the call to asterisk and get into voicemail box 5554321. I already have asterisk up and running with mysql setup for asterisk voicemail. Can somebody show me how to do it? Or

...erver: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? " Thanks in advanced Dipak Kumar Paul Tryarc LLC _________________________________________________________________ Post Classifieds on MSN classifieds. http://go.msnserver.com/IN/44045.asp Buy and Sell on MSN Classifieds.

...response-header: date => Mon, 12 Nov 2007 14:28:34 GMT D, [2007-11-12T12:22:36.590241 #24540] DEBUG -- : response-header: xxn => W4 D, [2007-11-12T12:22:36.590344 #24540] DEBUG -- : response-header: content-type => text/csv D, [2007-11-12T12:22:36.590430 #24540] DEBUG -- : response-header: msnserver => H: BAY124-W4 V: 12.0.1190.927 D: 2007-09-27T23:27:08 D, [2007-11-12T12:22:36.590509 #24540] DEBUG -- : response-header: content-encoding => gzip D, [2007-11-12T12:22:36.590586 #24540] DEBUG -- : response-header: content-disposition => attachment; filename="WLMContacts.csv" D,...

Hi, I am a beginner to Asterisk. Can anybody clear my following doubts regarding the configuration needed? 1) What is the ideal system configuratin required?(like processer, RAM, h/d space etc) 2) How many connections it can handle at a time? 3) How many Virtual PBXs it can handle? 4) Whether Postgres or Mysql is best suited? 5) How many IVR's it can handle simultaneously? 6) How many

Hi, I have received some replies for my previous mail (* configuration), asking for my goals in configuring Asterisk. So here they are: We are planning to host an Inter continental virtual PBX service that will enable our users to register for an account which give them a toll-free # or a DID. Once registered using a web based interface, that user can add as manay extensions he/she wants

Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future

Hi All, How to park and pickup a call? The scenario of park and pickup described as below. UserA made a call to UserB, and the call ware connected, Then UserB parked (or hold) the call, and told UserC to pickup the call on one line, and then, UserC pressed some keys to pickup the call. Who can tell me what's the Park/Pickup's typical flow in the Asterisk. And how to set the sip.conf,

Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for "ivr menu".)

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

...= yes writeable = yes printable = no [msubbu] comment = samba tester path = /home/msubbu guest ok = yes printable = no writeable = yes Thanks & Regards Murugan Kalicharan S _________________________________________________________________ Post Classifieds on MSN classifieds. http://go.msnserver.com/IN/44045.asp Buy and Sell on MSN Classifieds.