search for: noices

i have been using the asterisk for some three weeks. Previously i was using the softphone iax-phone and now i have to shift to the sip phone xlite. The problem is that there's always unbearable noice in sip to sip calls. Is there any way to get rid of this???? Kindest MM Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I use acts_as_ferret on an app that is in Danish and English. In Danish english words like "and" and "under" has meaning. Is it possible to make ferret search for these words? As it is now a seach for "under" returns nothing even-though I know the word is present in the index. Cheers Mattias

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems like SJPhone is using a version of G.729 other than Codec G.729 (Annex B) , probably...

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

Hello all, I am facing as serious problem when running asterisk in HP server.We are developing application to make the outbound calls in PRI lines .We normally uses IBM machine as our servers ,and it was working fine for all installation.For the cost reduction we this time tried with HP server. Model(HP proliant ml110). When we make the calls the there is a lots of disturbance in the sound even

And someone goes and redesigns the look and operation of the Trac. Quite noice, but a couple of comments: * The main toolbar ("Wiki", "Documentation", "Timeline") ends up under the puppet logo on the left hand side, which is a pest. * I can''t click on any link, text box, or button in the main part of the page. All of the navigation links work fine, and I

Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next

How can I add some size to SPEEX encoded packets without affecting decoding results? I need it to fit smaller (due to VAD) packets in CBR acm-wav file. i.e. I have 10 bytes packet with 0.02s of silence (only background noice without speech). I need to fit it in 16kbps CBR wav file, so I need to put 30 additional bytes to this packets, but decoder should still decode only 0.02s of background

Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server

Hello all! First of all, this is my first post to the list. I've tried to find my answers in the forums and by Googling , but no luck. My apologies if this question has been answered before. I've set up an Asterisk box with four local SIP users. The Asterisk box uses a SIP provider for placing external calls and receiving incoming calls as well. In other words, there's no PSTN

On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> I am currently having a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >>

...ow or just another boring sunny day. Let's find something cool to do during this weekend! Join the cool crowd that tests the test branch during evenings and weekends. The dudes and dudettes that proudly contributes by reporting everything from simple spelling errors to crashes and strange noices from their Asterisk boxes. The people who knows what is going on in the Asterisk development circles - the Asterisk Test Team! I've updated the test branch to the latest version of my SIP peermatch code. This is quite a large code change, but not as large a functional change. However, it c...

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

Aaaargggg.... The S-1-5-18 AND sid S-1-5-32-544 did not resolve. But sid S-1-5-32-544 first not then later on it works.? Sorry about the noice, but that one i wanted to point out also. I hate it when im almost done with typing and mr Penny comes first. ;-) :-p Greetz, Louis > > > > > -----Oorspronkelijk bericht----- > > Van: samba [mailto:samba-bounces at

On 07/30/2013 10:03 AM, Star Tan wrote: > Hi Tobias and all Polly developers, > > I have re-evaluated the Polly compile-time performance using newest > LLVM/Polly source code. You can view the results on > http://188.40.87.11:8000 > <http://188.40.87.11:8000/db_default/v4/nts/16?compare_to=9&baseline=9&aggregation_fn=median>. > > Especially, I also evaluated

Hi, silly question I feel a bit blind and lost right now :D.... I have a share on which I can delete files (what I should not be allowed to) which have the chmod 755. I'm not the owner, I'm just in the group which may read that file and Share ... How are ACLs and write/read permissions organized. ... Thanks for feedback/help/suggestions. Regards . Götz

I'm wondering if anyone has seen (heard!) this before. I have a site which has Grandstream Budgetone 100 phones (don't laugh, they weren't my choice and I was quite angry when I heard they'd been installed )-: They have an asterisk box with a TDM400 card in it with 4 FXO ports and 4 lines to the telco (BT, in the UK) It basically works and does what it says it's supposed

Hi! Jim is right. There is a bug in all GnuPG versions up to 1.0.3: If you have more than one cleartext signature in a file (or pipe that to gpg), gpg does not compare each signature but flags each document as good or bad depending on the first document in the file. This is a very serious bug in gpg's verification function. I have made a snapshot version which corrects this bug available

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being