similar to: ulaw blank spots but gsm fine

For some reason a couple weeks ago users began experiencing garbled audio in one direction when dialing out via our VoIP provider. This happened at multiple sites simultaneously. The VoIP provider doesn't think it's their problem. If I switch to another codec so that Asterisk transcodes everything is fine. On conference calls (where Asterisk gets in the middle to relay ulaw to all

tuning for ulaw g.711 - Polycom IP500 I've got QoS ironed out at this point (dedicated section of bandwidth with VoIP having priority), yet ulaw is still unusable due to small 'holes' in the audio. Running iperf tests shows packet loss at very low percentages (< 0.02 %), and when we do drop, it's only 1 packet at a time. I thought ulaw was supposed to handle this OK.

I've created a pretty complete HOWTO on creating a Linux Bridge (using Fedora) to shape LAN <--> WAN traffic. It includes installation instructions, a script to configure the bridge (which you install as a service), and 2 scripts to configure the network interfaces using traffic control. http://www.burnpc.com/website.nsf/all/3a64a6369757819686256f960068ad75!OpenDocument If anyone

For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100

Hi all, i have a little problem to understand this warning message, it's annoying and it cause a lot of spurious in the log files. Im working with this scenario: a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are always routed to this. a list of sip UAs that potentially can use any codec apart g729/g723. I setup the asterisk to do as mediaproxy so directmedia=no and

Hi, I am testing tinc for a very large scale deployment. I am using tinc-1.1 for testing. test results below are for tinc in switch mode. all other settings are default. test is performed in LAN env. 2 different hosts. I am getting only 24.6 Mbits/sec when tinc is used. without tinc on the same hosts/link I get 95 to 100 Mbits/sec using iperf. Over Tinc: iperf -c 192.168.9.9 -b 100m -l 32k -w

Hello all, We've discovered that VoIP (IAX2) + Citrix + Video is pegging the measly CPU on the Netopia router our ISP provided. We've got 3Mb/3Mb and will increase to 4/4 next year. The Netopia simply breaks out our WAN IPs, and we've got a switch hooked up to it on the inside (Actually I've got a QoS box in-between). ------------- | Internet | | on Cat5 | -------------

I'm looking for the best recording I can get of Allison saying "Steve" or "Ian". I found gsm recordings of both out there but was looking for something higher quality. Can anyone point me in the direction of a WAV or ULAW recording of those names? Thanks in advance Steve

Hi Guys, I have noticed a weird behavior in 1.4.12. When using Authenticate or DISA in the dial plan the channel immediately switches to gsm format (if you request a password) or slim (if you run DISA without password). The debug log says... =============================== [Oct 14 21:23:00] DEBUG[9013] channel.c: Set channel SIP/1970xxxxxx-0821aad0 to write format gsm [Oct 14 21:23:00]

> From: "John Hughes" <john at calva.com> > To: "Asterisk Users Mailing List, Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Sent: Thursday, May 14, 2020 2:10:45 AM > Subject: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and > alaw; asterisk wants to translate g729 -> alaw. WHY? > I am having a

Hello, I have figured out that my audio problem was just how I was converting the sound files. I am trying to convert the Asterisk gsm files to ULAW. I just did a: sox file.gsm file.ul, open it in Audacity. I used: Project, Import Raw, U-law, No endian, 1 channel, start offest 1 byte, sample rate 8000hz. The file sounds fine in Audacity. Now, if I do a record on Asterisk, using pcm, au, or

Hello all! If I place a call to our number, the call is routed to our Asterisk box from teliax --> IAX2 --> firewall w/ port forwarding --> * If that caller dials an extension that rings an outside line, where our * box makes an outbound connection to teliax to terminate the call, we get choppy audio. Internal extensions have been dialing outbound calls no problem for over a week. What

On 1/28/20 12:39 PM, hw wrote: > On Tuesday, January 28, 2020 9:00:22 AM CET Nataraj wrote: >> On 1/26/20 5:44 PM, hw wrote: >>> On Sunday, January 26, 2020 11:18:36 PM CET Pete Biggs wrote: >>>> First of all - disclaimer - I'm no network specialist, I just read and >>>> am interested in it. I may get things wrong!! >>>>

On 14/05/2020 08:10, John Hughes wrote: > > I am having a problem with one of my callers who is using either g729 > or alaw.  I can do alaw but not g729 so asterisk should negotiate alaw > right?  In fact from the sip debug it looks like it does, but then I > get the dreaded "channel.c:5630 set_format: Unable to find a codec > translation path: (g729) -> (alaw)"

On 11/04/2008, ogg.k.ogg.k at googlemail.com <ogg.k.ogg.k at googlemail.com> wrote: > > Ogg/Spots has no implementation since we agreed that Ogg/MNG would > > fully cover all the functionality of Ogg/Spots. However, Ogg/MNG has > > not progressed as much as we would all have liked. > > > > So, if you would like to get images into Ogg, I'd recommend >

Hi, please ignore this message if this is a stupid question: Would supporting APNG instead be any easier? Cheers, KJ On Fri, Apr 11, 2008 at 3:52 PM, Ian Malone <ibmalone at gmail.com> wrote: > On 11/04/2008, ogg.k.ogg.k at googlemail.com <ogg.k.ogg.k at googlemail.com> wrote: > > > Ogg/Spots has no implementation since we agreed that Ogg/MNG would > > >

Hi, are these two dead, or just progressing slowly ? The only code I've seen about Ogg/MNG was a player in ogg-tools, and I've seen nothing for Ogg/Spots. I'd looked at those before adding bitmaps to Kate, and I went back to that a few days ago (storing subtitles as images from a dvd - works great in vlc, but I store images without compression). If they are dead, I'd be inclined

On Thu, May 14, 2020 at 11:31 AM John Hughes <john at calva.com> wrote: > On 14/05/2020 08:10, John Hughes wrote: > > I am having a problem with one of my callers who is using either g729 or > alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? > In fact from the sip debug it looks like it does, but then I get the > dreaded "channel.c:5630

Dear all, I am looking for a way for creating a gui for managing qos in linux , i came across snmp extension to net snmp at http://x-ray.prokon.cz/data/snmp/ and i found qosd-0.0.1-13122003.tgz , in which there is a client server program using soap which was realy intresting but when i tried to compile it ended up with many errors ,, does and one have tried it , or have

On 4/11/08, ogg.k.ogg.k at googlemail.com <ogg.k.ogg.k at googlemail.com> wrote: > how (or do you mean ogg/mng or spots) ? People want an official standard on how to embed JPEG/PNG in single stream Vorbis files, because one of the extensions/hacks of MP3 ID3 allows it. > uuencode ^_^ Hmm...