similar to: G.723.1 and Asterisk

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

Hi to all, we are searching for a hardware based DSP solution for use with Asterisk based on PCI or MiniPCI to reduce main processor load and to use embedded boards with Digium E1/T1 cards like TE410P. does anyone know about any manufactorer of those cards or someone who is able to develop/build such cards? Specifications: PCI or MiniPCI up to 120 concurrent transcodings Codecs: G.729/G.729A or

Hi all, I've a problem using a user with administrator privileges for mounting or adding a machine to PDC. I created a common user called fernando #id fernando uid=1000(fernando) gid=544(Administrators) grupos=544(Administrators),512(Domain Admins) Mounting a resource with smbmount works fine, this the result from log.smbd check_ntlm_password: authentication for user [fernando] ->

i was following a thread fron this mailing list ,a web embed player for ogg stream i am in a = situation , need a player or a way to play stream that no require java -for download plugin , is heavy - i like to know how revsolve he problem or get a solution} sorry , I am Argentino ,i don?t speak english just i read a bit and now i?m writing dani

hi Adam Thank you for your fast answer I affraid that I don?t point? to the right problem I need a solution for all case , user don?t need install anything and dont know that player is default just click and go that is the problem , how to do? i think in a java player , minimal ,lite o something similar for windows users i put in the page a xmplay with a .pls pointing tu server, is a temporary

Hi, a new package ?HBV.IANIGLA? for hydrological modeling has been relesased. CRAN: [ https://cran.r-project.org/web/packages/HBV.IANIGLA/index.html | https://cran.r-project.org/web/packages/HBV.IANIGLA/index.html ] In case of doubts do not hesitate to write. Best, Ezequiel Toum -- Ezequiel Toum Ingeniero Civil Becario Doctoral

Hi, a new package ?HBV.IANIGLA? for hydrological modeling has been relesased. CRAN: [ https://cran.r-project.org/web/packages/HBV.IANIGLA/index.html | https://cran.r-project.org/web/packages/HBV.IANIGLA/index.html ] In case of doubts do not hesitate to write. Best, Ezequiel Toum -- Ezequiel Toum Ingeniero Civil Becario Doctoral

Hola, Ya estamos preparando la siguiente reunión del "Grupo de Usuarios de R de Madrid". Tenemos acordada ya parte de la agenda (os adjunto detalles), si alguien quiere unirse y presentar algo (corto, largo o medio-pensionista) no dude en ponerse en contacto conmigo para incluirle sin problemas. Jueves 20 Junio de 2013 - *Lugar:* Facultad de Ciencias - UNED. C/ Senda del Rey,

Dear list members, I am new to R and to the R-help list. I am trying to perform a mixed-model analysis using the lmer() function. I have a problem with the output anova table when using the anova() function on the lmer output object: I only get the numerator d.f., the sum of squares and the mean squares, but not the denominator d.f., F statistics and P values. Below is a sample output, following

Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _________________________________________________________________ Use MSN Messenger to send

Hello, I tried to do a 'sem' analysis for data of how blueberry consumption by birds is influenced by a pollution gradient, using distance and vegetation structural and composition variables, but I got the following error message: Error in sem.default(ram = ram, S = S, N = N, param.names = pars, var.names = vars, : S must be a square triangular or symmetric matrix This may be very

Hi, I want to use G.723.1 on *, I read it is supported in Pass Through mode, but I don't understand whats the meaning of that. I have a GW 5300 and an ATA 186 and I want to place calls to PSTN. I setup this config: [general] port = 5060 bindaddr = xx.xx.xx.xx context = sip tos=throughput maxexpirey=360 defaultexpirey=120 [gw5300] type=friend insecure=yes host=xx.xx.xx.xx

A new release of the open source G.729 patch has been issued. The new URL is: http://www.readytechnology.co.uk/open/ipp-codecs The memory leak in codec_g729 is now fixed. This was due to a problem in a section of code copied from the Intel example. Thanks to those who assisted in locating this bug. If you are still running the old version of the codec, your Asterisk process will run out

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Hello, I would like to put 1 asterisk box in Country A and 1 asterisk box in Country B. Let's assume : - Asterisk box in country A = GWA - Asterisk box in country B = GWB - Calling party number (located in country A) = CgPNA - Called party number (located in country B) = CdPNB - Second Called party number (located in country B) = sCdPNB - PSTN in country A = PSTNA - PSTN in country B = PSTNB

Buenas a todo, Una breve nota para comentaros que este jueves (4-Abril) mantendremos la siguiente reunión del "Grupo de Usuarios de R de Madrid". La agenda prevista es la siguiente: (Prevista) Jueves 4 de Abril de 2013 - *Lugar:* Facultad de Ciencias - UNED. C/ Senda del Rey, 9.<http://portal.uned.es/portal/page?_pageid=93,688166&_dad=portal&_schema=PORTAL> -

> Message: 1 > Date: Fri, 21 Jan 2005 17:38:27 -0600 > From: "Henry Devito" <hdevito@qwest.net> > Subject: [Asterisk-Users] SPA-2000 > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <000d01c50012$4ea49f30$4300000a@homeacxa7jw2xn> > Content-Type: text/plain; format=flowed;

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI> core show version Asterisk