Network Cabling/VoIP/Voice/Data Solution. CommITtel can save you up to 40 percent of you telephone bill

[CommIT.tel]

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Re: Magic Jack VOIP telephone

[Alejandro Imass]

On Thu, May 6, 2010 at 11:32 AM,   wrote:
> in message
> ,
> wrote Alejandro Imass thusly...
>>
> ...>
> Alejandro, how time & labour intensive is to set up Asterisk on a
> daily use laptop?  Could Asterisk not be used by itself for all VoIP

Ok, if you don't need dialplans, voicemails, and several extensions,
you don't need asterisk. Just install Skype client on FBSD (you will
need linux emulation to get the Skype client going), and purchase an
actual phone number with your area code from Skype, and some pre-paid
credit. That's it you will be receiving/making call from skype on
FBSD in no time.

Regardiong Asterisk it's actually very easy and without any prior
knowledge I was able in just about 4 days to make a complete
soft-phone based telephone system with 10 ekiga extensions, automated
menu (ivr) and the works (voice mail that is routed via e-mail) - the
works - just by installing from port and reading the free asterisk
book pdf you can download from many places (yes, the book is actually
free as in beer).

> needs per my superficial Asterisk understanding?
>

Just read the book.

> Was setting up Asterisk as simple as installing
> /usr/ports/net/asterisk-bristuff or /usr/ports/net/asterisk1{2,6}?
>

Yeap!
Alex

>
>  - parv
>
> --
>
>
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Re: Magic Jack VOIP telephone

[parv]

in message
,
wrote Alejandro Imass thusly...
>
...
> I [...] just stuck with Skype who now BTW offers international SIP
> services, so I also hooked up Skype + Asterisk. This means you can
> purchase Skype phone # and attach it to your Asterisk and
> everything in FBSD. No Windoze crappy software.

Alejandro, how time & labour intensive is to set up Asterisk on a
daily use laptop?  Could Asterisk not be used by itself for all VoIP
needs per my superficial Asterisk understanding?

Was setting up Asterisk as simple as installing
/usr/ports/net/asterisk-bristuff or /usr/ports/net/asterisk1{2,6}?


  - parv

-- 

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Re: Magic Jack VOIP telephone

[Outback Dingo]

not sure id waste my time on Magic Jack, let alone working with FreeBSD

On Wed, May 5, 2010 at 3:16 PM, Lee Shackelford
wrote:

>
> Good afternoon, FreeBSD enthusiasts.  Have any of you attempted to use
> Magic Jack VOIP telephone with an Intel workstation equipped with FreeBSD
> operating system?  It is supposed to work with Mac OSX, which I understand
> to be similar to FreeBSD, but I do not know if it is sufficiently similar.
> Would appreciate a response from anyone who has attempted such a use,
> whether successful, or unsuccessful.  Thank you.  L e e _ S h a c k e l f o
> r d @ d o t . c a . g o v
>
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Re: Magic Jack VOIP telephone

[Alejandro Imass]

On Wed, May 5, 2010 at 3:16 PM, Lee Shackelford
 wrote:
>
> Good afternoon, FreeBSD enthusiasts.  Have any of you attempted to use
> Magic Jack VOIP telephone with an Intel workstation equipped with FreeBSD

Hi there, and sorry to be so blunt.

MagicJack is a piece of crap. The dongle is just an a/d conerter to
plug-in a regular phone, the rest is pure software. I had to throow 2
away because of lack of Linux support and just stuck with Skype who
now BTW offers international SIP services, so I also hooked up Skype +
Asterisk. This means you can purchase Skype phone # and attach it to
your Asterisk and everything in FBSD. No Windoze crappy software.

> operating system?  It is supposed to work with Mac OSX, which I understand
> to be similar to FreeBSD, but I do not know if it is sufficiently similar.
> Would appreciate a response from anyone who has attempted such a use,
> whether successful, or unsuccessful.  Thank you.  L e e _ S h a c k e l f o
> r d @ d o t . c a . g o v
>
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Magic Jack VOIP telephone

[Lee Shackelford]

Good afternoon, FreeBSD enthusiasts.  Have any of you attempted to use
Magic Jack VOIP telephone with an Intel workstation equipped with FreeBSD
operating system?  It is supposed to work with Mac OSX, which I understand
to be similar to FreeBSD, but I do not know if it is sufficiently similar.
Would appreciate a response from anyone who has attempted such a use,
whether successful, or unsuccessful.  Thank you.  L e e _ S h a c k e l f o
r d @ d o t . c a . g o v

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Re: Zero size allocation with Yealink VOIP USB Phone, 7.2-RELEASE (Was: Re: hald: kmem_malloc error)

[Mel Flynn]

On Saturday 18 July 2009 12:20:17 Andrey Shuvikov wrote:

> How did you know it's Yealink? Just because
> it's the only uhid device?

Yes, that's why the dmesg was useful.

> Thanks a lot!

You're very welcome.

-- 
Mel
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Re: Zero size allocation with Yealink VOIP USB Phone, 7.2-RELEASE (Was: Re: hald: kmem_malloc error)

[Andrey Shuvikov]

On Sat, Jul 18, 2009 at 2:29 PM, Mel
Flynn wrote:
> [ Adding usb@ and keeping long context for that purpose ]
>
> On Saturday 18 July 2009 08:29:32 Andrey Shuvikov wrote:
>> On Sat, Jul 18, 2009 at 2:02 AM, Mel
>>
>> Flynn wrote:
>> > On Friday 17 July 2009 18:59:49 Andrey Shuvikov wrote:
>> >> I'm trying to configure X and according to the manual enabled DBUS and
>> >> HALD. But when hald is starting up I get kernel panic:
>> >>
>> >> kmem_malloc: entry not found or misaligned
>> >>
>> >> Does anyone know what could be wrong? I have memory dump if it can
>> >> help but it's big (173M).
>> >
>> > If you have a file /var/crash/vmcore.0, you will want to run the
>> > following command:
>> > kgdb /boot/kernel/kernel /var/crash/vmcore.0
>> > Then type bt at the prompt and paste output here.
>> >
>> > More info:
>> > > >neldebug.html>
>> >
>> > uname -a and dmesg output also help in diagnosing this problem.
>> > --
>> > Mel
>>
>> The uname output is:
>>
>> FreeBSD foxtrot.home 7.2-RELEASE FreeBSD 7.2-RELEASE #0: Fri May  1
>> 08:49:13 UTC 2009
>> r...@walker.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC  i386
>>
>> The kgdb output:
>>
>> GNU gdb 6.1.1 [FreeBSD]
>> Copyright 2004 Free Software Foundation, Inc.
>> GDB is free software, covered by the GNU General Public License, and you
>> are welcome to change it and/or distribute copies of it under certain
>> conditions. Type "show copying" to see the conditions.
>> There is absolutely no warranty for GDB.  Type "show warranty" for details.
>> This GDB was configured as "i386-marcel-freebsd"...
>>
>> Unread portion of the kernel message buffer:
>> panic: kmem_malloc: entry not found or misaligned
>> cpuid = 0
>> Uptime: 52s
>> Physical memory: 2034 MB
>> Dumping 176 MB: 161 145 129 113 97 81 65 49 33 17 1
>>
>> Reading symbols from /boot/kernel/linux.ko...Reading symbols from
>> /boot/kernel/linux.ko.symbols...done.
>> done.
>> Loaded symbols for /boot/kernel/linux.ko
>> #0  doadump () at pcpu.h:196
>> 196   pcpu.h: No such file or directory.
>>       in pcpu.h
>> (kgdb) bt
>> #0  doadump () at pcpu.h:196
>> #1  0xc07e25a7 in boot (howto=260) at /usr/src/sys/kern/kern_shutdown.c:418
>> #2  0xc07e2879 in panic (fmt=Variable "fmt" is not available.
>> ) at /usr/src/sys/kern/kern_shutdown.c:574
>> #3  0xc0a1cdc0 in kmem_malloc (map=0xc147108c, size=0, flags=2)
>>     at /usr/src/sys/vm/vm_kern.c:381
>> #4  0xc0a13357 in page_alloc (zone=0x0, bytes=0, pflag=0xe7b6497f "\002",
>>     wait=2) at /usr/src/sys/vm/uma_core.c:952
>> #5  0xc0a15e20 in uma_large_malloc (size=0, wait=2)
>>     at /usr/src/sys/vm/uma_core.c:2706
>> #6  0xc07d16f8 in malloc (size=0, mtp=0xc0c46580, flags=2)
>                            ^^
>>     at /usr/src/sys/kern/kern_malloc.c:393
>> #7  0xc0743044 in uhidopen (dev=0xc5713000, flag=1, mode=8192,
>> p=0xc5c6a460) at /usr/src/sys/dev/usb/uhid.c:428
> In kgdb can you print the entire softcell as follows:
> f 7
> p *sc
>
> Hopefully that will provide sufficient information for the usb developers to
> fix this problem.
>
>> #8  0xc07a56a0 in giant_open (dev=0xc5713000, oflags=1, devtype=8192,
>>     td=0xc5c6a460) at /usr/src/sys/kern/kern_conf.c:332
>> #9  0xc076e1fc in devfs_open (ap=0xe7b64a88)
>>     at /usr/src/sys/fs/devfs/devfs_vnops.c:908
>> #10 0xc0af88d2 in VOP_OPEN_APV (vop=0xc0c47ee0, a=0xe7b64a88)
>>     at vnode_if.c:371
>> #11 0xc0870829 in vn_open_cred (ndp=0xe7b64b7c, flagp=0xe7b64c78, cmode=0,
>>     cred=0xc5470100, fp=0xc5b57da8) at vnode_if.h:199
>> #12 0xc0870973 in vn_open (ndp=0xe7b64b7c, flagp=0xe7b64c78, cmode=0,
>>     fp=0xc5b57da8) at /usr/src/sys/kern/vfs_vnops.c:94
>> #13 0xc086e0a3 in kern_open (td=0xc5c6a460,
>>     path=0xbfbfe90c ,
>>     pathseg=UIO_USERSPACE, flags=1, mode=0)
>>     at /usr/src/sys/kern/vfs_syscalls.c:1042
>> #14 0xc086e610 in open (td=0xc5c6a460, uap=0xe7b64cfc)
>>     at /usr/src/sys/kern/vfs_syscalls.c:1009
>> #15 0xc0ae4495 in syscall (frame=0xe7b64d38)
>>     at /usr/src/sys/i386/i386/trap.c:1090
>> #16 0xc0ac9260 in Xint0x80_syscall ()
>>     at /usr/src/sys/i386/i386/exception.s:255
>> #17 0x0033 in ?? ()
>> Previous frame inner to this frame (corrupt stack?)
>> (kgdb) q
>>
>> The dmesg:
>>
>> Copyright (c) 1992-2009 The FreeBSD Project.
>> Copyright (c) 1979, 1980, 1983, 1986, 1988, 1989, 1991, 1992, 1993, 1994
>>       The Regents of the University of California. All rights reserved.
>> FreeBSD is a registered trademark of The FreeBSD Foundation.
>> FreeBSD 7.2-RELEASE #0: Fri May  1 08:49:13 UTC 2009
>>     r...@walker.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC
>> Timecounter "i8254" frequency 1193182 Hz quality 0
>> CPU: Intel(R) Core(TM)2 Duo CPU     E8400  @ 3.00GHz (2999.67-MHz 686-class
>> CPU) Origin = "GenuineIntel"  Id = 0x10676  Stepping = 6
>>
>> Features=0xbfebfbff>A,CMOV,PAT,PSE36,CLFLUSH,DTS,ACPI,MMX,FXSR,SSE,SSE2,SS,HTT,TM,PBE>
>> Features2=0x8e3fd>CM,SSE4.1> AMD Features=0x2010
>> 

