similar to: ISDN / Analog

Hi, You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy. Idris _____ From: phil.dawson@marnock.com [mailto:phil.dawson@marnock.com] Sent: Wednesday, June 21, 2006 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Monitor a particular SIP call for training purposes Hi, I've been asked if it is possible to allow a user to

Phil I have very good experience with the vegasteam ATA's devices.(you might also want to look @ sipura ATA's, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash ________________________________ From: asterisk-users-bounces@lists.digium.com

Hello: I have an odd problem with an IBM EServer XSeries 305 (aka x305). Sometimes it seems to just sit there, completely unresponsive to the command line (ssh and console), for several seconds. Other times it works normally. top will sometimes die with a floating point error while waiting for it to start. When top does work, it often shows a system load of 0.5, with high interrupt load. At

I've been attempting to calculate the predictions from a poisson glm object, along these lines: predict(foo.glm, type = "response") and predict(foo.glm, type = "response", se.fit = TRUE) foo.glm is arrived at this way: foo.glm <- glm(Insects ~ Dad * Mum + Location, offset = log(MM), family = "poisson", data = model.df) There are two

Hi, I'm looking for IBM hardware to support: 100 SIP hard phone users 10 fax machines on SIP ata's maybe later an additional 100 sip soft phones. Initially, all calls will be through PRI. Some conferencing. Don't know yet if this will even get used. Using 1.4 + ( probably business edition ) I'm looking for anyone who some experience / gotchas. I've google'd and

Still finding my feet here. I need a server which can take approx 100 sip users accessing 24 channels through pri. Approx 20 concurrent calls. 6000 calls a month. Are there any "rules of thumb" when it comes to sizing hardware. I've checked the wiki but nothing close to what I need plus some of the information is really old now and may not be relevant. Also, what server

Greetings (first post to the list)... The following patch is required to allow bridging to occur on an IBM x305 (1RU server). (It has been posted to the kernel-developers mailing list previously.) Background: - The Dual Broadcom NICs in the IBM x305 have a broken checksum routine which wants to calculate the checksum the ethernet packet where the 'source' IP address is the IP

Hello, I am have a strange problem with remote samba servers and I am hopping some one can give me a insite into what is happening. First is the Setup I have 2 remote sites connected to a central site by 2 T1 Lines. The two remote sites each have a IBM X305 Running RedHat 9 and Samba 3.0.2a per/Site. Each box per site has been configured to be the local master(cross Subnet Browsing).All Sites use

I'm looking for a function in R similar to t.test() which was generously pointed out to me yesterday, but which can be used for normally distributed data. To recap yesterday: > x <- scan() 1: 62 52 68 23 34 45 27 42 83 56 40 12: Read 11 items > alpha<- .05 > t.test(x) One Sample t-test data: x t = 8.8696, df = 10, p-value = 4.717e-06 alternative hypothesis: true

Hi, I'm currently looking to run Asterisk in the office to replace an old PBX and would appreciate a little help. We are moving offices and will have 8 digital lines. My questions are: As there are 8 digital lines is this known as PRI? Which Digium card would be the best fit? Would you recommend looking at the echo cancellation cards? We are UK based: is caller id supported by Asterisk

Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format:

Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Thanks,

Hi! > > > Since it is a PCIe card, it does not have the ability to host hardware > > > devices for networking, storage and console. We provide these devices > > > on X100 coprocessors thus enabling a self-bootable equivalent environment > > > for applications. A key benefit of our solution is that it leverages > > > the standard virtio framework for

On Wed, 2013-08-21 at 10:51 -0700, Sudeep Dutt wrote: > ChangeLog: > ========= > > v2 => v3: > a) Patch 1 data structure cleanups, header file include cleanups, > IDA interface reuse and switching to device_create_with_groups(..) > as per feedback from Greg Kroah-Hartman. > b) Patch 7 signal documentation, sleep workaround removal and sysfs > access API

On Thu, 2013-08-15 at 12:14 +0200, Pavel Machek wrote: > Hi! > Hi! > > > > Since it is a PCIe card, it does not have the ability to host hardware > > > > devices for networking, storage and console. We provide these devices > > > > on X100 coprocessors thus enabling a self-bootable equivalent environment > > > > for applications. A key

On Thu, 2013-08-15 at 12:14 +0200, Pavel Machek wrote: > Hi! > Hi! > > > > Since it is a PCIe card, it does not have the ability to host hardware > > > > devices for networking, storage and console. We provide these devices > > > > on X100 coprocessors thus enabling a self-bootable equivalent environment > > > > for applications. A key

Hi Everyone! I live in Sweden and can not get CallerID to work on analog incoming lines. I m trying to find out if DTMF style CallerID works on a FXO card (X100). I`v seen one solution with a modem attached in parallel with the X100 just to provide the ID on its serial port. It must be much better if this can be implemented in to the X100 driver. Any info about this would be highly appreciated.

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with

On Sat, Oct 23, 2004 at 08:37:53PM -0700, jm@motherfish-iii.xiph.org wrote: > Modified: > trunk/speex/libspeex/cb_search.c > Log: > Shouldn't warn with fixed-point anymore Does this mean fixed-point speex is debugged, now? Is there a strategy, then, for getting Speex into consumer-premesis VoIP equipment?

On Wed, Jul 24, 2013 at 08:31:31PM -0700, Sudeep Dutt wrote: > An Intel MIC X100 device is a PCIe form factor add-in coprocessor > card based on the Intel Many Integrated Core (MIC) architecture > that runs a Linux OS. It is a PCIe endpoint in a platform and therefore > implements the three required standard address spaces i.e. configuration, > memory and I/O. The host OS loads a