similar to: GrandStream HT-486 ATA as VoIP Gateway

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, 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

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

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)

Hi, When we use BudgeTone where the DTMF is set to "via RTP (RFC2833)" all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to "remote pick-up the call" through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Best

Hi List, Just received a HT-286 Analog Telephone Adapter. This device is allows the user to take a standard analog telephone set and connect it to a VoIP/SIP based gateway. A low cost way of having PSTN devices make use of VoIP/SIP services. For example you could take a credit card processing machine and punch it into a location where you have broad band access. You could take a FAX

Hi All, Is it correct to say that by design, asterisk wont make use of any cards hardware dsp capabilities ? I don't think that any of the hardware cards currently supported have any dsp capabilities, but I wanted to know if for example, in the future a driver was written for a card that did have dsp capabilities, would asterisk be able to make any use of it ? I am only just starting out

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. >

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Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

Hi, I have Debian Linux with kernel 2.6.6. The all packages compiled except ZAPTEL where I have the following error: voipgw:/usr/src/zaptel# make make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.6' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1165: warning:

Does anybody has got the some problem? The grandstream ATA 486 schould support almost all codecs, but it doesn't work. I get the following message when I force the use of different codec WARNING[9529]: chan_sip.c:2765 process_sdp: No compatible codecs! Feb 3 11:17:15 NOTICE[9529]: chan_sip.c:7395 handle_request: Unable to create/find channel What could I do to see some more detailed

While I was away on vacation, buried deeply in another thread (New Asterisk bounty: SIP simultaneous), Olle E. Johansson raised a question which is close to my heart - Asterisk's management model. A management model which simply manages telephone extensions and dial plans is irrelevant to most organisations. We need a model which manages users and their interaction with the PBX. I am

Hello Scott, SL> Does that include FP hardware? I don't believe that any of the SL> PDA Xscales do, I assume that at least some codecs need FP for SL> compression; without floating point hardware, it's going to be SL> really slow. 1st in Xscale is integrated Micro Signal Architecture (MSA), the new design incorporates DSP and microcontroller functions - Intel and Analog

Hello, I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1 line and currently have echocancel=yes and echocancelwhenbridged=yes on those spans in zapata.conf. I was discussing CPU load with another Asterisk user and he mentioned that PRIs don't need echo cancelation and that turning it off will reduced CPU load on the server. I checked many sample configs and the