Zero size allocation with Yealink VOIP USB Phone, 7.2-RELEASE (Was: Re: hald: kmem_malloc error)

[Mel Flynn]

[ Adding usb@ and keeping long context for that purpose ]

On Saturday 18 July 2009 08:29:32 Andrey Shuvikov wrote:
> On Sat, Jul 18, 2009 at 2:02 AM, Mel
>
> Flynn wrote:
> > On Friday 17 July 2009 18:59:49 Andrey Shuvikov wrote:
> >> I'm trying to configure X and according to the manual enabled DBUS and
> >> HALD. But when hald is starting up I get kernel panic:
> >>
> >> kmem_malloc: entry not found or misaligned
> >>
> >> Does anyone know what could be wrong? I have memory dump if it can
> >> help but it's big (173M).
> >
> > If you have a file /var/crash/vmcore.0, you will want to run the
> > following command:
> > kgdb /boot/kernel/kernel /var/crash/vmcore.0
> > Then type bt at the prompt and paste output here.
> >
> > More info:
> >  >neldebug.html>
> >
> > uname -a and dmesg output also help in diagnosing this problem.
> > --
> > Mel
>
> The uname output is:
>
> FreeBSD foxtrot.home 7.2-RELEASE FreeBSD 7.2-RELEASE #0: Fri May  1
> 08:49:13 UTC 2009
> r...@walker.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC  i386
>
> The kgdb output:
>
> GNU gdb 6.1.1 [FreeBSD]
> Copyright 2004 Free Software Foundation, Inc.
> GDB is free software, covered by the GNU General Public License, and you
> are welcome to change it and/or distribute copies of it under certain
> conditions. Type "show copying" to see the conditions.
> There is absolutely no warranty for GDB.  Type "show warranty" for details.
> This GDB was configured as "i386-marcel-freebsd"...
>
> Unread portion of the kernel message buffer:
> panic: kmem_malloc: entry not found or misaligned
> cpuid = 0
> Uptime: 52s
> Physical memory: 2034 MB
> Dumping 176 MB: 161 145 129 113 97 81 65 49 33 17 1
>
> Reading symbols from /boot/kernel/linux.ko...Reading symbols from
> /boot/kernel/linux.ko.symbols...done.
> done.
> Loaded symbols for /boot/kernel/linux.ko
> #0  doadump () at pcpu.h:196
> 196   pcpu.h: No such file or directory.
>   in pcpu.h
> (kgdb) bt
> #0  doadump () at pcpu.h:196
> #1  0xc07e25a7 in boot (howto=260) at /usr/src/sys/kern/kern_shutdown.c:418
> #2  0xc07e2879 in panic (fmt=Variable "fmt" is not available.
> ) at /usr/src/sys/kern/kern_shutdown.c:574
> #3  0xc0a1cdc0 in kmem_malloc (map=0xc147108c, size=0, flags=2)
> at /usr/src/sys/vm/vm_kern.c:381
> #4  0xc0a13357 in page_alloc (zone=0x0, bytes=0, pflag=0xe7b6497f "\002",
> wait=2) at /usr/src/sys/vm/uma_core.c:952
> #5  0xc0a15e20 in uma_large_malloc (size=0, wait=2)
> at /usr/src/sys/vm/uma_core.c:2706
> #6  0xc07d16f8 in malloc (size=0, mtp=0xc0c46580, flags=2)
^^
> at /usr/src/sys/kern/kern_malloc.c:393
> #7  0xc0743044 in uhidopen (dev=0xc5713000, flag=1, mode=8192,
> p=0xc5c6a460) at /usr/src/sys/dev/usb/uhid.c:428
In kgdb can you print the entire softcell as follows:
f 7
p *sc

Hopefully that will provide sufficient information for the usb developers to 
fix this problem.

> #8  0xc07a56a0 in giant_open (dev=0xc5713000, oflags=1, devtype=8192,
> td=0xc5c6a460) at /usr/src/sys/kern/kern_conf.c:332
> #9  0xc076e1fc in devfs_open (ap=0xe7b64a88)
> at /usr/src/sys/fs/devfs/devfs_vnops.c:908
> #10 0xc0af88d2 in VOP_OPEN_APV (vop=0xc0c47ee0, a=0xe7b64a88)
> at vnode_if.c:371
> #11 0xc0870829 in vn_open_cred (ndp=0xe7b64b7c, flagp=0xe7b64c78, cmode=0,
> cred=0xc5470100, fp=0xc5b57da8) at vnode_if.h:199
> #12 0xc0870973 in vn_open (ndp=0xe7b64b7c, flagp=0xe7b64c78, cmode=0,
> fp=0xc5b57da8) at /usr/src/sys/kern/vfs_vnops.c:94
> #13 0xc086e0a3 in kern_open (td=0xc5c6a460,
> path=0xbfbfe90c ,
> pathseg=UIO_USERSPACE, flags=1, mode=0)
> at /usr/src/sys/kern/vfs_syscalls.c:1042
> #14 0xc086e610 in open (td=0xc5c6a460, uap=0xe7b64cfc)
> at /usr/src/sys/kern/vfs_syscalls.c:1009
> #15 0xc0ae4495 in syscall (frame=0xe7b64d38)
> at /usr/src/sys/i386/i386/trap.c:1090
> #16 0xc0ac9260 in Xint0x80_syscall ()
> at /usr/src/sys/i386/i386/exception.s:255
> #17 0x0033 in ?? ()
> Previous frame inner to this frame (corrupt stack?)
> (kgdb) q
>
> The dmesg:
>
> Copyright (c) 1992-2009 The FreeBSD Project.
> Copyright (c) 1979, 1980, 1983, 1986, 1988, 1989, 1991, 1992, 1993, 1994
>   The Regents of the University of California. All rights reserved.
> FreeBSD is a registered trademark of The FreeBSD Foundation.
> FreeBSD 7.2-RELEASE #0: Fri May  1 08:49:13 UTC 2009
> r...@walker.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC
> Timecounter "i8254" frequency 1193182 Hz quality 0
> CPU: Intel(R) Core(TM)2 Duo CPU E8400  @ 3.00GHz (2999.67-MHz 686-class
> CPU) Origin = "GenuineIntel"  Id = 0x10676  Stepping = 6
>  
> Features=0xbfebfbffA,CMOV,PAT,PSE36,CLFLUSH,DTS,ACPI,MMX,FXSR,SSE,SSE2,SS,HTT,TM,PBE>
> Features2=0x8e3fdCM,SSE4.1> AMD Features=0x2010
>   AMD Features2=0x1
>   Cores per package: 2
> real memory  = 2146893824 (2047 MB)
> avail memory = 2091225088 (1994 MB)
> ACPI APIC Table: 
> FreeBSD/SMP: Multiprocessor Syste

Re: FreeBSD as VOIP PBX

[Chuck Robey]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

sergio lenzi wrote:
> Em Sex, 2008-06-20 às 21:45 +0200, Wojciech Puchar escreveu:
> 
 when i connected 56 cisco phones to my laptop (used 4*16 port switches
 ;), and having all of them working (called from first to second, from 
 third to
 fourth etc..) there was below 4% CPU load but it's 1200Mhz Pentium-3M.


>>> Yes... the cisco uses SIP, that is far more efficient...
>>>
>> i forget to say - SIP allows direct calls (data goes directly between 
>> phones), SCCP doesn't (at least asterisk module).
>>
>> in tests i intentionally disabled this to make asterisk server loaded
> 
> we use sip the same way you do with sccp because we need tranfer calls
> (,Tt) in the dial command
> E1 boards, the best we tested are from the chinese openvox... 
> without echo cancelation it seels for about U$750,00 for one port E1,
> US$1800 for 2 ports, 
> US$2800 for 4 ports...

My god, for the hardware involved, that's unbelieveably expansive.

> in my country (brazil) you may think it is too expensive, but as
> you 
> think that ONE port for a siemens pabx is about US$4000  (yes, 4K
> dollars)

For the Siemens pabx, you're paying for the switching capability, and the
literal ton of software to do all of the call handling.  Maybe I got you wrong,
in what I read above, I haven't seen those Openvox cards, but if they are only
voice interface (a T1 or E1 single channel) plus signally, wow, that's a lot.
If the interface an entire group, either T1 or E1, that's better, but it sure
includes a healthy kick for a profit factor.  I know, I've built them in the
past, there's just not THAT much to them.

