similar to: More asymmetrical call quality discussion

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

Hi all, It appears that something very strange is going on... Here's the deal -- whether someone calls in to my * server or I call out (doesn't matter), I can hear them perfectly: no gaps, no packet loss, nothing ... however, when I speak there seems to be very noticable latency and "choppiness" as if there were packetloss or lots of jitter. I'm using SIP for outgoing

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Hi everyone, This happens every time I start winedbg, no matter what program I want to debug. It's ubuntu feisty fawn and I have built wine from sources. user@machine:~$ winedbg "C:\Program Files\Diablo II\Diablo II.exe" WineDbg starting on pid 000a wine: Unhandled page fault on read access to 0x00000000 at address 0xb7d5cc23 (thread 0009), starting debugger... Unhandled exception:

Thanks Brian, and thanks again for the included definitions <grin> - that helped too. Your comments are really helping clear many questions. I suppose our intensions are to become an IXC. So if my local carrier is sporting old technology, they'll provide TDM services. So if I understood you correctly, the "in-band signaling" is typically SS7, and the alternative is typically

Dear all, I have problems to code contrasts for performing an asymmetrical anova with aov(). I am using aov() because I want to get the Mean Squares for further analyses. I didn't find any solution to my problem in the help files of functions aov(), contrasts(), C(), etc. Let's say I have three locations, one with treatment P, and two with treatment C: >

What is the best way to set up a relationship between two tables where table-1 has a one-to-one relationship with table-2 but table-2 has a one-to-many relationship with table-1? For example, suppose there is an events application (seminars, forums, conferences, etc). Each event has only one address but since many events can occur at the same location throughout the year, a given address can

Does anyone know how to suppress the "Missed Calls" indication -- perhaps on a per-line basis -- on the 7960 running SIP? Reason: I've configured a group of extensions to ring for inbound calls and it seems pointless to accrue missed calls on those line presentations. /rg

Greetings folks, I'm currently dealing with a company to let me set Caller-ID-Name on outbound calls. So far pretty happy with their services. The basic service works like this: * CLEC sets Point Code to point to this company * CLEC has to sign LOA saying they give me permission to set the Caller-ID-Name through this company. * I go into web interface and set name. However, the CLEC is

I know a SIP client behind a NAT trying to peer with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.com

Hello *, I've rc3 on Debian Lenny compiled from source. $ wine --version wine-1.0-rc3 However I typoed that command by mistake: $ wine -version and wine went looking for -version.exe. The strange behavior is the output. Is the folloing normal? $ wine -version wine: Unhandled page fault on write access to 0x00452000 at address 0x7efa04f4 (thread 0019), starting debugger... Unhandled

Hello everyone, I'm hoping someone can help me with this. I have a business customer in the U.S. (Michigan, AT&T Territory). I need to get 4 trunks into an asterisk Box. My intention is to use an Eicon Diva Server card with 2 BRI Circuits. The reason for this is that the business needs DID's on the trunks (20 of them). A full or fractional PRI is overboard for them, as they will

At the company where I work we use an application, that is made by an outside company. for now it works OK under windows, but now we plan to migrate the users to Linux and they don't plan to write a Linux app. So we are trying to run the app under wine, but it fails during installation. We use the latest development version of wine (1.1.24), because of another app, that needs it (it doesent

I'm trying to get the PocketFMS flight assistant program to work under Wine. This program is available for free at www.pocketfms.com . The installation of PocketFMS works OK with a few 'fixme' warnings, but running the installed program causes an immediate crash. As this program is still in development itself and not designed to run on anything but Windows, I'd like to know if

Hello! I'm trying to find where to send a bug report/fix, and this mailing list is the best I found. I noticed several problems in how Compiz handles full-screen windows*, and I hacked a bit around the source to try to find their source. (*: I use this pretty much as a replacement for full-screen, all the time.) The first two are together below, because they seem caused by the same code. For

Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine.

http://www.boingboing.net/2006/06/05/play_zork_by_phone.html Let me preface this idea with one comment: I don't have the time to do this - I don't even have time to eat these days. But someone out there has the cycles to do this... and it would be very cool. OK, so now Zork is attached to Asterisk, but using the less-than-clear Festival engine. There are beta tests of the LumenVox

I have the following configuration: CLEC -> T-1 -> Asterisk -> Adtran Channel Bank -> (analog) -> Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the

My client would like to add new functionality, such as call queuing, to their PBX. They presently have a Telrad PBX; it has a T1/PRI circuit from the CLEC and a few POTS lines. They use DID for some incoming calls and an attendant for calls on the main number. The client's commitment to customer service will not allow us to outright replace their PBX with a new system--we could never

Hello. For some reason, trying to run any Direct X 10 games (including the DLL test) blows up. My card is an Intel Corporation Mobile GM965/GL960 Integrated Graphics Controller, and uses the intel Xorg driver. Here is what happens when I run the test: Code: make[2]: Entering directory `/home/dan/Software/wine-1.1.10/dlls/d3d10/tests' ../../../tools/runtest -q -P wine -M d3d10.dll -T