similar to: IAX Channel Capacity

Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody

So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local & long distance for free. It also does 'internet'. 'Just give me

Now that Solaris 10 11/06 is available, I wanted to post the complete list of ZFS features and bug fixes that were included in that release. I''m also including the necessary patches for anyone wanting to get all the ZFS features and fixes via patches (NOTE: later patch revision may already be available): Solaris 10 Update 3 (11/06) Patches sparc Patches * 118833-36 SunOS 5.10:

Fresh Ubuntu 13.10 server build with all the latest patches, installed Xen 4.3 and XCP-XAPI. Got Xenbr0 working properly and created an LVM storage repository and a CIFS ISO repository. I can create VM''s no problem at all but whenever I try to start them I get the following error from the log. Only Windows VM. The error comes up almost instantly when trying to start it. Not sure where to

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

Hello, I have my sip stuff seemingly working fine as well as my zaptel stuff working great... But I have a problem with sip registration timers (I'm guessing here). In my extensions.conf file I have this... exten => 2244,1,Dial,Zap/2|25 exten => 2244,2,Dial,Sip/brian|25 exten => 2244,3,VoiceMail,u2244 But if I close my sip phone and a call goes through it will still wait the 25

I'm working on setting up a macro that will allow users to call their own DID number, and when they hear their voicemail greeting hit the * key and be prompted for their password to check vmail. For some reason though the background command isn't working as I'd expect it to: [macro-vmessage] exten => s,1,Answer exten =>

I think the problem it's in your dialplan, extensions.conf: ; voicemail management [voicemail] include => misc exten => 6245,1,VoiceMailMain2() exten => 6245,2,Hangup() Check the last line, I have the same problem and was because I wrote wrong the Hangup instruction... Regards! Date: Thu, 10 Jun 2004 09:54:32 -0600 From: Chris Hirsch <chris@base2technology.com> To:

Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I

Hi I hope this isn't off topics, but I have always found when I stepAIC() some glm I get an improvement in accuracy and kappa, but I have just done a case where I got a marginal deterioration. Is this possible, or should I be going through my figures carefully to see if I have messed up? Stephen Choularton 02 9999 2226 0413 545 182 -------------- next part --------------

During hardware and network impairment testing, I sometimes just hold down a DTMF key to keep the audio busy. I have found that if I restart the receiver while the transmitter is still encoding DTMF, the receiver plays out a different tone, until the DTMF key goes off an on again. Also, today I captured some Speex frames during a tone (but not including the start of the tone). When I later fed

Has anyone else noticed a significant zfs performance deterioration when running recent opensolaris bits? My 32-bit / 768 MB Toshiba Tecra S1 notebook was able to do a full opensolaris release build in ~ 4 hours 45 minutes (gcc shadow compilation disabled; using an lzjb compressed zpool / zfs on a single notebook hdd p-ata drive). After upgrading to 2007-05-25 opensolaris release bits

Hello list. I'm new here, and I'm joining because I'm having a problem wich I can't figure out the solution myself. Here's what's happening: A Samba 3.0.24 is serving files to seven Windows XP clients. Three of those clients run an application with a BDE database. The problem is that when running this application on Samba, it gets very, VERY slow, causing it to hang. I

>From what I have read on some mailing lists, there is a jack module for ices2 that works quite well. Why isn't this information on the ices2 web page or documentation? Ices is useless to me without this, yet so awesome with it. I saw requests from people for you to do this over a month ago. What's the problem? Thanks, Daniel P.S. I couldn't search this list to find out if

Dear All, I wonder if anyone can please offer any advice on a model including 2 fixed effects and 1 random effect, as well as a covariate? The experimental design is as follows: I have a two by two factor design, where the two factors, Age (A) and Group size (G), both have 2 levels (old or young, and 1 or 3 respectively), and I am interested in the effect of these factors upon a continuous

Hello all, A few months ago I saw a statement that L2ARC writes are simplistic in nature, and I got the (mis?)understanding that some sort of ring buffer may be in use, like for ZIL. Is this true, and the only metric of write-performance important for L2ARC SSD device is the sequential write bandwidth (and IOPS)? In particular, there are some SD/MMC/CF cards for professional photography and

Folks! I discovered some serious problem with several Sipuras 3000 but I don't know if the problem is with them or Asterisk. Basically, if I call a Sipura PSTN line, when there is a call already in progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I am able to get through and connect to dialed number. The other call gets disconnected but the originator of the

Don & others, Thank you for your answer. The fog maybe lifting ;). The zaptel.conf file has the following in its comments: # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a

All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing

Hi, I managed to fix 128kbit for an IP address of 192.168.200.3 as below: - tc qdisc add dev eth3 root handle 1: cbq avpkt 1000 bandwidth 100mbit tc class add dev eth3 parent 1: classid 1:1 cbq rate 128kbit allot 1500 prio 5 bounded isolated tc filter add dev eth3 parent 1: protocol ip prio 16 u32 match ip dst 192.168.200.3 flowid 1:1 tc qdisc add dev eth3 parent 1:1 sfq perturb 10 How can I