search for: mvicker

...t the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by 'pbx_config' ] '0001' => -1. Dial() [pbx_config] 1. Dial(SIP/mvickers) [pbx_config] '0002' => -1. Dial() [pbx_config] 1. Dial(SIP/jgb) [pbx_config] '0003' => -1. Dial() [pbx_config] 1. Dial(SIP/markw) [pbx_config] '0004' => -1. Dial() [pbx_config]...

...: 1 on Zap/19-1 Oct 6 10:59:56 DEBUG[28688]: File chan_zap.c, Line 1056 (zt_enable_ec): No echocancellation requested Oct 6 10:59:56 DEBUG[28688]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 6 10:59:56 DEBUG[28688]: File chan_sip.c, Line 857 (sip_call): Outgoing Call for mvickers Oct 6 10:59:56 DEBUG[5126]: File chan_sip.c, Line 568 (__sip_semi_ack): (Provisional) Stopping retransmission (but retaining packet) on '5ecb985524e3ec232cf3e54d59674900@172.20.1.67' Request 102: Found Oct 6 10:59:57 DEBUG[5126]: File chan_sip.c, Line 568 (__sip_semi_ack): (Provisional)...

What I want to do is have one phone number for multiple call bridges (meetme) so that first users are prompted for their call bridge ID then their password. exten => 7001,1,Playback(/var/lib/asterisk/sounds/if-you-know-the-extension-you want-to-dial-press-that-extension:gsm) exten => 7001,2,set $foo to whatever was entered on the phones keypad exten => 7001,3,Dial($foo,60) Thanks!

Thanks in advance root@asterisk01:~# modprobe wcfxo /lib/modules/2.4.20/misc/wcfxo.o: init_module: No such device /lib/modules/2.4.20/misc/wcfxo.o: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/misc/wcfxo.o: insmod /lib/modules/2.4.20/misc/wcfxo.o

Was there any resolution on this I also have a TE410P in an box with an Intel E7501 chipset? -Vickers -------- Original Message -------- Subject: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server Date: Tue, 8 Feb 2005 11:13:24 +1030 From: Peter Childs <peter.childs@nec.com.au> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

I have about a dozen SIP phones up and working, now I want to connect the asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers conencted to the Fujitsu PBX that I built with mgetty/pppd and have the lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M wink start, so I have confidence in the guys who set up the PBX. I've built a loop back plug

Greetings, According to the FAQ "When you load the module and have no circuit/channel bank the LED's should flash red" I get the "knight rider" lights before the module loads, but after the modules are loaded I don't get any lights, other equipment plugged in or not. The card has been on my shelf a while so I wonder if it needs a firmware upgrade ? Can the cards

I'm trying to do the winpopup emulation thing to an IP address instead of NetBIOS name. The following don't seem to work: smbclient -I 172.16.0.43 -M smbclient -M -I 172.16.0.43 The bottom one tries to interpret -I as the host name. What's up?... and don't say my head!!!! ;-) Thanks in advance, sorry if it's been asked before -Vickers