similar to: service provider connection problem

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 all, I am testing the Dial application with the fall through priorities for different cases what i want is the flow after failure of the Dial application which simulates response codes like 1)404 -- Not found 2)480 --Temporarily Unavailable 3)486 --User busy i did manipulate the priority flow like the following for the case 2 and 3 ... exten => _XX,1,Dial(SIP/extension) exten =>

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, 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 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 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, 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

Hi, Please help me through to overcome this error. I have my asterisk server and a xlite client in 172.16. network. I opened up port 5060 in my enterasys wireless router for my asterisk server and also opened port 8000 and 8001 for my xlite client. When i tried calling a remote system which successfully connected to my asterisk server i'm getting the error as " call failed: 499 Not

I've a problem connecting uniVoice (http://voice.uni.it) from asterisk. Using my account data I can place a call smoothly using xlite or my budgetone phone directly, but I'm not able to use uniVoice as a peer from asterisk. Registration seems to work correctly, but when I try do dial, the sip authentication fails every time. Their tech people told me that they are unable to make

Dear all I've got call queuing working when calls originate from my local site. After testing I migrated it to calls originating from our voip provider- it should ring an extension, then queue . All works well apart from if the caller hangs up when queued: the call hangs around in the queue as a phantom until one of the extensions answers it and it is destroyed Am I doing something wrong?

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

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

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

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

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

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the