similar to: Silence supression

Hey all! Have configured a Cisco 7960 with no problems, put up an TFTP server and it downloaded new sip binaries ? all went well. However, now I am having trouble getting two 7040s to work. Basically, my problem is the above stated error message. If I had any entries in the TFTP logfile, I could probably solve this problem. But the thing is, it just doesn?t look as if the phones are

I am planning to sign up for a VoIP service in the U.S. Can anyone recommend anything cheap, reliable and good quality? I want to use it for my primary house phone (I also own a cell phone). I also want the service to be asterisk friendly so I can play with it :-) Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi ocfs2 developers, there are some news about the schedule for a new ocfs2 release that solve the actual bug/limitations? I can see an 1.4.7 release tagged here: http://oss.oracle.com/git/?p=ocfs2-1.4.git;a=summary Is there a planned release date? in my environment (about 5000000 files with 300000 new files/deletion per day) I see load until 1000 and I/O almost blocked for some hours of the

OK I have to ask. Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem?

http://bugzilla.mindrot.org/show_bug.cgi?id=225 Summary: Supression of login warning banner for noninteractive commands Product: Portable OpenSSH Version: 3.0.2p1 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P4 Component: ssh AssignedTo:

http://bugzilla.mindrot.org/show_bug.cgi?id=225 ------- Additional Comments From dtucker at zip.com.au 2003-05-27 22:10 ------- Created an attachment (id=311) --> (http://bugzilla.mindrot.org/attachment.cgi?id=311&action=view) Suppress banner when -q specified (against OpenSBD) What about suppressing the banner when -q is specified? That way it's on by default but you can

http://bugzilla.mindrot.org/show_bug.cgi?id=225 djm at mindrot.org changed: What |Removed |Added ---------------------------------------------------------------------------- OtherBugsDependingO| |627 nThis| | ------- You are receiving this mail because: ------- You are the assignee for the

To creat a series of plots I've constructed the following, however this doesn't work because the colnames being feed into n are embraced by quotes which is stuffing up the plotting function. Is there a way to suppress to quotes? coercion in to a different format?? for (n in colnames(tt[7:36])) { par(ask=T) attach(tt) plot(july~n) ... }

Hi, I have a dataset where the residual variance decreases with on one of the predictors (population size). Currently, the full model looks like this: prior<-list(R=list(V=1e-16, nu=-2),G1=list(V=diag(2), nu=2)) m<-MCMCglmm(response~poly(population size,2)*poly(other predictor,2)+time, random=~us(1+time):population, data=data, prior=prior) Basically, it's a random regression with

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 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:

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

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

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

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

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

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

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

I?ve read various stories about the Asterisk SCCP implementation. Some say it works fine, others say it doesn?t ? stick with a SIP phone! My dilemma is that I?d like a WI-FI phone with multi-line capability, and so far Ciscos 7920 is the only one I?ve found. The disadvantage is that it cannot be upgraded to support the SIP-protocol, so if I purchase such a phone I am stuck with the SCCP.

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