similar to: OFF TOPIC: Core router upgrade for a voip colocation center

'morning everybody, Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e

I would like to ude the function "qdagum", to compite quantiles of dagum distribution, but if I use the form: >qdagum(0.1, 4, 7206, 0.648) Error in qdagum(0.1, 4, 7206, 0.648) : object 'Scale' not found Error will apper, Could you help me? Thank you very much in advance, Alena -- View this message in context:

I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 /

I am running an Asterisk server through a Cisco 7206 PSTN gateway. I am able to make outgoing SIP calls without a problem, though incoming calls have been somewhat of a problem. I am not sure exactly how sip.conf should look in such a scenario. I believe most Cisco gateways are just managed through ACL's, with no authentication, so I think I have the outgoing "peer" statement

Newbie to Asterisk I've been looking around for a little while, can't seem to find some sample configs for using a Cisco 7206 as a gateway. The below link is an initial plan of an Asterisk solution that may replace our Cisco Call Manager 3.1/ IPCC / IVR setup. We currently have all of the hardware below. Just take a peak and see if there is anything that is off base. I don't know

# I would like to outline the squares in the legend with a black line. Does anyone know how to do this? x.t <- structure(c(5987.387, 4354.516, 3685.789, 6478.592, 5924.315, NA, 8386, 5559.468, NA, 4651.273, 3967.5, NA, 4339.167, 5053.56, NA, 4631.978, 4808.694, NA, 5217.306, 4017.632, NA, 5846.903, 3969.883, NA, 3867.825, 3910.236, NA, 3886.434, 3782.094, NA, 3959.668, 3961.286, NA, 3848.853,

Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy

On 09/19/2010 03:48 PM, Ben Laurie wrote: > Another suggestion I've been given is this: > http://gcc.gnu.org/wiki/CompileFarm. You wouldn't even have to supply > hardware. The compile farm is a set of machines that is shared between different users. They are great to try your code on different architectures, but the level of control is a lot lower than the level you get from

Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. Regards, Sahil Gupta VoiceValley

Try these guys: http://easyspeedy.com/ Haven't tried them, but when I was looking into a while back they responded quickly. -- Bjorn -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Sahil Gupta Sent: Wednesday, May 03, 2006 1:47 PM To: asterisk-users@lists.digium.com Cc: asterisk-biz@lists.digium.com Subject:

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

Another suggestion I've been given is this: http://gcc.gnu.org/wiki/CompileFarm. You wouldn't even have to supply hardware. On 17 September 2010 17:13, Daniel Berlin <dberlin at dberlin.org> wrote: > Can't offer it in the bay area, but i can offer space in oregon state > university (which is where the osuosl machines reside). > http://osuosl.org/services > > Why

Hi Danny, On Fri, Sep 17, 2010 at 9:13 AM, Daniel Berlin <dberlin at dberlin.org> wrote: > Can't offer it in the bay area, but i can offer space in oregon state > university (which is where the osuosl machines reside). > http://osuosl.org/services > > Why do you need access if you have people willing to handle support > tickets and remote console/power control cards?

Can't offer it in the bay area, but i can offer space in oregon state university (which is where the osuosl machines reside). http://osuosl.org/services Why do you need access if you have people willing to handle support tickets and remote console/power control cards? I have yet to see a problem that i could not solve (since the cards can display console, control power to the machine, etc)

Here is the setup: Dell 6250, Quad Proc P3 500Mhz. Digium Single Span T1 card. System has 71 sip peers/users. All calls are G729; we have 10 licenses. All calls follow this path: UA -> Asterisk -> Digium PRI -> Class4/5 switch. The switch dictates if it should go out local PRI for local termination or out PRI to our Cisco AS5300 for LD termination. Our biggest problem is echo.

Hi all, I don't have very high hopes, but does anyone happen to have available some kind of colocation services in the Bay Area that they could donate to LLVM? I would really like to establish a proper LLVM lab somewhere instead of relying on hardware which is residing at various companies. I would be willing to personally donate some hardware, but don't have any place to put it. My

Well, I finally did it..and used centos. I started out in 1997 with my little website with not much on it. It was a shared host account. Eventually I added some more sites and got to grow out and went to this new fangled thing called VPS. Lots of problems plagued me throughout that experience, some on ensim control panel, some on others. When chost.net (think that was their name) blew up (1999

> Message: 5 > Date: Fri, 12 Sep 2003 13:24:13 -0400 (EDT) > From: "David C. Troy" <dave@toad.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > All, > > I know you can use, say, a 2620 w/2 port FXO card as > a SIP gateway. Clearly you can use the 5300, 5800, and > MGX8850 too. > >

Completely fresh,virgin install of b115 SXCE ,zero modifications done. If select XVM from boot menu,it comes all the way till "Starting Desktop login on Display:0..." and reboots. Regular Solaris(not XVM) boots fine. Intel XEON CPU,8G RAM Intel S3210SHLX motherboard(http://www.intel.com/Products/Server/Motherboards/Entry-S3200SH/Entry-S3200SH-overview.htm) Please help and thanks a lot

On 18 Oct 2015, at 23:08, Sanjoy Das <sanjoy at playingwithpointers.com> wrote: > > Supporting only basic block level granularity for "try ranges" may not > be sufficient for Java -- if a basic block has more than one null check > in it then throwing the NullPtrException for the first null check (if > it fails) is semantically different from throwing the