similar to: Hardware Requirements for 1M minutes

We are using G711 for phones to talk to Asterisk and G729 licenses at asterisk to talk to ITSP Could you please suggest transcoder to use from G711 and G729 and which is comptible with Asterisk. We will like to avoid using TDM if possible Also i remember that initially we didn't have G729 and were using only 711 for with vicidial but then also we had same problems. at that time it was only 2

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice

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

Hello Does anyone know of any possible concerns in moving a "production" CentOS 4 install from one machine to another where the "only" difference is that the processor changes from a 1.8 Intel P4 "without" Hyper-Threading to a 3.0 Intel P4 "with" Hyper-Threading ? I am guessing not, yet I "have to" ask. Regards, - rh

Hello, Question #1: I have * with g729 installed, and two phones - Cisco 7960 and Cisco 7905. If g729 is the only available codec for 7905's configuration, then call from 7960 to 7905 goes without any problem and both phones use g729. But if I call from 7905 to 7960 the following is displayed on * console: WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

Hi, my setup is like: phones (g729/g711)--(SER)--> Asterisk <--(oh323)--gateway (supports g729&g711) problem begin when phone supports only g711 and Asterisk doesn't negotiate this codec in full path (from phone to gateway), but tries to do transcoding (and because I haven't g729 codec in asterisk, the call fail). Is there any solution how to tell to Asterisk to negotiate

Hi guys I know this has been asked on the list before, but my hard drive crashed and I lost all of the past posts, I need to know what motherboard works ok for asterisk, I have no problems with the Dual and Quad Xeon processor boards I have used. Now I plan on building a Pentium 4 3.0 with hyper-threading. I looked through the wiki and could not find the recommended P4 board. Does anyone have

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

Hi, Does anyone know if Asterisk is a good tool to be used for a large quantity of g711 and g729 transcoding? What is the best alternative for that? -- Woody Dickson woodydickson at gmail.com <woody.dickson at gmail.com> US and Worldwide Termination ============ Contact me for the following offering ============ USA Onnet - 0.0049/min USA Offnet - 0.011/min USA Mobile starting

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -

We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

Hello, Didn't get much help on asterisk-dev, so here it is. Please help. Thanks. From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Mark Hamilton Sent: March 13, 2008 2:10 PM To: asterisk-dev at lists.digium.com Subject: [asterisk-dev] Hardware and CentOS tweaks. Hello, We're working on using Asterisk as an

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

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

Hello, After reading [1] (in french), I would be very happy if I could get answers to: 1. Does this 13.7+20161113-3 package version has any relation with asterisk's version it complements ? Current asterisk version in repo is 13.14.0. Does this 13.7 complies with it ? 2. From package description, is this package enough or not to allow transcoding with G711 ? For instance, in the following

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github pointer. Michael On 7/5/23 16:46, asterisk at phreaknet.org wrote: > On 7/5/2023 4:19 PM, Michael