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Hi all, I did some call using an asterisk 1.4 PBX and 2 softphone in a private network; call is up, but with bad quality. Someone knows how to debug this problem ? Thanks in advance for any help. -- Giancarlo Lombardo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091111/83d6d4ff/attachment.htm

I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091027/bb5e7b1a/attachment.htm

Dear all, I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have trouble, I see on XLITE console: Registration Error: 503 - Service unavailable. Someone have a tutorial or a step by step description how to do that ? Thanks in advance -- Giancarlo Lombardo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I just configured some user in sip.conf and extensions.conf; they works fine. Now I'm trying to do the same with Extensions feature of FreePBXAdministration, but I cannot see what I have done manualy. Is gui working with other an source, How can I access such data ? Thanks in advance for any suggestion. -- Giancarlo Lombardo -------------- next part -------------- An HTML attachment

Hello, I have been facing an issue that voice is getting distorted sometimes in MOH (MusicOnHold) application. I have checked and confirmed that lame is properly installed, even tried native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH can't be eliminated. I came to know about requirement of timing device for MOH and MeetMe and a very good illustration by Andrew

Dear all, I just installed asterixnow, but no graphical interface start automaticaly neither linux nor some other, just command line. Shall I do something or shall I install something more ? Many Thanks in advance for any help. Giancarlo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I explain what I want to do.. All the operators share their phones. The number of the operator isn't constant, so it's possible that two operators share all the phones. They need to move all around, so they pick up the first phone they find. If there are only few operator is very annoying for them to ear the other phones ringing while they are at the phone! So I'dd like to limit

Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: *Registration error: 404 Not found* Here my configuration file of asterisk: *[root at dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* *[root at

Hi People, I work in a company that are using asterisk as pbx. I need a way to identify what client my employees are calling. For example: - For each call that an employee of my company make to a customer, must identify the client name in the CDR table. - Is there a way of my employee enter a code to identify the client and then enter a phone number to make the call? I would like to identify

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten

Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign

Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers -

I'm doing some quick research on how to get our videos from AstriCon available in a "reasonable" format that allows easy viewing, reduces our bandwidth costs, and allows good tracking for who/where/what is viewing the videos. YouTube seems to have a very nice set of tools and statistics collection methods, and might be perfect EXCEPT.... Their main limitation right now

Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register => 908366554:396444 at carrier.jazzey.com register => 908366554:396444 at sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444

// Please reply this email to jai-Oc46A42Di8jtC6uA/Zb986xOck334EZe@public.gmane.org // Hi, My name is Jai, I am a Technical Recruiter with International Business Solutions. Ltd, NJ. My client in NJ is looking for a JD Edwards Technical Consultant. I have included a description of the project below. Please take a few moments to review, and let me know if this is something you might be interested

Hi all, I was wondering if anyone could help me, I have searched for some reason but cannot find any. After migrating from a previous samba-3.0.10 (I think) PDC server to a newer samba-3.0.23c server on Centos5, I have the strange problem that my XP clients no longer map all the drives listed in the logon.bat during logon, however if I run logon.bat manually after I'm logged in, all

Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140110/18f3f1e2/attachment.html>

is there any link for step by step guide in file upload in database ... im follwing the link http://www.tutorialspoint.com/ruby-on-rails/rails-file-uploading.htm.. which shows file upload into folder not in database ..pls help.. -- Posted via http://www.ruby-forum.com/. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups

why does this not work? i simply want to hear the recorded message exten => s,1,Answer() ;exten => s,n,Record(zipcodegutter1.gsm) ;zcg1 exten => s,n,Playback(zipcodegutter1) exten => s,n,Dial(SIP/c000001s/12222222259,120,A,(demo-thanks))

Dear R-users, I couldn't find a function or some help in R-project web about the Generalized Singular Value Decomposition. In MatLab there is a simple function for this algebric issue (gsvd). Is there anything like that in R? And, if not, could you help me to apply this method in R? Thanks in advance, Giancarlo +++++ This mail has been sent through the MPI for Demographic Rese...{{dropped}}