similar to: Dial application response code--help required

Hello all, Here is the requirement from my side to use Asterisk.NET API to generate an automated call (outgoing) from asterisk and then link to one of the extensions which plays a sound file for the callee. For this i have worked out in the follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under

Hello users, Actually i wanted to implement Text-To-Speech engine from cepstral voice using swift application i tried the documentation of doing this and i was unsuccessful at doing this work with asterisk can anybody please help me out finding the solution to installation thanks in advacnce srinivas Antarvedi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am struggling with sending voicemail as an attachement in Email. When i have given the email like someone at gmail.com it is delivering to my gamil account perfectly(of course to spam folder). But when i given the email like someone at mycompanymail.com it is not delivering to my company email account.. What should i do ? Actually my company is using a third party email server..

Hello Users, My Setup is like this openser --Registrar asterisk --Callflow using asterisk-b2bua + radius for accounting My Intention was to generate a Acct-Stop Packet when there is a failure of RTP media from one of the UAC's( callee or caller) who is in dialog. so that the Caller will not be charged for Meaning less network problems Because there is no way asterisk knows about

Hello users, i have successfully installed the cepstral voice and in the text only mode its working fine when i swift applicaiton in dtmf mode like exten =>111,1,Swift(hello user| 5000|1) exten =>111,n,NoOp(dtmf is ${SWIFT_DTMF}) exten => 111,n,Hangup() case1: when i am listening to the hello user prompt if i press any key 1,2,3,4,5,6,7,8,9,0,*,# i am getting the ${SWIFT_DTMF } value

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Hello all i have an asterisk setup integrated with mysql via odbc driver myproblem is: when i reading my voicemails, in advanced options the following functions not working with realtime asterisk but working with flat files. 1. Reply to the message(option no:1) 2.Leave a message(option no:5) i have following settings in my general section _ searchcontexts=yes _sendvoicemail=yes [test1] 1001

Hello users, have been facing problems with t38 passthrough using asterisk 1.6.0.3. observed also that in case of SendFAX we are not having major issues, its almost successfull. ReceiveFAX has problems most of the time. we have been using a ringcentral account for testing this receivefax. so ringcentral is trying for 3 times if the sending fax failed for the first time. what i observed is

Hello all, Can anyone have any experience working with service provider like Talkfree . They are giving the user accounts based on the single user accounts and those needs to be directly register to the service provider not to the local system i have taken a connection which when configured to service providers domain direclty ,xlite can make calls without any problem but if i want to use it

Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Thanks in

I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -------------- next

> -----Original Message----- > From: Khawade, Kirtikumar (GE Infra, Transportation) > Sent: Wednesday, February 15, 2006 7:49 PM > To: 'srinivas.prasad@yukthi.com'; 'sales@deeproot.co.in'; 'info@thelinuxconsultancy.co.uk'; 'info@stelias.com' > Cc: Pai, Usha S (GE Infra, Transportation); Khadmale, Subodh (GE Infra, Transportation); Nk, Radha (GE

Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken,

Here is my Extensions.conf file (Default Context). When an individual calling in dials the extension, the response time seems very slow. It doesn't immediately go to the next step, but hangs out for a few seconds (silence)... Suggestions? Thanks in advance... /pj [default] exten => _XX.,1,Wait,2 ; Wait a second, just for fun exten => _XX.,n,Answer

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

Strange. Any explanation as to how that did it? -- Christopher Barry Manager of Information Systems InfiniCon Systems http://www.infiniconsys.com -----Original Message----- From: Srinivas Murty [mailto:srinivas.murty@verizon.net] Sent: Tuesday, March 25, 2003 6:06 PM To: samba@lists.samba.org Subject: [Samba] Got it working.....after a fashion I found that even without touching anything on my

I've the following dialplan. exten => _XX,hint,SIP/${EXTEN} exten => _XX,1,Dial(SIP/${EXTEN},10,j) exten => _XX,2,VoiceMail(${EXTEN}@default,u|j) exten => _XX,3,Hangup() exten => _XX,102,Goto(110) exten => _XX,103,Playback(pbx-invalid) exten => _XX,104,Hangup() exten => _XX,110,VoiceMail(${EXTEN}@default,b|j) exten => _XX,111,Hangup() exten =>

I am trying to setup an extension in asterisk which dials an extension on the NEC DSX. i.e. If an asterisk user dials 402 I want it to connect to the NEC DSX @ 192.168.1.57 and connect to extension 402. ( 404 would be the NEC DSX sip account that I have the credentials for ). [402] deny=0.0.0.0/0.0.0.0 secret=pass1 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic type=friend

Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at

Hi, I was trying to get an answer to this issue: bookRatingData <- read.table(file.choose(),header=TRUE,nrows=1048570) Warning message: In read.table(file.choose(), header = TRUE, nrows = 1048570) : incomplete final line found by readTableHeader on 'C:\Users\srinivas\Downloads\BX-Book-Ratings (2).xlsx' I tried opening the data file (BX-Book-Ratings (2).xlsx) and added a new line