Maybe I'm missing something.

Actually, in the present case, the cost of doing switching has dropped in a
major way, so the cost, which used to be justifiable at $4k/channel, well, it's
certainly not that way any more.

Let's see, from memory, I think that the old Northern Telecom DMS250 ran about
2.5 million plus the cost of channel banks, I think.  I was always doing
engineering,  not sales, but your cost figures, they sure do seem high to me.

As far as handling the software, the old tandem switches used to use
mini-computers to run maybe 4,000 channels in one switch.  I forget the name of
the most famous tandem switch, but I do know they used a single mini.  Today's
computers are far more capable, and so could very easily power a whole switch.
Course, doing that kind of software, well, it's the most difficult stuff to do
that has ever been accomplished.  The folks that did it never got enough credit.


> you may imagine that for the price of only one board for a siemens you
> can mount
> the pbx, the cpu, the FreeBSD.
> 
> you mount a 100 phones pbx for less than half the price of a siemens
> equipment
> including the 50 ATAs linksys pap2.
> 
> The poor the country, the more you pay  that is the rule.
> 
> Philips, nortel, alcatel are even more expensive..
> 
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Re: FreeBSD as VOIP PBX

[sergio lenzi]

Em Sex, 2008-06-20 às 21:45 +0200, Wojciech Puchar escreveu:

> >> when i connected 56 cisco phones to my laptop (used 4*16 port switches
> >> ;), and having all of them working (called from first to second, from 
> >> third to
> >> fourth etc..) there was below 4% CPU load but it's 1200Mhz Pentium-3M.
> >>
> >>
> >
> > Yes... the cisco uses SIP, that is far more efficient...
> >
> i forget to say - SIP allows direct calls (data goes directly between 
> phones), SCCP doesn't (at least asterisk module).
> 
> in tests i intentionally disabled this to make asterisk server loaded

we use sip the same way you do with sccp because we need tranfer calls
(,Tt) in the dial command
E1 boards, the best we tested are from the chinese openvox... 
without echo cancelation it seels for about U$750,00 for one port E1,
US$1800 for 2 ports, 
US$2800 for 4 ports...
in my country (brazil) you may think it is too expensive, but as
you 
think that ONE port for a siemens pabx is about US$4000  (yes, 4K
dollars)
you may imagine that for the price of only one board for a siemens you
can mount
the pbx, the cpu, the FreeBSD.

you mount a 100 phones pbx for less than half the price of a siemens
equipment
including the 50 ATAs linksys pap2.

The poor the country, the more you pay  that is the rule.

Philips, nortel, alcatel are even more expensive..

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Re: FreeBSD as VOIP PBX

[sergio lenzi]

> when i connected 56 cisco phones to my laptop (used 4*16 port switches 
> ;), and having all of them working (called from first to second, from third 
> to 
> fourth etc..) there was below 4% CPU load but it's 1200Mhz Pentium-3M.
> 
> 

Yes... the cisco uses SIP, that is far more efficient...

the problem is the E1 boards... that all the processing of hdc and
blocking/deblocking/signalling 
is done by the main board processor... so the high irq rate

We think FreeBSD is a good choice for VOIP asterisks... BEsides put us
(the company) in a higher
level.. not because Linux is bad, but because the linux people did not
made good servers in the past,
so, we decided to link our name (the company) to UNIX (BSD) it is a
stronger name...

With the famous rock solid  of FreeBSD, our customers are satisfyed
whith the performance
of the machine and the quality of the product
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Re: FreeBSD as VOIP PBX

[Chris St Denis]

As well as the ever popular Asterisk, there is also

/usr/ports/net/sipxpbx

If all you want is SIP this will do nicely.


Thomas Mullins wrote:

Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software
are you using?  And any recommendations for software to look at would be
greatly appreciated.

 


Shane

 


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--
Chris St Denis
Programmer
SmarttNet (www.smartt.com)
Ph: 604-473-9700 Ext. 200
---
"Smart Internet Solutions For Businesses" 


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Re: FreeBSD as VOIP PBX

[Wojciech Puchar]

each board fires 2 irqs and 1000 irq/sec each...  so is 2000 irq per
board,
at a max of 6000 irq /sec

Cpu use (using top) is about 10%  only when receiving fax (because of


still quite a lot ;)

when i connected 56 cisco phones to my laptop (used 4*16 port switches 
;), and having all of them working (called from first to second, from third to 
fourth etc..) there was below 4% CPU load but it's 1200Mhz Pentium-3M.





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Re: FreeBSD as VOIP PBX

[sergio lenzi]

Em Sex, 2008-06-20 às 19:19 +0200, Wojciech Puchar escreveu:

> > How much cpu
> > we use always AMD 64 X2 with 2Gb of memory for
> > up to 3 PCI boards of 4 E1 each, that is: 12 E1.
> 
> what codec do you use? if no recoding (alaw) i think it uses very 
> little CPU in order of few% of single or less.

We use ulaw (for the logarithm mode of the volume) so no recoding is done...


each board fires 2 irqs and 1000 irq/sec each...  so is 2000 irq per
board,
at a max of 6000 irq /sec

Cpu use (using top) is about 10%  only when receiving fax (because of
the spandsp), goes to 20%... when recieving 4 simultaneous fax

We use Hylafax  + iaxmodem...  works like a charm... the same way
24/7... 
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Re: FreeBSD as VOIP PBX

[Wojciech Puchar]

greatly appreciated.


I bought the book "Asterisk: The Future of Telephony" a little while ago,
and it was an *excellent* introduction.  I used it to configure Asterisk on


i must say it was *excellent* for me too.
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Re: FreeBSD as VOIP PBX

[Wojciech Puchar]

How much cpu
we use always AMD 64 X2 with 2Gb of memory for
up to 3 PCI boards of 4 E1 each, that is: 12 E1.


what codec do you use? if no recoding (alaw) i think it uses very 
little CPU in order of few% of single or less.

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Re: FreeBSD as VOIP PBX

[Kirk Strauser]

On Friday 20 June 2008, Thomas Mullins wrote:
> Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software
> are you using?  And any recommendations for software to look at would be
> greatly appreciated.

I bought the book "Asterisk: The Future of Telephony" a little while ago, 
and it was an *excellent* introduction.  I used it to configure Asterisk on 
my FreeBSD server.

link: http://oreilly.com/catalog/9780596009625/
-- 
Kirk Strauser


signature.asc
Description: This is a digitally signed message part.

Re: FreeBSD as VOIP PBX

[sergio lenzi]

Em Sex, 2008-06-20 às 14:55 +0200, Wojciech Puchar escreveu:

> > Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software

I work on a copany that does this

PABX (large ones) only using FreeBSD -> http://www.levier.com.br

FreeBSD 7.0 asterisk 1.4 openvox hardware
it is incredible stable...  and using postgres as a database + dial
controller...
(the dial logic is done in sql stored procedures) the possibilities are
far beyond
the needs of the clients

We have several call centers using it... in a 24/7 basis... 
Using hardware from openvox (E1/ISDN) 

How much cpu
we use always AMD 64 X2 with 2Gb of memory for
up to 3 PCI boards of 4 E1 each, that is: 12 E1.
Each board trigges 2000 interrups/second... so
there are 6000 irqs/sec on the system
for an universe of 200 ATA's  (cisco PAP2...)  and 400 users...
on line, fulltime... 

On the bench mark, FreeBSD was superior performance over
other OS... including the new Linux kernels.. 

The asterisk  is 1.4.20 build from the ports with codec negotiation
disable

Hope you have an idea now


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Re: FreeBSD as VOIP PBX

[Jerry B. Altzman]

On Fri, Jun 20, 2008 at 7:49 AM, Thomas Mullins
<[EMAIL PROTECTED]> wrote:
> Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software
> are you using?  And any recommendations for software to look at would be
> greatly appreciated.

[EMAIL PROTECTED] ~]$ uname -a
FreeBSD iridium.xxx.com 6.1-RELEASE FreeBSD 6.1-RELEASE #0: Sun May  7
04:32:43 UTC 2006
[EMAIL PROTECTED]:/usr/obj/usr/src/sys/GENERIC  i386
[EMAIL PROTECTED] ~]$ ps auxww| grep asterisk|grep -v grep
root 439  0.0  8.4 39396 32312  ??  Ss4Jun08  63:35.57
/usr/local/sbin/asterisk
[EMAIL PROTECTED] ~]$ pkg_info |grep asterisk
asterisk-1.2.9.1_1  An Open Source PBX and telephony toolkit

Works reasonably well; I haven't updated to 1.4 yet because this is a
production server for my company. We have a digium TDM13 in there (3
inbound POTS, 1 outbound phone) + several VoIP providers (IAX) + a
hand-hacked callplan.

> Shane

//jbaltz
-- 
jerry b. altzman [EMAIL PROTECTED] www.jbaltz.com
foo mane padme hum
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Re: FreeBSD as VOIP PBX

[Matthias Fechner]

Hi Thomas,

Thomas Mullins wrote:

Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software
are you using?  And any recommendations for software to look at would be
greatly appreciated.


I have here asterisk (from the port) running with 2 ISDN cards (HFC) 
using ISDN4BSD with capi.

The computer makes the connections from
external ISDN, SIP, IAX2 <-> internal ISDN, SIP, IAX2.

Works for more then 2 years absolutly stable.

Best regards,
Matthias

--
"Programming today is a race between software engineers striving to 
build bigger and better idiot-proof programs, and the universe trying to 
produce bigger and better idiots. So far, the universe is winning." -- 
Rich Cook


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RE: FreeBSD as VOIP PBX

[Wojciech Puchar]

if you want to suffer alot of pain on an incomplete system you could go


people often pay for using such shortcuts. but that case is 
special - they could end up paying REAL MONEY.

