search for: ds0

...ng the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go off hook), but no dial tone. My config is: controller T1 7/1 framing esf linecode b8zs cablelength short 133 ds0-group 0 timeslots 1 type fxs-loop-start ds0-group 1 timeslots 2 type fxs-loop-start ds0-group 2 timeslots 3 type fxs-loop-start ds0-group 3 timeslots 4 type fxs-loop-start ds0-group 4 timeslots 5 type fxs-loop-start ds0-group 5 timeslots 6 type fxs-loop-start ds0-group 6 timeslots 7 typ...

...nts to do a trial before making a large investment. For this test, he is willing to buy a Wildcard TE410P. Hes got a proposal from his pstn provider to convert his analog lines to digital ones, but I have some doubts. Let's say he converts only 2 or three of the lines. He would get two or three DS0's, wouldn't he? Now, if that were the case, would it be possible to connect these to the TE410P? Is it just that only three time slots are used and the others are blank, or signalling is different between the cases you don't have a full E1? Thanks in advance Jaime Atahualpa Lopez Soll...

...u specify in a tablegen pattern that all destination registers are also source registers? I know I could just duplicate them, but I was wondering if there was a way it could be done without doing this. Basically an inplace operation. So for example, take a hypothetical swap w/ sqrt. if I had swap $ds0, $ds1 which swaps ds1 and ds0 and applies sqrt on the registers afterwards. Micah -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20130625/76068770/attachment.html>

I have a PRI that if you dial a number that is busy, the channel does not hang up, it then sends "h|1"to the phone company which will then plays back to the end sip user "You don't need to dial a one or zero" I am running stable CVS-v1-0-01/20/05-02:45:17. I have placed the important bit from the extension and sip configs below. Simplest possible example that will show

...of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am I missing here.. ?? Hmm, but outbound calls would be more complicated I think.. Let see, SIP user dials a number, we'll eventually place a dial out on the MGCP line, but we need to first send a few SIP-T messages to find out where to put it.. Just swiming a...

Has anyone here used T1s with RBS with asterisk? Cary Fitch

...f the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing. I know the T1 is good because I hooked it up to a Nortel KSU and the DID's work fine. I am receiving 4 digits and the T1 only has 4 ds0's. I have attached a sho run, any help would be appreciated. TIA, Jason service timestamps debug uptime service timestamps log uptime service password-encryption ! hostname gw1 ! boot-start-marker boot system tftp mc3810-a2isv5-mz.123-10a.bin 192.168.5.104 boot-end-marker network-clock base...

hi, i have some newbie questions about channel banks. i have an adtran act-1241 sitting around. it accepts D4 modules, and it contains a number of e&m cards. first of all, how does this thing work? a t1 contains 24 channels, and i noticed that the channel bank has space for 24 cards. what do these cards do? what are their outputs? the ones that are in there have some outputs on the front

...e operands adding a constraint. You may be able to define an instruction with duplicate operands and then lower it to one without duplicates using a custom inserter, but I'm not sure if that is really any better. -Tom > So for example, take a hypothetical swap w/ sqrt. > if I had swap $ds0, $ds1 > > which swaps ds1 and ds0 and applies sqrt on the registers afterwards. > > Micah > _______________________________________________ > LLVM Developers mailing list > LLVMdev at cs.uiuc.edu http://llvm.cs.uiuc.edu > http://lists.cs.uiuc.edu/mailman/listinfo/...

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate =&gt...

...in Advance for any help. David Stubbs ============= Questionnaire Follows ================= 1. DS1 Facilities Required: - Superframe or Extended (ESF/SF)? - Line Coding (B8ZS,AMI,. Etc.)? - Number of DS1 facilities required? - Channel numbers used for this trunk group? 2. Circuit Signaling (DS0 or DS1 Signal): DSO Analog Line Trunk Interface 2 wire, 4 wire, (or "na", not applicable)? - Supervision Loop Start, Ground Start, Reverse Battery, E & M "DNIS - Dialed Number Identification service" Not applicable with DS0 Analog Line trunks. DS1 Digital...

Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the Masses

...o have 2 * servers go down at any given time and not actually have the carrier see that anything has happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike

...ver and I see the name I statically assign in the Cisco appears on the terminating end (linksys ata) I use the command in the Cisco under sip-ua calling-info pstn-to-sip from name set name timers buffer-invite 5000 I have also tried to add the commands... remote party-id -voice service voip,sip ds0-num Basically I need to take the field Remote-party id and place it in the sip message "From" Here is some debug from the sip messages in the Cisco... this is an example of "no caller id Name" Sent: INFO sip:5132017005@65.23.9.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 66.148.165.xx...

While I was tying down a supernet to the discard interface, the box crashed on me. Its a STABLE box from June 4th. I was in zebra at the time and thought I would route a /24 to ds0 instead of to the IP on ds0 (which I had done for a number of other aggregate routes). The only other "strange" thing about the box is that ds0 is loaded via kld. I will see if I can reproduce it on a non production box. ---Mike IdlePTD at phsyical address 0x0032e000 initial pcb a...

...in the Digium list archives that discussed ISDN BRITE support. There was also some discussion on 4:1 bitrate conversion. I did some searching on the source code and didn't see any reference to BRITE or bitrate conversion. Does the current code properly pass a D-channel from a BRI port to a DS0 channel on a channelized T1? BRITE seems pretty straight forward, but I didn't see any examples on how to do this. Are there any plans for supporting the more complicated bitrate conversion that muxes the D-channels from BRI ports to a single DS0 channel? The context for these questions is t...

...PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15 is-slave = off DS0 Provisioning: analog-begin = 1 analog-end = 16 data-begin = 17 data-end = 24 alignment = same (The following is what our Vina T1 Integrator currently has in its settings. Our Linux box will replace the Vina) Synchronous Interface: encapsulation = Frame-Relay HDLC-inversion = off Encap-data...

...ines and have them "received" with the incoming number intact. Having these forwarded to one of the analog numbers is a thought, but I am concerned about various issues, data corruption, etc, going that route. Thoughts vary to second T1, with channel bank, "breaking out" some DS0's into a channel bank, or finding a T1/fax board (do they exist?), to go directly into the FAX server (PC/linux based) joe a.

...11alaw ! voice class codec 1 codec preference 1 g711alaw ! voice class codec 7 codec preference 1 g711alaw ! ! ! ! ! ! ivr prompt memory 16384 files 1000 fax interface-type fax-mail mta receive maximum-recipients 0 call-history-mib max-size 500 dial-control-mib max-size 1200 ! controller E1 0 ds0-group 0 timeslots 1-15,17-31 type r2-digital ds0 busyout 15,27 hard ! controller E1 1 clock source line primary pri-group timeslots 1-31 ! controller E1 2 shutdown ! controller E1 3 shutdown pri-group timeslots 1-31 ! gw-accounting h323 gw-accounting h323 vsa gw-accounting voip ! ! inte...

...n the file. If you run puppet agent multiple times in a row each time it will cut off at always different parts. The error manifests itself like this: Debug: node supports formats: b64_zlib_yaml pson raw yaml; using pson Debug: Using cached certificate for ca Debug: Using cached certificate for ds0.lab.avere.net Debug: Using cached certificate_revocation_list for ca Info: Retrieving plugin Debug: file_metadata supports formats: b64_zlib_yaml pson raw yaml; using pson Debug: Finishing transaction 70037484276920 Info: Loading facts in /var/lib/puppet/lib/facter/serialports.rb Debug: catalog su...