similar to: best format for playback/generation

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've have installed a new Asterisk 1.4.15 system after having previously used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though the newer one is actually a slower processor. On the new system, playback of gsm files is noticeably poorer (voice quality is flakely) on any connected phone (sip or isdn, internal or external).

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

Hi everybody, I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work with new versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should work fine. I have sound breaking many times a day over the trunks. Server is AMD Athlon 2.4 GHz with

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing to be transcoded on the fly and it's not getting transcoded fast enough. I did a file convert to create gsm versions (currently they are referenced in my dial plan with no extension Seem to have same problem. How do I determine which file

Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the

hi all, Here is the situation. We are currently upgrading our infrastructure to use LDAP as our authentication mechanism and I intend to also have Samba look at LDAP. I had this working in a test case using 2.2 as a PDC awhile ago. Should I be using 3.0 or 2.2 for this implementation? I realize that 3.0 is still quite alpha-quailty, but I've also read one or two interview with the

Hi, The subject says it all. Is it possible to put the IP address of the peer in the CDR records? Using AGI maybe? -- Kind regards, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbressers at signet.nl

Hi there, up till now I had this two-box setup in mind: * no.1: public IP * no.2: private IP, registers with no.1, serves a small office with clients behind NAT See we'd get something like this: SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA The codec of the SIP client (on the Internet) I don't have full control over, that depends on the

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've been notified of a CVE to be publish against FLAC early next week. I have a patch for it and have thoroughly tested it as well as had it verified by the Google Security Team member who found it. I've been asked not to publish that patch until Tuesday. Once I publis the patch, I like to release FLAC 1.3.1 immediately and would therefore like to make sure that all the

Well I upgraded from 4.4.6 to 4.5.0 and discovered that one of my user accounts is completely borked. What is very strange is that everything in Samba looks okay. Here is the first problem symptom. The data is from the DC. total 80 drwxr-xr-x. 7 root root 4096 Oct 9 01:15 . drwx------+ 77 SAMDOM\prg-11868bg SAMDOM\domain users 20480 Oct 9 00:55

lvqcl wrote: > 1) > Current MSVC solution (FLAC.sln and numerous .vcproj files) was made with > VC2005 Express and doesn't allow to build 64-bit files/libraries. > > IMHO it's time to add 64-bit support for MSVC builds, but AFAIK only Visual Studio > 2012/2013 Express are free and allow to build 64-bit files. > > VS 2005/2008 use .vcproj files, and VS

Hi all, It sees that the most serious bug in the flac bug tracker: https://sourceforge.net/p/flac/bugs/413/ has been fixed in git. This fix alone is worth a new release so its time to work towards one. Things I need to do for this new release: * Deal with all current patches on the mailing list. * Review all bugs reported against 1.3.0 on the sf.net. * Testing and coordination of testing

Hi All, I'm going to be upgrading our 1.2 Asterisk system. At the moment we use the Enicomms SLN files. Are there major differences in the 1.4 default voicefile packs, or will I be able to re-use Enicomms?? In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? Thanks Adrian

Folks, Is anyone still using the Visual Studio solution files in the win32 directory? If they aren't being maintained, they should probably be removed (to avoid any confusion). Thanks for any feedback, snaroff

If you look at an .sln file, there's a long sequence of digits associated with every project, something like this 8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942. For the lack of a better word I called it "encoding". After I changed the dependencies between projects in the solution, msvs replaced those numeric sequences by new ones literally for every project in the solution file. The diff now

Hello, I have a large collection of audio files contains music in mp3 format, due to need to free space of hard disk, I need to reduce their size. It seems opus is the best format for this purpose, in order to have the quality of original mp3 files, currently I use ffmpeg command to convert them to FLAC and then use opusenc, the official opus encoder, to convert FLAC files to opus. By using one

On 17.01.2011 12:35, Raju wrote: > Dear friends > Recently i have downloaded Theora video codec from http://www.xiph.org/ > , > here its having both encoder and decoder , how can i build only > decoder on VC + windows platform , if i build now i will be getting " > Cannot open include file: 'ogg/ogg.h': No such file or directory " > pls send me solution for

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama