similar to: ISDN / Analog

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

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

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

Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) I can see the call answered/placed on the CLI and then silence follows. I've been provisioned 25 out of the 31 channels only

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

Well, it works for me .. YMMV. Yesterday I had a problem where I had a meridian talking to * via a PRI card, and from * to the pstn via an isdn30 link. The problem was that if the number was bad, or engaged then the meridian line simply dropped, not giving the operator any indication of what occurred. With much help from this list, I managed to construct a dialplan which solved our issues.

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

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 guys, Currently I just purchased the TE205P for the installation of my ISDN30 setup. I compiled, setup the card correctly I think. When I do the ztcfg -vvv it shows all 31 channels clear, with 30 bchannels using for voice ranging from 1-15, 17-31, and 16 is the dchannel. However, when I do a cat /proc/zaptel/1 I see only 24 channels configured, and then, when I do a cat /proc/zaptel/2 I see

Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's

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?