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Re: FreeBSD as VOIP PBX

[Wojciech Puchar]

Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software

yes


are you using?  And any recommendations for software to look at would be

ports/net/asterisk
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RE: FreeBSD as VOIP PBX

[OutBackDingo]

if you want to suffer alot of pain on an incomplete system you could go
this way, id still build my own from ports 

On Fri, 2008-06-20 at 14:43 +0200, Johan Hendriks wrote:
> >Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software
> >are you using?  And any recommendations for software to look at would
> be
> >greatly appreciated.
> 
> >Shane
> 
> Also you can try the following
> 
> http://www.askozia.com/
> 
> based on FreeBSD
> working almost out of the box!
> 
> Regards,
> Johan
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RE: FreeBSD as VOIP PBX

[Johan Hendriks]

>Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software
>are you using?  And any recommendations for software to look at would
be
>greatly appreciated.

>Shane

Also you can try the following

http://www.askozia.com/

based on FreeBSD
working almost out of the box!

Regards,
Johan
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Re: FreeBSD as VOIP PBX

[OutBackDingo]

Yes
FreeBSD 7
Asterisk
Asterisk-GUI

both from ports

On Fri, 2008-06-20 at 07:49 -0400, Thomas Mullins wrote:
> Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software
> are you using?  And any recommendations for software to look at would be
> greatly appreciated.
> 
>  
> 
> Shane
> 
>  
> 
> ___
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FreeBSD as VOIP PBX

[Thomas Mullins]

Is anyone using FreeBSD for their VOIP PBX needs?  If so, what software
are you using?  And any recommendations for software to look at would be
greatly appreciated.

 

Shane

 

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Re: VoIP and SSH

[Andrew Falanga]

On Jan 7, 2008 8:45 AM, Jon Krause <[EMAIL PROTECTED]> wrote:

>  Andrew Falanga wrote:
>
> On Friday 04 January 2008 14:55:00 Jon Krause wrote:
>
>
>  Andrew Falanga wrote:
>
>
>  Hi,
>
> I don't understand this one and I'm hoping someone here might know.  My
> father's router wasn't forwarding connection requests for any port that
> we'd configured for sshd to listen on.  After changing out his linksys
> router and his Cable MODEM (the company said it was a very old modem),
> the problem was still present.  Oddly enough, if he unplugs his VoIP box
> from his network, all this problem goes away and connection requests over
> ssh and port 22 are forwarded fine.  With the VoIP box present, it
> doesn't work.
>
> Neither the FreeBSD machine or the VoIP box share IPs, but it doesn't
> work with the VoIP in the network.  Any ideas?
>
>
>  The "VoIP box" is usually an MTA, many include a router/firewall also.
> It should have an admin interface usually 192.168.0.1 or 192.168.1.1
> The cable company technical support should be able to walk you through
> getting access (or check any documentation that came with the MTA)
>
> They may or may not have port options (open or forward) that may allow
> ssh to work for you.
>
> Good Luck,
>
> Jon
>
>
>
>  Andy
> [EMAIL PROTECTED] mailing 
> listhttp://lists.freebsd.org/mailman/listinfo/freebsd-questions
> To unsubscribe, send any mail to"[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
>
>  Thanks to all for the suggestions.  I'll see what I can find.  Something one
> of you mentioned has me curious.  That being whether or not there is a DHCP
> server running.  My father's linksys router doles out IPs from 192.168.1.100 
> -  (I forget now).  Once, while trying to get this
> working, he logged into his system (from his system) using ssh.  What was odd
> was that he was able to log into his system by using the IP address of 
> 192.168.100.101, but using ifconfig he'd always tell me that the IP was 
> 192.168.1.100.  I'm betting that his VoIP box must be doling out IPs as well
> as his Linksys router, or something like that.
>
> Jon, what do you mean when you say, "The 'VoIP box' is usually an MTA?"  I'm
> used to MTA meaning Message Transfer Agent.  Is it the same in this case too?
>
>
>  Sorry for the late response.
>
> MTA = "multimedia terminal adapter"
> It's a Cable industry term most recently replaced by eMTA (embedded) where
> the MTA is embedded in the cablemodem.  Most commonly used by Comcast, Time
> Warner and others for Packet Cable telephone service.
>
>
Regardless of the response being late, thank you very much for the
response.  In fact, my father's cable modem service comes from Time Warner's
Road Runner service in upstate NY.  I'll bet this must be the problem.

Thanks again.

-- 
A: Because it messes up the order in which people normally read text.
Q: Why is it such a bad thing?
A: Top-posting.
Q: What is the most annoying thing on usenet and in e-mail?
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Re: VoIP and SSH

[Jon Krause]

Andrew Falanga wrote:

On Friday 04 January 2008 14:55:00 Jon Krause wrote:
  

Andrew Falanga wrote:


Hi,

I don't understand this one and I'm hoping someone here might know.  My
father's router wasn't forwarding connection requests for any port that
we'd configured for sshd to listen on.  After changing out his linksys
router and his Cable MODEM (the company said it was a very old modem),
the problem was still present.  Oddly enough, if he unplugs his VoIP box
from his network, all this problem goes away and connection requests over
ssh and port 22 are forwarded fine.  With the VoIP box present, it
doesn't work.

Neither the FreeBSD machine or the VoIP box share IPs, but it doesn't
work with the VoIP in the network.  Any ideas?
  

The "VoIP box" is usually an MTA, many include a router/firewall also.
It should have an admin interface usually 192.168.0.1 or 192.168.1.1
The cable company technical support should be able to walk you through
getting access (or check any documentation that came with the MTA)

They may or may not have port options (open or forward) that may allow
ssh to work for you.

Good Luck,

Jon



Andy
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Thanks to all for the suggestions.  I'll see what I can find.  Something one 
of you mentioned has me curious.  That being whether or not there is a DHCP 
server running.  My father's linksys router doles out IPs from 
192.168.1.100 -  (I forget now).  Once, while trying to get this 
working, he logged into his system (from his system) using ssh.  What was odd 
was that he was able to log into his system by using the IP address of 
192.168.100.101, but using ifconfig he'd always tell me that the IP was 
192.168.1.100.  I'm betting that his VoIP box must be doling out IPs as well 
as his Linksys router, or something like that.


Jon, what do you mean when you say, "The 'VoIP box' is usually an MTA?"  I'm 
used to MTA meaning Message Transfer Agent.  Is it the same in this case too?
  

Sorry for the late response.

MTA = "multimedia terminal adapter"
It's a Cable industry term most recently replaced by eMTA (embedded) 
where the MTA is embedded in the cablemodem.  Most commonly used by 
Comcast, Time Warner and others for Packet Cable telephone service.


Some service providers use the standard VoIP solutions (MTAs) or there 
are 3rd party solutions such as Vonage (also considered MTA). 
Most MTAs connect as follows:  Cablemodem > MTA > (phone line plugs into 
MTA) (ethernet port for the Internet)


The MTA acts as a router similar to a regular D-Link or Linksys (Cisco) 
home router.  They usually have a web interface for configuration, they 
have DHCP serving the 192.168.x.x IPs.  So it sounds like the MTA has 
DHCP'ed a 192.168.x.x address to the Linksys and the Linksys is doing 
it's own thing for his network.


You need to get into the Linksys status page to see what IP the MTA has 
issued to it.  Then try to access the MTA and see if you can "open" the 
ports of choice to the Linksys, then open the ports on the Linksys to 
his network or work-station.


Best,

Jon

Thanks,
Andy
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Re: VoIP and SSH

[Andrew Falanga]

On Friday 04 January 2008 14:55:00 Jon Krause wrote:
> Andrew Falanga wrote:
> > Hi,
> >
> > I don't understand this one and I'm hoping someone here might know.  My
> > father's router wasn't forwarding connection requests for any port that
> > we'd configured for sshd to listen on.  After changing out his linksys
> > router and his Cable MODEM (the company said it was a very old modem),
> > the problem was still present.  Oddly enough, if he unplugs his VoIP box
> > from his network, all this problem goes away and connection requests over
> > ssh and port 22 are forwarded fine.  With the VoIP box present, it
> > doesn't work.
> >
> > Neither the FreeBSD machine or the VoIP box share IPs, but it doesn't
> > work with the VoIP in the network.  Any ideas?
>
> The "VoIP box" is usually an MTA, many include a router/firewall also.
> It should have an admin interface usually 192.168.0.1 or 192.168.1.1
> The cable company technical support should be able to walk you through
> getting access (or check any documentation that came with the MTA)
>
> They may or may not have port options (open or forward) that may allow
> ssh to work for you.
>
> Good Luck,
>
> Jon
>
> > Andy
> > ___
> > freebsd-questions@freebsd.org mailing list
> > http://lists.freebsd.org/mailman/listinfo/freebsd-questions
> > To unsubscribe, send any mail to
> > "[EMAIL PROTECTED]"

Thanks to all for the suggestions.  I'll see what I can find.  Something one 
of you mentioned has me curious.  That being whether or not there is a DHCP 
server running.  My father's linksys router doles out IPs from 
192.168.1.100 -  (I forget now).  Once, while trying to get this 
working, he logged into his system (from his system) using ssh.  What was odd 
was that he was able to log into his system by using the IP address of 
192.168.100.101, but using ifconfig he'd always tell me that the IP was 
192.168.1.100.  I'm betting that his VoIP box must be doling out IPs as well 
as his Linksys router, or something like that.

Jon, what do you mean when you say, "The 'VoIP box' is usually an MTA?"  I'm 
used to MTA meaning Message Transfer Agent.  Is it the same in this case too?

Thanks,
Andy
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Re: VoIP and SSH

[Jon Krause]

Andrew Falanga wrote:

Hi,

I don't understand this one and I'm hoping someone here might know.  My 
father's router wasn't forwarding connection requests for any port that we'd 
configured for sshd to listen on.  After changing out his linksys router and 
his Cable MODEM (the company said it was a very old modem), the problem was 
still present.  Oddly enough, if he unplugs his VoIP box from his network, 
all this problem goes away and connection requests over ssh and port 22 are 
forwarded fine.  With the VoIP box present, it doesn't work.


