Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017
hi brian, the wildcard s400p from digium will be available with 4 FXS ports. regards kapejod Am Don, 2003-03-20 um 22.19 schrieb Brian J. Schrock:> Has anyone gotten the voicetronix boards to work with Asterisk, what > would it take? Or does anyone know where I can get 4 ports or more fxs > PCI cards that do work with asterisk? > > Brian J. Schrock > Network Engineer, RHCE, CCNA > Anistone Technologies > Phone: 614-798-9106 > FAX: 614-573-7165 > 6926 Avery Rd. > Dublin, OH 43017 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Maybe I got that backwards, I want a pci card that will accept 4 or more analog telephone handsets or stations. On Thursday, March 20, 2003, at 04:41 PM, Klaus-Peter Junghanns wrote:> hi brian, > > the wildcard s400p from digium will be available with 4 FXS ports. > > regards > kapejod > > Am Don, 2003-03-20 um 22.19 schrieb Brian J. Schrock: >> Has anyone gotten the voicetronix boards to work with Asterisk, what >> would it take? Or does anyone know where I can get 4 ports or more fxs >> PCI cards that do work with asterisk? >> >> Brian J. Schrock >> Network Engineer, RHCE, CCNA >> Anistone Technologies >> Phone: 614-798-9106 >> FAX: 614-573-7165 >> 6926 Avery Rd. >> Dublin, OH 43017 >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > >Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017
thats what a 4 port fxs card is :-) Hopefully soon Digium will release their card as I personally have a 4 projects on hold waiting for them !! On Thu, 20 Mar 2003 17:06:33 -0500, Brian J. Schrock wrote:>Maybe I got that backwards, I want a pci card that will accept 4 or >more analog telephone handsets or stations. > >On Thursday, March 20, 2003, at 04:41 PM, Klaus-Peter Junghanns wrote: > >> hi brian, >> >> the wildcard s400p from digium will be available with 4 FXS ports. >> >> regards >> kapejod >> >> Am Don, 2003-03-20 um 22.19 schrieb Brian J. Schrock: >>> Has anyone gotten the voicetronix boards to work with Asterisk, what >>> would it take? Or does anyone know where I can get 4 ports or more fxs >>> PCI cards that do work with asterisk? >>> >>> Brian J. Schrock >>> Network Engineer, RHCE, CCNA >>> Anistone Technologies >>> Phone: 614-798-9106 >>> FAX: 614-573-7165 >>> 6926 Avery Rd. >>> Dublin, OH 43017 >>> >>> _______________________________________________ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >Brian J. Schrock >Network Engineer, RHCE, CCNA >Anistone Technologies >Phone: 614-798-9106 >FAX: 614-573-7165 >6926 Avery Rd. >Dublin, OH 43017 > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users.
anyone out there got * to work with a voicetronix openline4 card ??? Regards Mick
Sound is OK on inside phone When calling in sounds bad Regards Mick -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of andrewg@felinemenace.org Sent: Saturday, 11 October 2003 10:39 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] voicetronix On Sat, Oct 11, 2003 at 10:25:37PM +0930, mick@netexpress.com.au wrote:> Andrew > > I am having trouble with > > Sound ( only if you dialling from outside )Hmm. not really. uhm. could play with the volume levels. or it could possibly be something else.> > Cisco phone can not dial outSounds more like an asterisk configuration problem. IIRC, I had Dial(VPB/1/1) to dialout using port 1 of board 1. You might want to double check your config file. The VPB4 stuff uses port 1 - 4.> > If I phone in and select extension number of Cisco phone > > * dies > > Any ideas ???Well. asterisk -vvvvvgc and mebe start it under gdb, and when it crashes, do backtrace and then type x/3i $eip info registers that should be more useful for the developer people, though it may remind me of something. To start asterisk under gdb, type gdb /path/to/asterisk then once gdb has started, type set args -vvvvvgc then type run if/when it crashes, use the above commands and record the output and post to the group. I find the type typescript program useful for that. just type typescript hit enter, and type exit after you exit gdb. Hmm. this email is a bit arse about, but meh. More of ideas flowing than actual concurrent thoughts.> > Regards Mick > >Hope this helps, Andrew Griffiths _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
When I ring in and dial the extension to my Cisco 7940 * dies and I get this message asterisk: vpbdial.cpp:895: void vpbdial_playtone (int, VPB_TONE *): Assertion `t onestate[chdev] == 0' failed. Aborted This only happens if in Extensions.conf I place a r in the extion line eg. exten => 1004,2,Dial(sip/mick@192.168.0.216,20,r) If I leave the " r " out the phone rings once and then hangs up. Regards Mick
Hi
I have this situation at a client: Asterisk installed on Linux, Voicetronix
Openswitch 12 card installed. Testing with ringstat (as told by voicetronix
support) makes all connected handsets ring (as should be the case). When
calling Asterisk says it's ringing (I'm calling to the PSTN, in this
case my
cellular phone), my SIP phone (X-Lite) is ringing, but my cellular phone is
not ringing. Please help, what is the problem? Do I need to alter something
in chan_vpb.c? Could someone send config file examples that work? Does
anyone know how to help me??
