similar to: G.729 for Asterisk

hello I dont want to disgrace any company but i want to know from your(user)experience which one is good in case of g.729 (performace etc) is it Howler(http://www.howlertech.com/products/howlets) OR its Digium ( http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5 ) plz note i dont want to degrade any company... But to know what experience you

Good afternoon gentlemen (and ladies). A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729

Dear all, I have implemented an Asterisk SIP server for a WAN VPN over Internet. We have users distributed along all my country (Argentina) that register to my Asterisk in order to talk among them. I'll plan to have voice and voicemail with GSM codec, because we can't afford the payment for the G.729 licenses (it's an administrative problem of our company, not an echonomical problem).

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Hello I have two questions ! 1. What is the best speech recognition engine for asterisk? I have searched and asked on forums and found that lumen vox is best for asterisk bala bla bla 2. For TTS (text to speech) which TTS engine will be better to use ? I have tested Flite , cepstral (i have not buyed lisence for it trial only) but still thinking may be i have a good option ? -- Best Regards

Hello list I had the following echo-test extension on my Asterisk 1.2 setup. exten => 1003,1,Wait(1) exten => 1003,n,Playtones(!1050/1000) exten => 1003,n,Wait(1) exten => 1003,n,StopPlaytones exten => 1003,n,Echo exten => 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones

Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hello everybody, I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used as PSTN gateway to asterisk in a small office. Everything works just fine, except that sometimes, and it seems that only for long incoming calls, the IVR menu appears on the middle of the call(like a three way call, call goes on with prompts playing over the parties). Dialing an extension at the prompt at

>Regarding whether notifies are still plain UDP. Yes, the config parser >doesn't accept additional arguments to "notify" and judging by the xfrd >code anything to do with notify is using UDP, so no TLS yet. Is there currently or will there by provision for NOTIFY over TCP? Knot-DNS only supports NOTIFY over TCP and not UDP, so interoperabilty is in danger. nsd.conf(5)

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

Halo, fnmatch() and getopt_long are necessary to compile module-init-tools with klibc out of the box. Would you consider accepting patches to implement those 2 functions? Can we use the bsd libc functions? is the lisence acceptable to you? .bill

Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What can cause that problem? Thanks in

Hello Every one! I am little bit new to asterisk. I am doing research on different telecom options as well. I have question for you professionals In order to get E1 line working with Asterisk. What E1 line parameters need to be specified in Asterisk(configuration files). They vary from country to country. What is difference in one countries E1 and others E1. the most important If we want to

I was pointed to the following: http://asteriskforum.ru/viewtopic.php?t=1761 It is in Russian, which I don't speak, but it references an Asterisk patch. Is this patch in 1.4.17? Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?) Anyone work with this?

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <span class="postbody">Greetings,<br> I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a </span>VPMADT032<span

Hello Everyone This samba server was working perfectly without problems. Running as an Domain member vs Win2K ADS One day it stopped working? All that happened 5 days ago was a change of the administrator/root password We adjusted the wbinfo ?set-auth-user towards the new password. But nothing have worked since. install:/ # wbinfo -V Version 3.0.13-1.1-SUSE What might be wrong when the following

Hi Andreas, The suggestions I captured in GitHub issues. Thanks for sharing. Regarding whether notifies are still plain UDP. Yes, the config parser doesn't accept additional arguments to "notify" and judging by the xfrd code anything to do with notify is using UDP, so no TLS yet. I've added a GitHub issue for this too. Thanks for the suggestions. They make for nice

Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I

On 1.6.1-beta4: Trying to receive faxes over a pstn line. extensions.conf: [incoming-pstn-line] exten => fax,1,NoOp(Fax Detected) exten => fax,2,GoTo(incoming-fax,s,1) exten => fax,n,Hangup() ............ [incoming-fax] exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM}) exten => s,2,ReceiveFAX(${FAXFILE}.tif) exten =>