similar to: How does ISDN really work?

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

Following is the last few lines of output when i try to launch Asterisk:- [app_zapscan.so] => (Scan Zap channels application) == Registered application 'ZapScan' [app_saycountpl.so] => (Say polish counting words) == Registered application 'SayCountPL' [func_cut.so] => (Cut out information from a string) == Registered custom function CUT == Registered custom

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?

No Jeremy, excuse me, the error was in my email. The correct command is /bin/echo "Channel: Local/$1@chiamamezzi-dialout";\ /bin/echo "Variable: callid=123456|number=$1|url=pippo|menuid=FOP|redirectnum=0554202880";\ /bin/echo "Context: chiamamezzi-Wave";\ /bin/echo "Exten: s";\ /bin/echo "Priority: 1";\ /bin/echo "Callerid: Asterisk Automatic

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

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

Hi there, I am seeing some very interesting thing with the latest Zaptel 1.2.X, hope may be someone can shed some light on this. Normally, to dial via your Zaptel T1 card, you would do something like: ;Dial to PSTN exten => _9.,1,Dial(Zap/g1d/{EXTEN:1}) by not adding any option after the extension e.g. no "r" and no "m", Asterisk will pass thru the session media from the

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!

ICecast i capable of sorts of streams, that is not your problem. Your stream source client (DSP) is the part who must be capable of streaming the format you want. For AAC use oddcast DSP www.oddsock.org capabale of AAC, LAME Mp3, and OGG (Free to use) very good at OGG specially with the vorbis 1.1 aoTuVb3 DLLs For AAC Plus (HE_AAC) use Orban opticodec for PC (witch is capable of every bitrate

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 have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

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

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile

Hi, on a configuration with one external ISDN S bus (to telco) and one internal S bus (to ISDN telephone), where Asterisk is in the middle (using HFC hardware), I noted the following: - when a GSM phone or ISDN phone calls in, the Transfer capability is Requested transfer capability: 0x00 - SPEECH - when an analog phone calls in (either from an analog line or an analog ISDN

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb