similar to: DTMF CLIP in Sweden and others

What server hardware would you recommend for an Asterisk system which are really critical? The additional hardware will probably be two digium TE110P cards, and an Adit 600 platform. If it's possible to run on -48VDC, It would be great! Are there any experiences with any HP or FujitsuSiemens systems? Or other "complete server" systems? Thanks! BR Daniel Nystr?m

Are there much performance differences when using XEON or not? In my case, I will go with muLaw both in and out of Asterisk. Are there really any processing at all if it's using same codec all the way?

Hi again folks! ;) As before, I will transform one E1 30 Channel PRI into 30 FXS channels using Adit 600. Now I'm into choosing server platform. And the two opponents are: * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1) * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1) As I've seen people having problem with HP server, I havn't looked at it at all. What experience do you

Sorry to all that are angry about the form of my previous mail. I didn't realise what would happen :((. Here it is in (hopefully) plain text (if my mailer doesn't spoil it again): ############## Dear developers, I have a problem with some discrepancy between R 1.2.1 for Windows and R 1.2.2 (and less) for Linux. While trying to correct the wilcox.test (see my previous bug report) I

I'm trying to use ZapRAS to enable ppp connection through my E1. After the ZapRAS command is executed, all sound is crappy on all lines! The only solution is to reboot the machine (or halt it, and then power it on since Digium's hardware doesn't like reboots). Anyone know how this can happen?! I'm using * 1.0.6 on Dell PowerEdge 1850 which are told (too late though) not to work

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router using MGCP IP protocol, instead of controlling it through an E1. Have anyone tried this configuration? How does MGCP works? I've tried to search for it on Google, but I only find the protocol specification for it. Is Asterisk fully capable of this? I can't find any documentatin covering the use of MGCP in Asterisk.

Where can I get E1 and/or Euro-ISDN specifications/data sheets? Are there specs for other E./G./Q./etc. protocols as well? Thanks!

I've finally got my Adit 600 and are configuring it right now. But I have to say, there aren't much documentation for it. I've setup MGCP and Asterisk seems to find it. But all channels (40 FXS channels) are "Down"! But the MGCP itself is "Up" according to the statistics. I can't find any documents how to set each channel to "Up" in the CLI. Any

A little off-topic maybe, but it's still for the Adit used with Asterisk. ;) I wonder where I can buy 50 pin Amphenol cables, with connector on one side, and open cables on the other for mounting in our own patch panels. In Europe, or Sweden preferably. It's said to be very common on telcos, but I can't find it for sale anywhere! Thanks!

I want to achieve high availability with SMB3.0. Now ，if I shutdown the node of business , that ctdb will take over the public IP ,but the business will stop .I want to achieve the business not stop. My environment is: 1.samba version 4.7.0 2.The system is centos 7.0 3.The cluster is constitute of three nodes But I do not know how to configure samba and how to test it. Can you help me ? thank

I've got some strange problem with NUT on my 5115 on Debian Etch: ---snip--- $ sudo ./bcmxcp_usb -DDDDDDD -a 5115 Network UPS Tools - BCMXCP UPS driver 0.13 (2.3.0-1181) Warning: This is an experimental driver. Some features may not function correctly. debug level is '7' Can't set POWERWARE USB configuration Unable to find POWERWARE UPS device on USB bus Things to try: -

I will be using one E1 to the telco, and there will be 4 static phone numbers, and one number serie with 1000 numbers ranging from e.g. 555-1000 to 555-1999. There will be 30 FSX lines on the other side of Asterisk. Is it possible to "group" those 30 FSX lines pending on which number was dialed? Let's say line 1-4 are for the static numbers, and 5-30 for the other 1000 numbers. Are

According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is this? This should not be a problem for Asterisk? Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN? BR Daniel Nystr?m

Do anyone knows abount European resellers of these products: * Digium Wildcard TE410P * CarrierAccess Adit 600 Preferably in Sweden, but Europe is also better. Have anyone within EU ordered products from these companies directly from the US? In that case, how is the service and delivery? Thanks, folks!

Is it possible to turn off DTMF recognition (and all transfer services etc.) pending on CallerID (or FXS channel)? Some of the FXS channels I will setup soon, is going to work exactly like POTS. It will be used by people not knowing their within Asterisk. They will be pretty confused when "Transfer" is playbacked in the handset. :)

Hi! I've been trying to get ZapRAS or PPPD to work. Neither does! All i get is LCP: timeout sending Config-Requests But after trying, all voicelines get crazy! It sounds like robots when somebody calls! And since the zaptel drivers can't unload (the server hangs totaly if I try!), I have to reboot the whole server! The robot-voice is only on our side, it sounds fine at the other end.

I'm trying to setup DATA calls with Dial(Zap/g1d/12345678), but with PRI DEBUG SPAN 1 on, it seems to connect a regular SPEECH call. I'm using 1.0.6. Is this feature broken in stable release? There seems to be support in the source, but it doesn't work. Does the Telco set what each PRI channel support? Like DATA or SPEECH etc.. Do I have to specify in zapata.conf or zaptel.conf that

Hi, does anyone have any idea whether VIA VT6102 "Rhine II" and Realtek RTL 8100 Ethernet chips are supported by FreeBSD? The hardware list for supported chips does list 'VT86C100A "Rhine II"' but according to VIA website VT86C100A is actually just "Rhine" and it is obviously not the same chip. (The 'VT3043 "Rhine I"' chip mentioned in

Hi -- I''m going to be giving a free (but limited enrollment) one-day Rails workshop in Malmo, Sweden, on September 17. The event is presented by Polar Rose (http://www.polarrose.com). For more info: http://www.polarrose.com/about/ruby-on-rails-workshop-by-david-black Hop on over from London if you''re at RailsConf Europe! (Which [London] is where I am now.) David --

Anyone recommend anyone who can provide me (actually a customer, but I'm asking on their behalf) DIDs in Sweden? They already have an asterisk box (in Sweden), now want a local number for it! Thanks, Gordon