Neither the FreeBSD machine or the VoIP box share IPs, but it doesn't work 
with the VoIP in the network.  Any ideas?


  
The "VoIP box" is usually an MTA, many include a router/firewall also.  
It should have an admin interface usually 192.168.0.1 or 192.168.1.1  
The cable company technical support should be able to walk you through 
getting access (or check any documentation that came with the MTA)


They may or may not have port options (open or forward) that may allow 
ssh to work for you.


Good Luck,

Jon

Andy
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Re: VoIP and SSH

[usleepless]

On Jan 4, 2008 9:29 PM, Ryan Phillips <[EMAIL PROTECTED]> wrote:
> Andrew Falanga <[EMAIL PROTECTED]> said:
> > Hi,
> >
> > I don't understand this one and I'm hoping someone here might know.  My
> > father's router wasn't forwarding connection requests for any port that we'd
> > configured for sshd to listen on.  After changing out his linksys router and
> > his Cable MODEM (the company said it was a very old modem), the problem was
> > still present.  Oddly enough, if he unplugs his VoIP box from his network,
> > all this problem goes away and connection requests over ssh and port 22 are
> > forwarded fine.  With the VoIP box present, it doesn't work.
> >
> > Neither the FreeBSD machine or the VoIP box share IPs, but it doesn't work
> > with the VoIP in the network.  Any ideas?
>
> Does the VoIP box provide DHCP?  Perhaps that conflicts with the router's
> DHCP service.

does the voip-box provide a ssh service? is it using upnp to
reconfigure the modem?

regards,

usleep
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Re: VoIP and SSH

[Ryan Phillips]

Andrew Falanga <[EMAIL PROTECTED]> said:
> Hi,
> 
> I don't understand this one and I'm hoping someone here might know.  My 
> father's router wasn't forwarding connection requests for any port that we'd 
> configured for sshd to listen on.  After changing out his linksys router and 
> his Cable MODEM (the company said it was a very old modem), the problem was 
> still present.  Oddly enough, if he unplugs his VoIP box from his network, 
> all this problem goes away and connection requests over ssh and port 22 are 
> forwarded fine.  With the VoIP box present, it doesn't work.
> 
> Neither the FreeBSD machine or the VoIP box share IPs, but it doesn't work 
> with the VoIP in the network.  Any ideas?

Does the VoIP box provide DHCP?  Perhaps that conflicts with the router's
DHCP service.

-ryan
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VoIP and SSH

[Andrew Falanga]

Hi,

I don't understand this one and I'm hoping someone here might know.  My 
father's router wasn't forwarding connection requests for any port that we'd 
configured for sshd to listen on.  After changing out his linksys router and 
his Cable MODEM (the company said it was a very old modem), the problem was 
still present.  Oddly enough, if he unplugs his VoIP box from his network, 
all this problem goes away and connection requests over ssh and port 22 are 
forwarded fine.  With the VoIP box present, it doesn't work.

Neither the FreeBSD machine or the VoIP box share IPs, but it doesn't work 
with the VoIP in the network.  Any ideas?

Andy
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Re: VoIP problems

[Predrag Punosevac]

Yes I do have mike connected (not muted) and permission files altered. I 
read 120 pages of OSS documentation and I tried every bloody thing. I 
will try one more time. Maybe I missed something. Thanks for your output.

Can you use VoIP? What is your sound card?

Thanks
Predrag


Norberto Meijome wrote:

On Mon, 27 Aug 2007 14:31:34 -0700
Predrag Punosevac <[EMAIL PROTECTED]> wrote:

  

Norberto Meijome wrote:


On Sat, 25 Aug 2007 23:05:20 -0700
Predrag Punosevac <[EMAIL PROTECTED]> wrote:

  
  
I was perfectly able to hear people using Skype but they could not hear 
me so my conclusion was that sound card is not properly configured
Another indication was that I was not able to record using ossrecord or 
at least when I play back I do not hear anything.



Predrag,
what's the output of mixer ? 
  


  
  
  
I am sending you two outputs. One is output of the native mixer and the 
second one is the output of the ossmix


[EMAIL PROTECTED] /usr/home/Pedja]$ mixer
Mixer pcm  is currently set to  68:68
Recording source:



fair enough, u use the OSS drivers...

  

[EMAIL PROTECTED] /usr/home/Pedja]$ ossmix
Selected mixer 0/
Known controls are:
pcm [:] (currently 68:68)
rear [:] (currently 48:48)
rear.rec ON|OFF (currently OFF)



rec off

  

center [:] (currently 48:48)
center.rec ON|OFF (currently OFF)



rec  off

  

ext.spread ON|OFF (currently OFF)
ext.loopback ON|OFF (currently OFF)
ext.recordvol  (currently 128)
ext.recordsrc  (currently MIC)
vmix0-src  (currently 
Fast)

vmix0-vol  (currently 25.0 dB)
vmix0-out :] (currently 0:0)
vmix0-out.pcm5  (currently 25.0 dB)
vmix0-out :] (currently 0:0)
vmix0-out.pcm6  (currently 25.0 dB)
vmix0-out :] (currently 0:0)
vmix0-out.pcm7  (currently 25.0 dB)
vmix0-out :] (currently 0:0)
vmix0-out.pcm8  (currently 25.0 dB)
vmix0-out :] (currently 0:0)
vmix0-in :] (currently 0:0)



I don't know too much about OSS utils / devices, but I would start by enabling 
the rec devices and seeing what changes every time increase the volume of 
everything all the way up (careful with your ears though!).

for reference, mine shows:

$ mixer
Mixer vol  is currently set to  86:86
Mixer pcm  is currently set to  85:85
Mixer speaker  is currently set to  49:49
Mixer mic  is currently set to   0:0
Mixer cd   is currently set to  40:40
Mixer rec  is currently set to 100:100
Recording source: mic

Did you check OSS docs  for samples? you DO have a mic connected to one of your inputs...and the mic is NOT muted right? (trust me, it happens to me every second day..) 


sorry I can't be of more help atm.

cheers,
beto
_
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If you find a solution and become attached to it, the solution may become your 
next problem.

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Reading disclaimers makes you go blind. Writing them is worse. You have been 
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Re: VoIP problems

[Norberto Meijome]

On Mon, 27 Aug 2007 14:31:34 -0700
Predrag Punosevac <[EMAIL PROTECTED]> wrote:

> Norberto Meijome wrote:
> > On Sat, 25 Aug 2007 23:05:20 -0700
> > Predrag Punosevac <[EMAIL PROTECTED]> wrote:
> >
> >   
> >> I was perfectly able to hear people using Skype but they could not hear 
> >> me so my conclusion was that sound card is not properly configured
> >> Another indication was that I was not able to record using ossrecord or 
> >> at least when I play back I do not hear anything.
> >> 
> >
> > Predrag,
> > what's the output of mixer ? 

> >   
> I am sending you two outputs. One is output of the native mixer and the 
> second one is the output of the ossmix
> 
> [EMAIL PROTECTED] /usr/home/Pedja]$ mixer
> Mixer pcm  is currently set to  68:68
> Recording source:

fair enough, u use the OSS drivers...

> [EMAIL PROTECTED] /usr/home/Pedja]$ ossmix
> Selected mixer 0/
> Known controls are:
> pcm [:] (currently 68:68)
> rear [:] (currently 48:48)
> rear.rec ON|OFF (currently OFF)

rec off

> center [:] (currently 48:48)
> center.rec ON|OFF (currently OFF)

rec  off

> ext.spread ON|OFF (currently OFF)
> ext.loopback ON|OFF (currently OFF)
> ext.recordvol  (currently 128)
> ext.recordsrc  (currently MIC)
> vmix0-src  (currently 
> Fast)
> vmix0-vol  (currently 25.0 dB)
> vmix0-out :] (currently 0:0)
> vmix0-out.pcm5  (currently 25.0 dB)
> vmix0-out :] (currently 0:0)
> vmix0-out.pcm6  (currently 25.0 dB)
> vmix0-out :] (currently 0:0)
> vmix0-out.pcm7  (currently 25.0 dB)
> vmix0-out :] (currently 0:0)
> vmix0-out.pcm8  (currently 25.0 dB)
> vmix0-out :] (currently 0:0)
> vmix0-in :] (currently 0:0)

I don't know too much about OSS utils / devices, but I would start by enabling 
the rec devices and seeing what changes every time increase the volume of 
everything all the way up (careful with your ears though!).

for reference, mine shows:

$ mixer
Mixer vol  is currently set to  86:86
Mixer pcm  is currently set to  85:85
Mixer speaker  is currently set to  49:49
Mixer mic  is currently set to   0:0
Mixer cd   is currently set to  40:40
Mixer rec  is currently set to 100:100
Recording source: mic

Did you check OSS docs  for samples? you DO have a mic connected to one of your 
inputs...and the mic is NOT muted right? (trust me, it happens to me every 
second day..) 

sorry I can't be of more help atm.

cheers,
beto
_
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If you find a solution and become attached to it, the solution may become your 
next problem.

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Reading disclaimers makes you go blind. Writing them is worse. You have been 
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Re: VoIP problems

[Predrag Punosevac]

If you want to post a  question you need to start a new thread instead 
of erasing the content of mine and replacing with your question.

This is not how we behave on this mailing list.

So what did you have to say about configuring Open Sound System? I am 
all ears.

Sincerely,
Predrag Punosevac

Robert Nicholson wrote:

Hi everyone,

I just installed FreeBSD 6.0 on a HP Compaq DC7600 Small Form Factor PC at
work. It uses an intel 945G chipset. I could not get the broadcom NIC to
work so we replaced it with a D-Link NIC and that works.

The other problem is that the four USB ports on the machine are recognized
and the /dev/ directory has character devices usb1 to usb4 but plugging in
any usb drive (including a USB pen drive) causes the system to hang for
about 4 seconds and then  the drive is not recognized. There are no /dev/da*
devices, no dmesg messages and no /var/log/messages either.

I checked the kernel config. Devices umass, ehci, ohci , uhci, usb. da as
well as scbus are all enabled. I am at a loss on how to solve this problem.
Please help.

Thanks and Regards,
Michael.
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Re: VoIP problems

[Predrag Punosevac]

Norberto Meijome wrote:

On Sat, 25 Aug 2007 23:05:20 -0700
Predrag Punosevac <[EMAIL PROTECTED]> wrote:

  
I was perfectly able to hear people using Skype but they could not hear 
me so my conclusion was that sound card is not properly configured
Another indication was that I was not able to record using ossrecord or 
at least when I play back I do not hear anything.



Predrag,
what's the output of mixer ? 


_
{Beto|Norberto|Numard} Meijome

"The people have always some champion whom they set over them and nurse into 
greatness...
 This and no other is the root from which a tyrant springs; when he first appears he 
is a protector."
   Plato

I speak for myself, not my employer. Contents may be hot. Slippery when wet. 
Reading disclaimers makes you go blind. Writing them is worse. You have been 
Warned.
  
I am sending you two outputs. One is output of the native mixer and the 
second one is the output of the ossmix


[EMAIL PROTECTED] /usr/home/Pedja]$ mixer
Mixer pcm  is currently set to  68:68
Recording source:


[EMAIL PROTECTED] /usr/home/Pedja]$ ossmix
Selected mixer 0/
Known controls are:
   pcm [:] (currently 68:68)
   rear [:] (currently 48:48)
   rear.rec ON|OFF (currently OFF)
   center [:] (currently 48:48)
   center.rec ON|OFF (currently OFF)
   ext.spread ON|OFF (currently OFF)
   ext.loopback ON|OFF (currently OFF)
   ext.recordvol  (currently 128)
   ext.recordsrc  (currently MIC)
   vmix0-src  (currently 
Fast)

   vmix0-vol  (currently 25.0 dB)
   vmix0-out :] (currently 0:0)
   vmix0-out.pcm5  (currently 25.0 dB)
   vmix0-out :] (currently 0:0)
   vmix0-out.pcm6  (currently 25.0 dB)
   vmix0-out :] (currently 0:0)
   vmix0-out.pcm7  (currently 25.0 dB)
   vmix0-out :] (currently 0:0)
   vmix0-out.pcm8  (currently 25.0 dB)
   vmix0-out :] (currently 0:0)
   vmix0-in :] (currently 0:0)


I tried to play with that put I guess I am just ignorant.

Thanks
Predrag


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Re: VoIP behind NAT and FreeBSD

[Kurt Dethier]

Marwan Sultan wrote:

First thanks for you all, for the cooperating,

 My setup is as follow,

 Router <-> vr0 FreeBSD fxp0 <-> Switch <-> Clients
 Two NICs attached,
  vr0 connected to the router (internet interface) has the static 
192.168.0.2
  fxp0 connected to the Switch connected to clients acting as DHCP 
192.168.182.1

  and clients are assigned 192.168.182.* ofcourse..

  kernel as you know configured for ipfw and NAT
  chillispot installed, and controlling the fxp0 device thro the DHCP
  assigning for clients the IPs. however chillispot is not an issue.

  This is the configuration,
  I think chillispot using pf rules. once a user will sign in, all 
blocks will be remmoved.

  just for the info,


Hi Marwan,
I have never used ipfw before, so I suggest to do a quick check what the
NAT type is. Last time I checked, there was a simple client in the 
vovidia stun implementation. They also run a public server for testing.


If your NAT type is not symmetric NAT, you can use STUN (you will need
2 ip addresses on the internet side of your gateway for STUN to work).

If your NAT type is symmetric NAT, I suggest to look at UPNP. Most 
clients support it.
If UPNP is not an options, I guess proxying the media streams, or 
rewriting the signaling is your only options left.


Kurt


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Re: VoIP behind NAT and FreeBSD

[Marwan Sultan]

First thanks for you all, for the cooperating,

 My setup is as follow,

 Router <-> vr0 FreeBSD fxp0 <-> Switch <-> Clients
 Two NICs attached,
  vr0 connected to the router (internet interface) has the static 
192.168.0.2
  fxp0 connected to the Switch connected to clients acting as DHCP 
192.168.182.1

  and clients are assigned 192.168.182.* ofcourse..

  kernel as you know configured for ipfw and NAT
  chillispot installed, and controlling the fxp0 device thro the DHCP
  assigning for clients the IPs. however chillispot is not an issue.

  This is the configuration,
  I think chillispot using pf rules. once a user will sign in, all blocks 
will be remmoved.

  just for the info,

  Any I appreciate your help gurus,
  Any advices how to setup?

  Marwan Sultan



Hi all,
It is very possible to use VOIP behind a symmetric NAT, but STUN is not
going to be any help. Depending on the setup and clients I have
implemented a number of solutions over the past years.
Unfortunately I haven't found a single solutions that always works.


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Re: VoIP behind NAT and FreeBSD

[Garrett Cooper]

Kurt Dethier wrote:

Derrick Edwards wrote:

On Saturday 02 December 2006 09:22, Joe Holden wrote:

Kurt Dethier wrote:

STUN will only work if you have the correct NAT implementation on
your gateway. If you are using pf, you get what the STUN RFC calls
a symmetric NAT. STUN will not help you in such an implementation.
I'm not sure how the other NAT solutions on FreeBSD are implemented.

If you need a solution for a symmetric NAT, you need something that
understands the signaling protocol and can add fw/nat rules on demand
on your gateway, or a media proxy (like TURN).

Kurt

 >>

It is entirely possible to use voip behnd symmetric nat, but it also
entirely depends on the setup, some more details will help.

Hi, I am not sure of your setup either, but I have Vongae working 
behind a FreeBSD Firewall/Router using PF with NAT. 


Hi all,
It is very possible to use VOIP behind a symmetric NAT, but STUN is not
going to be any help. Depending on the setup and clients I have
implemented a number of solutions over the past years.
Unfortunately I haven't found a single solutions that always works.

Kurt


Forgive me if I'm not understanding the issue properly, but don't you 
have port-forwarding setup on the FreeBSD box for the machine that you 
are trying to use VoIP with? It seems like the problem would *sort* or 
be that simple to solve, unless the VoIP setup uses a P2P type 
configuration where it picks multiple ports for listening and 
transferring data.


Maybe it's just my misunderstanding of VoIP..
-Garrett
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Re: VoIP behind NAT and FreeBSD

[Kurt Dethier]

Derrick Edwards wrote:

On Saturday 02 December 2006 09:22, Joe Holden wrote:

Kurt Dethier wrote:

STUN will only work if you have the correct NAT implementation on
your gateway. If you are using pf, you get what the STUN RFC calls
a symmetric NAT. STUN will not help you in such an implementation.
I'm not sure how the other NAT solutions on FreeBSD are implemented.

If you need a solution for a symmetric NAT, you need something that
understands the signaling protocol and can add fw/nat rules on demand
on your gateway, or a media proxy (like TURN).

Kurt

>>

It is entirely possible to use voip behnd symmetric nat, but it also
entirely depends on the setup, some more details will help.

	Hi, 
I am not sure of your setup either, but I have Vongae working behind a FreeBSD 
Firewall/Router using PF with NAT. 


Hi all,
It is very possible to use VOIP behind a symmetric NAT, but STUN is not
going to be any help. Depending on the setup and clients I have
implemented a number of solutions over the past years.
Unfortunately I haven't found a single solutions that always works.

Kurt

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Re: VoIP behind NAT and FreeBSD

[Derrick Edwards]

On Saturday 02 December 2006 09:22, Joe Holden wrote:
> Kurt Dethier wrote:
> > Marwan Sultan wrote:
> >> Hello All,
> >>
> >>  Well, maybe the subject says all,
> >>  Im running 6.1R acting as NAT, gateway ofcourse, hotspot.
> >>  I have many clients trying to use Vonage, motorola, VoIP devices and
> >>  and few more products.
> >>
> >>  The problem is as its described in some websites..
> >>  They can call, receive a call, hear the dailtone but no one can hear
> >> the other party.
> >>
> >>   After researching i found out there is a problem iusing FreeBSD NAT
> >> with voip protocols.
> >>
> >>   i have been advised to use STUN servers, (Simple Traversal of UDP
> >> through NATs.)
> >>   I found out there is net/stund port and its installed successufly!
> >>   but still lost.
> >>
> >>   Can someone kindly and please shade a light on howto
> >>   make voip behind NAT works in my FreeBSD ? im in trouble because of
> >> this.
> >
> > Hi Marwan,
> > STUN will only work if you have the correct NAT implementation on
> > your gateway. If you are using pf, you get what the STUN RFC calls
> > a symmetric NAT. STUN will not help you in such an implementation.
> > I'm not sure how the other NAT solutions on FreeBSD are implemented.
> >
> > If you need a solution for a symmetric NAT, you need something that
> > understands the signaling protocol and can add fw/nat rules on demand
> > on your gateway, or a media proxy (like TURN).
> >
> > Kurt
>
> It is entirely possible to use voip behnd symmetric nat, but it also
> entirely depends on the setup, some more details will help.
>
> Ta,
> Joe

Hi, 
I am not sure of your setup either, but I have Vongae working behind a FreeBSD 
Firewall/Router using PF with NAT. 

Vonage Traffic viewed via PFTOP.

udpOut 172.16.1.1:1  69.59.242.83:1   MULTIPLE:MULTIPLE
2688m48  18146  9967K

udpOut 172.16.1.1:10112  69.59.243.178:12044  MULTIPLE:MULTIPLE 
  
6133   3382 674816

udpOut 172.16.1.1:10113  69.59.243.178:12045SINGLE:NO_TRAFFIC   
  
56 2 14   2348

v/r
Derrick



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Re: VoIP behind NAT and FreeBSD

[Joe Holden]

Kurt Dethier wrote:
> Marwan Sultan wrote:
>> Hello All,
>>
>>  Well, maybe the subject says all,
>>  Im running 6.1R acting as NAT, gateway ofcourse, hotspot.
>>  I have many clients trying to use Vonage, motorola, VoIP devices and
>>  and few more products.
>>
>>  The problem is as its described in some websites..
>>  They can call, receive a call, hear the dailtone but no one can hear
>> the other party.
>>
>>   After researching i found out there is a problem iusing FreeBSD NAT
>> with voip protocols.
>>
>>   i have been advised to use STUN servers, (Simple Traversal of UDP
>> through NATs.)
>>   I found out there is net/stund port and its installed successufly!
>>   but still lost.
>>
>>   Can someone kindly and please shade a light on howto
>>   make voip behind NAT works in my FreeBSD ? im in trouble because of
>> this.
> 
> Hi Marwan,
> STUN will only work if you have the correct NAT implementation on
> your gateway. If you are using pf, you get what the STUN RFC calls
> a symmetric NAT. STUN will not help you in such an implementation.
> I'm not sure how the other NAT solutions on FreeBSD are implemented.
> 
> If you need a solution for a symmetric NAT, you need something that
> understands the signaling protocol and can add fw/nat rules on demand
> on your gateway, or a media proxy (like TURN).
> 
> Kurt

It is entirely possible to use voip behnd symmetric nat, but it also
entirely depends on the setup, some more details will help.

Ta,
Joe
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Re: VoIP behind NAT and FreeBSD

[Kurt Dethier]

Marwan Sultan wrote:

Hello All,

 Well, maybe the subject says all,
 Im running 6.1R acting as NAT, gateway ofcourse, hotspot.
 I have many clients trying to use Vonage, motorola, VoIP devices and
 and few more products.

 The problem is as its described in some websites..
 They can call, receive a call, hear the dailtone but no one can hear 
the other party.


  After researching i found out there is a problem iusing FreeBSD NAT 
with voip protocols.


  i have been advised to use STUN servers, (Simple Traversal of UDP 
through NATs.)

  I found out there is net/stund port and its installed successufly!
  but still lost.

  Can someone kindly and please shade a light on howto
  make voip behind NAT works in my FreeBSD ? im in trouble because of this.


Hi Marwan,
STUN will only work if you have the correct NAT implementation on
your gateway. If you are using pf, you get what the STUN RFC calls
a symmetric NAT. STUN will not help you in such an implementation.
I'm not sure how the other NAT solutions on FreeBSD are implemented.

If you need a solution for a symmetric NAT, you need something that
understands the signaling protocol and can add fw/nat rules on demand
on your gateway, or a media proxy (like TURN).

Kurt

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Re: VoIP behind NAT and FreeBSD

[Joe Holden]

Marwan Sultan wrote:
> Hello All,
> 
>  Well, maybe the subject says all,
>  Im running 6.1R acting as NAT, gateway ofcourse, hotspot.
>  I have many clients trying to use Vonage, motorola, VoIP devices and
>  and few more products.
> 
>  The problem is as its described in some websites..
>  They can call, receive a call, hear the dailtone but no one can hear
> the other party.
> 
>   After researching i found out there is a problem iusing FreeBSD NAT
> with voip protocols.
> 
>   i have been advised to use STUN servers, (Simple Traversal of UDP
> through NATs.)
>   I found out there is net/stund port and its installed successufly!
>   but still lost.
> 
>   Can someone kindly and please shade a light on howto
>   make voip behind NAT works in my FreeBSD ? im in trouble because of this.
> 
>   Thank you.
>   Marwan Sultan,
> 
This isn't FreeBSD NAT specific, its a problem with NAT in general, if
you've installed the stund port, all you need todo is run: stund -h
1.2.3.4 -a 1.2.3.4, replacing 1.2.3.4 with the machines ip obviously,
and then tell the voip phones to use that stun server.

HTH,
Joe
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VoIP behind NAT and FreeBSD

[Marwan Sultan]

Hello All,

 Well, maybe the subject says all,
 Im running 6.1R acting as NAT, gateway ofcourse, hotspot.
 I have many clients trying to use Vonage, motorola, VoIP devices and
 and few more products.

 The problem is as its described in some websites..
 They can call, receive a call, hear the dailtone but no one can hear the 
other party.


  After researching i found out there is a problem iusing FreeBSD NAT with 
voip protocols.


  i have been advised to use STUN servers, (Simple Traversal of UDP through 
NATs.)

  I found out there is net/stund port and its installed successufly!
  but still lost.

  Can someone kindly and please shade a light on howto
  make voip behind NAT works in my FreeBSD ? im in trouble because of this.

  Thank you.
  Marwan Sultan,

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Re: anyone using voip?

[Greg 'groggy' Lehey]

On Thursday, 23 February 2006 at  0:43:22 +, Alec Berryman wrote:
> Peter on 2006-02-22 18:55:34 -0500:
>
>>
>> --- Alec Berryman <[EMAIL PROTECTED]> wrote:
>>>
>>> I'm using Skype from ports with an analog headset and it works
>>> fantastically well.
>>
>> I installed skype from ports.  Without a man page it was difficult to
>> begin.  I decided to execute the binary and it just sat there without any
>> output whatsover.  Do you have any notes?
>
> It just started up for me.  It's a Linux binary; have you properly
> enabled Linux compatibility as described in the handbook?  You might
> refer to the documentation on www.skype.com, but it was very much
> plug-and-play.

I'd guess that Peter is asking: "OK, it works.  Now how do I use
it?".  That's where I was left with Skype.  I suppose that would
change if I had anybody to talk to over it.  But it's a proprietary
solution where there are open solutions; why choose it?

Before somebody answers, I know the answer: it gives you ($) access to
POTS, and it works (apparently) better than some open VoIP solutions.
But I'd rather fix the open solution.

Greg
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pgpsxrd2bSwxj.pgp
Description: PGP signature

Re: anyone using voip?

[Alec Berryman]

Peter on 2006-02-22 18:55:34 -0500:

> 
> --- Alec Berryman <[EMAIL PROTECTED]> wrote:
> > 
> > I'm using Skype from ports with an analog headset and it works
> > fantastically well.
> 
> I installed skype from ports.  Without a man page it was difficult to
> begin.  I decided to execute the binary and it just sat there without any
> output whatsover.  Do you have any notes?

It just started up for me.  It's a Linux binary; have you properly
enabled Linux compatibility as described in the handbook?  You might
refer to the documentation on www.skype.com, but it was very much
plug-and-play.


pgpWdygbvX2py.pgp
Description: PGP signature

Re: anyone using voip?

[Peter]

--- Alec Berryman <[EMAIL PROTECTED]> wrote:

> Peter on 2006-02-21 22:07:11 -0500:
> 
> > Hi, I'm looking for comments from people who are using a voip solution
> > with FreeBSD.  The archives of this group show mixed results.  I see
> > there is a skype port available.  To me that implies that this is
> > possible.  What of hardware?  USB phones?
> 
> I'm using Skype from ports with an analog headset and it works
> fantastically well.

I installed skype from ports.  Without a man page it was difficult to
begin.  I decided to execute the binary and it just sat there without any
output whatsover.  Do you have any notes?






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Re: anyone using voip?

[Alec Berryman]

Peter on 2006-02-21 22:07:11 -0500:

> Hi, I'm looking for comments from people who are using a voip solution
> with FreeBSD.  The archives of this group show mixed results.  I see
> there is a skype port available.  To me that implies that this is
> possible.  What of hardware?  USB phones?

I'm using Skype from ports with an analog headset and it works
fantastically well.


pgpvcSD5ct6I8.pgp
Description: PGP signature

Re: anyone using voip?

[Greg 'groggy' Lehey]

On Tuesday, 21 February 2006 at 22:07:11 -0500, Peter wrote:
> Hi, I'm looking for comments from people who are using a voip solution
> with FreeBSD.  The archives of this group show mixed results.  I see there
> is a skype port available.  To me that implies that this is possible.
> What of hardware?  USB phones?

I've done a lot of investigation.  Summary: (not only under FreeBSD):
VoIP software is *really* bad.  Asterisk may work if you can
understand the arcane documentation, but it's overkill for a simple
VoIP phone solution.  The others are almost completely undocumented
and difficult to use.  This applies to commercial offerings too, some
of which are free to use.

I'm currently using linphone, mainly because it has a command-line
client, and I don't see why I should have to use a mouse to make a
phone call.  The client is buggy, though, and very non-intuitive.  For
example:  to call me, you might enter

  $ linphonec sip:[EMAIL PROTECTED]

This will work.  But I'd then expect the program to wait until
termination or until I hit ^C; instead, it returns with a prompt.
Hitting ^C stops the program without terminating the call.  To
terminate the call, you need to enter the entire text "terminate".
Enter ^D to exit the program and it loops.  That's when you need the
^C.

From a hardware point of view, I'm using a standard analogue headset
with microphone.  You'll need to set the recording source to
microphone:

  $ mixer =rec mic

Greg
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Re: anyone using voip?

[Ken Stevenson]

Peter wrote:

Hi, I'm looking for comments from people who are using a voip solution
with FreeBSD.  The archives of this group show mixed results.  I see there
is a skype port available.  To me that implies that this is possible. 
What of hardware?  USB phones?


--
Peter



I've started playing around with it at home.

I installed the asterisk port on FreeBSD and downloaded the free 
Counterpath X-Lite softphone for XP. Using a headset (Plantronics 
USB), I've gotten as far as calling the asterisk demo on my server.


I got an account on FWD (Free World Dialup) for testing, and also 
downloaded and registered with Skype.


Check out voip-info.org as a resource.

As far as hardware, you might want to look at SNOM phones or Polycom 
phones for SIP based IP phones, and Sipura as an ATA to use with 
standard phones.


