similar to: voice-coloring with asterisk

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

I''m trying to build a wan latency test environment, where packets from different "remote" locations get delayed by different amounts of time, depending on which remote location we''re pretending they are from. Currently, I''m doing this using the ''prio'' qdisc to obtain multiple bands, and hanging a different netem qdisc off each of the branches

Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call:

Hello! I have developed an IVR using AGI and so far it works great. I'm using Cepstral voices, but now want to use the voices from AT & T that are on a Windows server to be heard best. With cepstral what I do is to generate audio files from shipping and this text I reproduce this method it has worked very well. Now, try to do the same by creating the audio file in windows with the

I have an analog phone connected to a Sipura 2100 which in turn connecteds to * over a 100mbps LAN. When I do ChanSpy on a bridged call, it causes massive jitter. When I attempt ChanSpy with a Grandstream GXP-2000 the monitored call is clear. Has anyone had this happen? Any suggestions? ScriptHead

I am seeking to implement the following behavor: When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement? ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060120/0892441d/attachment.htm

Mmm...I swear I read somehwere that the 7912G did SIP? Cisco lists it as an SCCP only phone? -- Undocumented Features quote of the moment... "It's not the one bullet with your name on it that you have to worry about; it's the twenty thousand-odd rounds labeled `occupant.'" --Murphy's Laws of Combat

I just noticed that quality of .gsm files for using with asterisk is not that good.. is there any way to make then sound better? asterisks sample voices sound way better than theones recorded using applications like wavepad or with asterisk like unavailable messages... any tips? Do you know the command line for sox to adjust the volumen levels or gsm files (make the louder)? Also, do they have

Hello, > 2. I don't have a good source of wav data for testing. I've noticed that > introducing bugs into speex (even gross ones like returning completely > incorrect codebook entries) tends to, sometimes subtly, degrade quality > instead of blowing up. Is there an existing set of source data - and > maybe even a test harness that will do binary comparison, so I can avoid

Hi list, I just downloaded the ulaw prompts from this website: http://www.gmvoices.com/solutions/detail/asterisk/ as had a few complaints about the levels and accent of the existing prompts. The downloaded prompts sound great when listened to on my desktop machine, however when I use them on our CentOs 5.2 / asterisk 1.4.21.2 server there is a fairly loud popping noise in between each sound and

-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Tuesday, June 19, 2007 6:38 PM To: Michael Shatz Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Blackfin inline assembler and VisualDSP++ toolchain >> Yes, data footprint in the new version is quite manageable. Still I would >> wish better documentation for speex_alloc_scratch(). >

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

On Dec 17, 2009, at 11:02 PM, Brian Willoughby wrote: > > On Dec 17, 2009, at 19:00, Ron Decline wrote: >> On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: >>> On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> wrote: >>> >>>> Is it possible to switch the Left / Right channel when encoding in FLAC? >>>> (I have

Hey guys, I'm trying to get music on hold working. I have a wav file. It plays fine on my windows laptop in all sorts of audio applications. If I put it on our asterisk 1.2.4 box and do something like: sox -V nov_2005.wav /var/lib/asterisk/mohmp3/nov_2005.raw sox: Detected file format type: wav sox: Chunk fmt sox: Chunk fact sox: Chunk data sox: Reading Wave file: Microsoft U-law format, 1

> Hi Daniel, > > I'm intrigued by this and wanted to try it out - but I'm wondering how > you get Asterisk to call sox at all during Voicemail()? Our server > doesn't even have sox installed, so I'm not sure how to go about > tricking Asterisk into running a different one. To do anything useful you would have to get sox installed on your server. But to get

On Dec 17, 2009, at 19:00, Ron Decline wrote: > On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: >> On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> >> wrote: >> >>> Is it possible to switch the Left / Right channel when encoding >>> in FLAC? >>> (I have some flac files with incorrect left/right channel

I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't seem to get it to work correct. Could someone point me to what I need to do? I have attached what I believe are the relevant parts. [globals] ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 ; uncomment this line if you are using Ogg Vorbis

I'm on Centos 5.5, and would like to use sox to strip out any periods of silence > 5 seconds from a batch mp3 audio files. Googling I found sox, but it does not seem to support mp3 files by default. The man page says: .mp3 MP3 Compressed Audio MP3 audio files come from the MPEG standards for audio and video compression. They are a lossy compression format that achieves

For those of you who want to be able to decode Ogg Vorbis files into other formats, SoX (the swiss army knife of audio programs) now has Ogg Vorbis support in CVS. It can read Ogg Vorbis files and export the audio in any format SoX supports. You can also write Ogg Vorbis with it, but, because of limitations in the design of SoX, only output at 128kbps is supported. Note: Instructions on how to