search for: lscpd

...tions for mapping FXO ports to the T1 DSO. DPT = Dial Pulse Termination EMDW = E&M Delayed Wink start EMI = E&M Immediate start EMICPD = E&M Immediate Start with Calling Party Disconnect EMW = E&M Wink start GS = Ground Start GSRB = Ground Start with Reverse Battery LS = Loop Start LSCPD = Loop Start Calling Party Disconnect LSRB = Loop Start with Reverse Battery VoIP = Voice over IP (CMG only) I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. I don't know if this has any relevance but I am also originat...

...p set a:1:1-24 type voice set a:1:1-24 signal ls set a:2 down set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id "CAC DS1# A:2" set a:2 linecode b8zs set a:2 loopdetect csu set a:2:1-24 side drop set a:2:1-24 type voice set a:2:1-24 signal ls -Setting slot 1. set 1:1-8 signal lscpd set 1:1-8 txgain -3 set 1:1-8 rxgain -6 -Setting slot 2. set 2:1 signal lscpd set 2:1 txgain -3 set 2:1 rxgain -6 set 2:2-8 signal ls set 2:2-8 txgain -3 set 2:2-8 rxgain -6 -Setting slot 3. set 3:1-8 signal ls set 3:1-8 txgain -3 set 3:1-8 rxgain -6 set 3:1-8 linelength short -Setting slot 4....

Hey everybody, I've had issues with detecting call disconnects on the Centrex system and an Adit 600. I've discovered that if I set the channel bank's FXO to lscpd, even though the CO says they don't support, that Asterisk can see the hangups. The catch? Asterisk can no longer seize the line. I've used this to my advantage, since I have 13 lines to work with. (5) +1 for auto-attendant are setup for incoming only (lscpd), the rest for outgoing(l...

Hello All, Just a question, I have an adit600 and I am looking for a way to pull the incoming cid into asterisk. Does anyone know if this is just not possible via t1? Or is it only available on PRI? Thanks, Greg

We have an Adit 600 with some fxo lines. Callers calling in on the fxo ports do not hear echo, but the party answering hears bad echo. Here is the gain settings on the Adit: Corporate Adit 600> show 1 SLOT 1: Settings for FXO: Channel 1: Signaling: LSCPD RxGain: -2 dB TxGain: -3 dB And here are the zap settings: busydetect=no callprogress=no echotraining=800 rxgain=0.0 txgain=-6.0 I have tried changing the txgain back and forth, but it is confusing to me which gain I need to change. Thanks for any tips. P.S. I...

...op set a:1:1-24 type voice set a:1:1-24 signal ls set a:2 down set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id "CAC DS1# A:2" set a:2 linecode b8zs set a:2 loopdetect on set a:2:1-24 side drop set a:2:1-24 type voice set a:2:1-24 signal ls -Setting slot 1. set 1:1-8 signal lscpd set 1:1-8 txgain -3 set 1:1-8 rxgain -3 -Setting slot 2. set 2:1-8 signal ls set 2:1-8 txgain -3 set 2:1-8 rxgain -6 -Setting slot 3. set 3:1-8 signal ls set 3:1-8 txgain -3 set 3:1-8 rxgain -6 set 3:1-8 linelength short -Setting slot 4. set 4:1-8 signal ls set 4:1-8 txgain -3 set 4:1-8 rxgai...

Hi, No one seems to have any issue with the following posting. Can any one suggest how to install/configure channel drivers to work. Basically I am trying to send the SIP calls to GNUGK but Asterisk reports the error "No channel driver found". >>> I was trying to compile the oh323 channel driver but unable to compile the openh323_1_13_5 (which is the only required version as