similar to: Cisco SIP Phone 7960 & DTMF Problem

Hi, Are there any optimized codecs for Analog Blackfin DSP? If yes, from where we can download it? We are looking for Speech, Audio and Video codecs. Best Regards, Miroslav Nachev

Hi, I would like to ask you for advice how to solve the following case: I have a client (who happened to be my friend) and I have convinced him that the IP PBX solution is much better than the conventional telephone centrals (PBX). At the beginning he wanted to buy PBX Panasonic, but at this moment he is waiting for my decision. Because at the moment we are not so deeply familiar with these

Hi, I would like to find some way for hardware coding instead software (using the Host CPU). Are there any PCI boards just with codecs (DSP) or other way? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel: (+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: m_natchev@yahoo.com miro@space-comm.com http://www.space-comm.com

Hi, I can't find anywhere on the Asterisk web the license terms for commercial use of Asterisk software. Do I have to pay something (and how much) if I want to use the Asterisk in our IP PBX solutions? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel: (+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: m_natchev@yahoo.com

Hi, We have a case where we need of 16 x FXS, 12 x FXO and 1 x E1. To do this using Digium products I need of 8 PCI slots. This is not possible to be done in one computer and that's why I try to start using TDMoE. Unfortunately all my tries are without success. The network is crashed everytime. Can you give me some ideas/suggestions how to solve this case? Best Regards,

Hi, Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? -- Best Regards, Miroslav Nachev

Hi, Thank you. I will try it. Do you know some G.72x, GSM, and iLBC optimized for Blackfin ? I mean open source. -- Best regards, Miroslav mailto:miro@space-comm.com Wednesday, March 23, 2005, 9:05:11 PM, you wrote: JMV> Hi, JMV> As far as I understand, the last patch (for TI C5x) I merged in SVN also JMV> allows Blackfin to work, but I haven't

Hi, We have potential client which would like to offer to him VoIP solution for 2000 subscribers (SIP based Phones) and 2 x PRI ISDN interfaces to the PSTN. In the next stage the subscribers will be increased up to 13,000. Because I am not haven't done similar big project I don't know how to calculate the price. The one way is using number of subscribers and the other is using PSTN

Dear Jean, The source code for G.729 can be download from ITU for free. Also, some developer can do yourself as open source G.729 codec without any help. In this case each who use this codec which source code is free and open source must pay, but not to the developer. Best Regards, Miroslav Nachev JMV> Le jeudi 24 mars 2005 ? 10:08 +0000, John Villar a ?crit : >>

Hi, I would like to make a call and then when I am connected to the destination to transfer the call to my coleague in the office. When we receive the call it is easy using "#". But when I am the originator the "#" doesn't work. Can you give me some suggestions? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel: (+359-2)

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that

http://sip.house.com.ar/operator Its a server/client combo that displays the status of your Asterisk PBX in a web browser in real time. You can also perform some actions. Hang-up channels and Transfers via drag and drop. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. It's a work in progress... so

Just upgraded to cvs Head this morning and noticed our voicemail notification (via email) is failing with: Jul 1 07:48:38 WARNING[1217669936]: app_voicemail.c:837 sendmail: E-mail addres s missing for mailbox [3000]. E-mail will not be sent. However, a valid address in voicemail.conf has been working just fine until now. Sendmail is running, etc. If I add a "second" email address

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.

Hi, As far as I understand, the last patch (for TI C5x) I merged in SVN also allows Blackfin to work, but I haven't tested. Jean-Marc Le mercredi 23 mars 2005 ? 10:01 +0200, Miroslav Nachev a ?crit : > Hi, > > Are there any optimized codecs for Analog Blackfin DSP? If yes, > from where we can download it? > We are looking for Speech, Audio and Video codecs. >

Hey Nicolas, That did it. I ran that export command you suggested, then launched *, everything worked fine. I'm still looking for info on what that command actually does. Can you shed some light please? Thanks. JR -----Original Message----- From: JR Richardson [mailto:jr.richardson@cox.net] Sent: Tuesday, December 30, 2003 6:44 PM To: 'asterisk-users@lists.digium.com' Subject:

Hello everybody, I''ve been using RSpec as a tool to create web applications for some time now, in Rails, and using plain Ruby with WEBrick as well. The tool suits my needs and the story runner is great. Now there are things that aren''t solvable on the web, you''ll need a _real_ desktop application for those problems. So I''ve toyed a bit around with various GUI

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hi All, Today, debugging information is encoded in LLVM IR using various llvm.dbg intrinsics, such as llvm.dbg.stoppoint. For exmaple, !1 = metadata !{i32 458769, i32 0, i32 12, metadata !"foo.c", metadata !"/tmp", metadata !"clang 1.0", i1 true, i1 false, metadata !"", i32 0} ... call void @llvm.dbg.stoppoint(i32 5, i32 5, metadata !1) store i32

I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip-> PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of