Some one said that there was a problem with Voicetronix that it did not wait
until having a dialtone before dialing, that you should put a comma
"," in
the dialstring, I've tried this, but it changes nothing. Below is the output
from Asterisk, which clearly says that it's ringing, and further below my
vpb.conf and extensions.conf
Read_channel ## vpb/1-9: Setting record mode, bridge = 0
-- 1-9 requested, got: [vpb/1-9]
-- Calling 1-9/,3487446196 on vpb/1-9
-- VPB Calling 1-9/,3487446196 [t=0] on vpb/1-9 returned 0
-- Called 1-9/,3487446196
-- vpb/1-9 is ringing
-- hangup on vpb (vpb/1-9)
-- Hungup on vpb/1-9 complete
== Spawn extension (default, 3487446196, 1) exited non-zero on
'SIP/110-db94'
comment: The hangup is after I hangup the SIP phone
In my extensions.conf:
VPB=,
exten => _XXXXXXXXXX,1,Dial(vpb/1-9/${VPB}3487446196)
My vpb.conf:
[interfaces]
echocancel = on
board = 1
context = default
mode = fxo
channel = 9
channel = 10
channel = 11
channel = 12
mode dialtone
channel = 1
channel = 2
channel = 3
channel = 4
channel = 5
channel = 6
channel = 7
channel = 8
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Hi. With voicetronix Openswitch12, I have installed the latest drivers, Asterisk 1.0-RC1 and so far so good. I want the simplest of all cases to begin, all 12 channels FXS. But, it doesn't work. When I pick up the phone, and dial "1", which in my extensions.conf should make my sip phone ring, asterisk doesn't register that I've pushed the "1" on the analoge phone. But Asterisk has registered that I've picked up, and is sending the dialtone. If I try to ring to it from my sip phone, the analouge phone rings for an instance then a hangup is done. If I stop Asterisk, rmmod vpbhp, and insmod vpbhp, and try again, sometimes it works, and I can ring, but if I make the call from analouge to sip, then I can hear nothing in the sip phone, in the analouge perfect. If I call from the sip phone, bouth parties hear perfect. Even worse if I make a call from analouge to analouge, I hear perfect in bouth phones. But when I hang up, it is not registered, and there is a bridged call left in Asterisk and only way to get rid of it is to close Asterisk. My conf files are below: extensions.conf [vpb-fxs] exten => s,1,Wait,4 exten => s,2,Answer exten => s,3,Hangup ; call to sip, dial 1 exten => 1,1,Wait,2 exten => 1,2,Dial(SIP/116,30,t) ; to make call analouge to analouge (line 3) dial 2 exten => 2,1,Wait,2 exten => 2,2,Dial(vpb/1-3/,30,t) [from-sip] exten => _41,1,Dial(vpb/1-1/,30,t) exten => _42,1,Dial(vpb/1-2/,30,t) exten => _43,1,Dial(vpb/1-3/,30,t) vpb.conf [general] cards = 1 type = v12pci [interfaces] board = 1 context = vpb-fxs mode = dialtone channel = 1 channel = 2 channel = 3 channel = 4 channel = 5 channel = 6 channel = 7 channel = 8 channel = 9 channel = 10 channel = 11 channel = 12
hi, I need help on how to setup voicetronix openswitch 12 card with asterisk ! Can someone help me with tahat? And how i configure a cisco 7940, phone skinny protocol to work with saterisk too? or may possible to convert it to sip? Thanks in advance Prof. Marcelo Kruk -- Prof. Marcelo Kruk - System Manager & Webmaster - Voip Consultant Colegio Nacional Jose Pedro Varela - Colonia 1637 CP 11200 Phone: +598 2 4097020 Fax: +598 2 4093219 Data: +598 2 4095977* Montevideo Uruguay South America URL: http://www.reu.edu.uy Internet Society Member 1336660 ICANN Member - 218338 Linux User 18471
Anybody using voicetronix cards? The 12 ports for example? What has been your experience and how many cards can be put into one server? Do they have the same IRQ problems as Digium ones? AK