search for: plar

Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown). The Oreilly (Asterisk: Future of Telephony) book mentions it in passing saying that all you need to enable it is to set immediate=yes in zapata.conf. Has anyone implemented this in brokerage trading environments? Thanks in advance. -------------- next part ------...

Greeting, I am trying to setup PLAR signalling in asterisk. I have modified the FXSLS TX bits in dahdi-base.c on line 2580, and I can make calls. .sig_type = DAHDI_SIG_FXSLS, .bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ .bits[DAHDI_TXSIG_OFFHOOK] = (0), /*changed by for PLAR*/ .bits[DAHDI_TXSIG_START] =...

...opbjjMUnyhg/ > ! > username admin password 7 07002C494908 > ! > ! > ! > ! > ip subnet-zero > ip name-server 66.178.37.211 > ! > ! > ! > ! > voice-port 1/0/0 > ! > voice-port 1/0/1 > ! > voice-port 1/1/0 > ! > voice-port 1/1/1 > connection plar 9999 > ! > ! > dial-peer voice 1000 voip > max-conn 4 > destination-pattern 9999 > req-qos guaranteed-delay > codec g711ulaw > ip precedence 5 > no vad > session target ipv4:66.178.37.169 > ! > ! > interface Ethernet0/0 > ip address 66.178.37.169 25...

...t need general direction, not details yet..i'll try to figure those...just looking to avoid brick walls...bottlenecks...inefficiencies etc upfront. Hardware: motorola vt2442 - trixbox Apps: Dot Net application that operates the Manager API and the FASTAGI interfaces. I have the 2442 set as a PLAR so as soon as the ext is off-hook, it dials into the *61 dial-plan. This is what I want to happen: Ext goes off-hook vt2442 PLAR auto-dials *61 *61 dial-plan gets some info from the inside user Need help for everything below: inside user is put on hold..queued or parked or put in a conf??? Sep...

Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

...lso use cross validation as a criterion too. I used cv.lars to give me the lowest predicted error & fraction. But I'm short of a step to arrive at the number of variables to be included in the final model. How do we do that? Is it the predict.lars function? i tried > logprostate.plars.cv=predict.lars(logprostate.lars.cv, M, type = "fit", mode="fraction") but it gives me error message: Error in dim(data) <- dim : attempt to set an attribute on NULL. Please help! thanks! _________________________________________________________________ With tax season...

say I needed to send a broadcast message that I wanted every user to hear when the pick up thier phone? can I "Play,message" on a line just before they get dialtone? or maybe after they dial before ring? how about a "ringdown" to a voicemail box and on end return them to thier line for the dialout? can * do ringdowns? when a user picks up an extension it automagically

Basically, I am looking to set up an extension which will be used as a "help-line". I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an

...b8zs pri-group timeslots 1-24 description RCN PRI at SF7 ! interface FastEthernet1/0 no ip address duplex auto speed auto ! interface Serial3/0:23 no ip address dialer-group 1 isdn switch-type primary-5ess isdn incoming-voice voice no cdp enable ! voice-port 3/0:23 connection plar 1000 ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern 1000 session protocol sipv2 session target ipv4:1.2.3.4:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2 pots destination-pattern 9T port 3/0:23 ! sip-ua retry invite...

...lex no cdp enable ! interface Ethernet0/1 no ip address no ip route-cache no ip mroute-cache shutdown half-duplex no cdp enable ! ip http server no ip http secure-server ip classless ! ! ! ! call rsvp-sync ! voice-port 1/0/0 input gain 10 output attenuation 10 no comfort-noise connection plar opx 1001 station-id number 100 caller-id enable ! voice-port 1/0/1 input gain 10 output attenuation 10 no comfort-noise caller-id enable ! voice-port 1/1/0 ! voice-port 1/1/1 ! ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice 100 pots destination-pattern .T progress_ind s...

Hi all, I have looked through the wiki for any information on how to make an extension autodial another extension when it goes off hook. Anyone done this or know how it's done. regards Dave

I have cisco phones running SCCP, and a cisco 2600 with FXO I'm using for PSTN access. I can dial out, but inbound calls are not ringing a phone. Please see my config In the 2600 I'm PLAR'ing the line and I have a SIP Dial-Peer for 4001 voice-port 1/1/0 output attenuation 0 echo-cancel coverage 32 no comfort-noise timing hookflash-out 50 connection plar opx 4001 dial-peer voice 2 voip description Route calls starting with 4 to the Asterisk PBX destination-pa...

...udder>... they're really not the right way to do it. There must be some clever way of doing this with a new channel specification that would allow bridging into an existing channel identifier. I.E.: Dial(Bridge/SIP/2203-bed5) Other related topics: - The auto-dial I can handle with PLAR ("hotline" calling - pick up the phone, and automatically a number is dialed) and DISA on the Asterisk side. In other words, when someone picks up line #1 on their Cisco 7960 (or whatever phone) I can have the system auto-dial into my * server. Using the caller ID, I can determine w...

I'm having great success using Cisco 7960s with Asterisk but one Cisco 7960 specific function that's puzzling me is programming the 'line buttons' (6 buttons on the upper right side of the phone) for speed dial. I'm not using any Cisco applications like Call Manager and the phones are SIP. *All* of the documentation I've found over the last several days continues to refer

...Let them know it's over exten => 600,4,Goto(s,6) ; Start over exten => 601,1,Dial(SIP/1@voice-gw) exten => 602,1,Dial(SIP/ben) [default] include => demo Cisco 3640 config: ! voice-port 1/0/0 input gain 10 output attenuation 11 no comfort-noise connection plar 1001 ! voice-port 1/0/1 input gain 10 output attenuation 11 no comfort-noise connection trunk 1 ! ! mgcp profile default ! ! ! dial-peer cor custom ! ! ! dial-peer voice 100 pots destination-pattern 9.... port 1/0/0 forward-digits 4 ! dial-peer voice 2 voip destination-pattern 1......

I am unable to boot Dom0 on a machine with Reiser file systems. Reiser support is built into the xen0 kernel. I have similar setup with ext3 file systems working just fine, so I don''t think it is a setup problem. I have tried booting with and without and initrd file, booting with the original Linux distro initrd file, etc, but nothing seems to work. I recall seeing a thread about ReiserFS