Re: [asterisk-users] Call Pickup Works w/Linksys ATA, not with Cisco 7940G

2009-04-08 Thread Vincent Li
/pipermail/asterisk-users/2004-April/036869.html Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Asterisk is not designed for University with largeuser base?

2009-03-17 Thread Vincent Li
On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too

[asterisk-users] Asterisk is not designed for University with large user base?

2009-03-16 Thread Vincent Li
manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running in my office for small user base, I don't have experience with large scale Asterisk implementation. I know little about sipX. Does anyone in the community has any input about this? Vincent Li

Re: [asterisk-users] oslec + dahdi

2009-01-22 Thread Vincent Li
-m += echo.o' /usr/src/dahdi/drivers/staging/echo/Kbuild cd /usr/src/dahdi make make install cd /usr/src tar zxvf dahdi-tools-2.1.0.2.tar.gz cd /usr/src/dahdi-tools-2.1.0.2 ./configure make make install Hope it helps. Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t

[asterisk-users] Hearing transfer during call

2008-04-04 Thread Vincent Li
Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word transfer, I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could

[asterisk-users] Hearing transfer during call

2008-04-03 Thread Vincent Li
Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word transfer, I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could

[asterisk-users] Asterisk to make multiple extensions simultaneous calls on a single telephone line

2007-12-14 Thread Vincent Li
Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each

[asterisk-users] A simple IVR extension problem

2007-08-02 Thread Vincent Li
Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf context=incoming signalling=fxs_ks channel = 4 context=internal signalling=fxo_ks channel = 1 -