Here's an article on how one guy setup asterisk for use at home:

http://arstechnica.com/guides/tweaks/voip.ars

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Re: anyone using voip?

[Erik Norgaard]

Peter wrote:

Hi, I'm looking for comments from people who are using a voip solution
with FreeBSD.  The archives of this group show mixed results.  I see there
is a skype port available.  To me that implies that this is possible. 
What of hardware?  USB phones?


You are looking for a VoIP client? there are some available such as 
kphone, alternatively linphone. With the exception of skype which does 
not support the open protocols I haven't had much luck to find a client 
that can traverse natting firewalls.


Cheers, Erik
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Re: anyone using voip?

[Kris Anderson]

--- Peter <[EMAIL PROTECTED]> wrote:

> Hi, I'm looking for comments from people who are
> using a voip solution
> with FreeBSD.  The archives of this group show mixed
> results.  I see there
> is a skype port available.  To me that implies that
> this is possible. 
> What of hardware?  USB phones?
> 
> --
> Peter
Check out /usr/ports/net/asterisk/pkg-descr

It's a start in some direction.

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anyone using voip?

[Peter]

Hi, I'm looking for comments from people who are using a voip solution
with FreeBSD.  The archives of this group show mixed results.  I see there
is a skype port available.  To me that implies that this is possible. 
What of hardware?  USB phones?

--
Peter






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Re: Native voip software for FreeBSD

[Eric Kjeldergaard]

水曜日 11 1月 2006 01:30、User Gandalf さんは書きました:
>   Hello,
>
> Do you know any native voip software for FreeBSD?
> I hate to install Linux compatibility mode just to have skype working.
> Thanks,
>
>Les

<[EMAIL PROTECTED]> [/usr/ports/net] #ls | grep phone
cphone
kphone
linphone
linphone-base
ohphone

Those as well as kiax seem good options, I think.  I've been meaning to see if 
openwengo can compile and run on freebsd.  Best of luck,

Eric

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Re: Native voip software for FreeBSD

[Andrew P.]

On 1/10/06, User Gandalf <[EMAIL PROTECTED]> wrote:
>
>   Hello,
>
> Do you know any native voip software for FreeBSD?
> I hate to install Linux compatibility mode just to have skype working.
> Thanks,
>
>Les
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>

mbone/speak_freely
net/kphone
security/cutlass
asterisk-related
...
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Native voip software for FreeBSD

[User Gandalf]

 Hello,

Do you know any native voip software for FreeBSD?
I hate to install Linux compatibility mode just to have skype working.
Thanks,

  Les
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Re: Voip - solutions!

[Erik Norgaard]

On Wed, 19 Oct 2005, Carstea Catalin wrote:


Please give me some free (open source) solutions for VoIP over FreeBSD!
--
Any help would be greatly appreciated.


Do you have any idea of what you need? Are you looking for server 
solutions or VoIP clients?


* ser is a sip router
* siproxd a sip proxy if your clients are behind a NAT/firewall
* asterix is both - I think

For clients: linphone and kphone, I have tried to play with both 
and linphone appaers most stable. For clients, some supports 
automatic discover of NAT and external IP and need no proxy, 
others don't and require a proxy to work behind a firewall.


BTW: Anyone knows a some sort of VoIP answering machine I can 
call up and test agains without bothering other people too much?


Cheers, Erik
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Re: Voip - solutions!

[Jason Stewart]

On 19/10/05 05:02 -0700, Carstea Catalin wrote:
> Please give me some free (open source) solutions for VoIP over FreeBSD!

www.asterisk.org

Jason
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Voip - solutions!

[Carstea Catalin]

Please give me some free (open source) solutions for VoIP over FreeBSD!
--
Any help would be greatly appreciated.
regards,
Carstea Catalin
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Re: Trying to use FreeBSD as a home router, how to setup VOIP to pass through?

[Reko Turja]

computers) behind the freebsd machine (computer 4).  The problem is 
that i'd like to move the voip router behind the freebsd machine. 
I'm assuming i need to do some sort of port forwarding to pull this


I'm using FreeBSD 5.4 stable (week or two old), ipfw and natd with a 
divert rule in place and practically no other configuration.  Does 
anyone have any resources on forwarding voip traffic?


I once had done a bit similar setup for our company. Due some quick & 
dirty solutions in our network nodes the VoIP and networked machines 
used same switches but were in two separate network segments with one 
gateway for each in the end of the mother company.


As routing two different IP address ranges via same firewall proved to 
be impossible due gateway issues, FreeBSD bridging firewall came to the 
rescue (with added bonus of getting "invisible" packet filter in the 
route).


There is IIRC quite good documentation on the FreeBSD site concerning 
setting up bridging firewall, so you could check that and use those 
instructions. To be honest I can't remember the VoIP stuff and the FW 
setup I made anymore, but some poking around in web should be able to 
recover those details.


-Reko 


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Trying to use FreeBSD as a home router, how to setup VOIP to pass through?

[Lucas Holt]

My home network is currently setup with a lowend router from my phone  
company.  Its not very good and i have my only phone service through  
it using VOIP.  I've managed to get the rest of my network (3  
computers) behind the freebsd machine (computer 4).  The problem is  
that i'd like to move the voip router behind the freebsd machine.   
I'm assuming i need to do some sort of port forwarding to pull this  
off.  I've done some googling but can't seem to find good  
documentation on what ports to forward or how one would do port  
fowarding to another host.


I'm using FreeBSD 5.4 stable (week or two old), ipfw and natd with a  
divert rule in place and practically no other configuration.  Does  
anyone have any resources on forwarding voip traffic?


My long term goal is to get voip working and then setup an ip 6  
tunnel with HE.


I am not currently subscribed to questions so please CC me in replies.

Here's a rough idea what my network is like

cable modem
   +
   +
  voip router
  +
  +
  freebsd machine
  +  + +
  pcmacmac

Everything is setup wtih IP4 at the moment.

Lucas Holt
[EMAIL PROTECTED]

FoolishGames.com  (Jewel Fan Site)
JustJournal.com (Free blogging)
FoolishGames.net (Enemy Territory IoM site)

Think PC.. in 2006 you can own an Apple PCintosh. Whats next, windows  
works?


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IPF, VOIP, packet priority?

[Matt Staroscik]

I use a 4.10 FreeBSD box as a NAT device, IPF firewall and DHCP server. 
I recently got VOIP service from my ISP (Speakeasy). The VOIP box they 
sent me goes in between the NAT box and the DSL modem, and it is 
supposed to prioritize VOIP traffic so calls are clear. But, if I am 
doing a download my calls do get a little choppy. It's tolerable but I 
would like to improve it.

How can I prioritize the VOIP traffic under 4.10 with IPF? Googling has 
shown me (http://www.muine.org/~hoang/freenat.html) that I need the 
ipfirewall + dummynet options in my kernel to enable QoS, but I am 
unlcear on how that will work with my existing ipf firewall rules.

Thanks for any leads.
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VOIP

[Sean]

All,
	Can anyone recommend a VOIP software package that will run on an AMD64 
in 64 bit mode?

Thanks
Sean
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Re: FreeBSD box as a VOIP gateway for calling card co.

[Eihab E. Ibrahim]

- Original Message - 
From: "Hadi Maleki-Baroogh" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 01, 2004 1:39 AM
Subject: FreeBSD box as a VOIP gateway for calling card co.


Hi,
Anyone have any howto pages or any web sites where I can find on setting up 
a freebsd box as a voip gateway for a phone card co im looking into?
I'm not quiet sure, but I think you'll find Asterisk and SER interesting.
Check:
Asterisk: http://www.asteriskpbx.com
SER: http://www.iptel.org/ser/
They're both available in the ports collection:
ports/net/asterisk and ports/net/ser.
VOIP info http://www.voip-info.org can be very helpful as well.
Hope this helped.
Eihab E. Ibrahim
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FreeBSD box as a VOIP gateway for calling card co.

[Hadi Maleki-Baroogh]

Hi,
Anyone have any howto pages or any web sites where I can find on setting up 
a freebsd box as a voip gateway for a phone card co im looking into?

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Recommends for VoIP handset to use with 5.3 on laptop?

[Marc]

Hello -

I'd like to be able to use VoIP to make in and outbound calls, via a
laptop running FreeBSD 5.3. Can anyone make recommendations for devices
that can be used for this purpose?

For example, I have an i300 handset (keypad, ear speaker and mic) that
works under Windows/XP. It is a USB device. Are there similar devices that
work with skype or some other software package that people can recommend?

Please include me directly in responses, as I am not a subscriber to this
mailing list.

Thanks in advance - Marc
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Re: VoIP: sip client

[Erik Norgaard]

jason wrote:

> I found this http://www.voip-info.org/wiki-SIP.  It might help you out.

Thanks, I have been through a number of sites, asterisk.org, iptel.org,
voip-forum.com and the above - but I missed that page.

On freshmeat I have found a project that looks interesting: minisip, see
www.minisip.org. It appears to do just what I have been looking for -
that is VoIP/SIP and not much else. But the interesting part is that
they add some security enhancements, SIP over TLS, SRTP and MIKEY which
are still work-in-progress IETF standards.

However, it is for Linux, GPL, in alpha and not ported - anyone wants to
help?

Cheers, Erik
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Re: VoIP: sip client

[jason]

Erik Norgaard wrote:
Hi,
I am trying to find a SIP client to work behind an ADSL router with NAT.
I have tried linphone, but it seems not to support STUN, and I have
tried kphone which crashes regularly and I have no sound.
Is there another SIP client that works? Or should I try setup Asterisk
or SER to proxy calls from linphone?
Sorry, I'm new to VoIP and asking the _right question_ (TM) is
difficult. Any suggestions, directional pointers or references would be
greatly appreciated.
Thanks, Erik
 

I found this http://www.voip-info.org/wiki-SIP.  It might help you out.
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VoIP: sip client

[Erik Norgaard]

Hi,

I am trying to find a SIP client to work behind an ADSL router with NAT.
I have tried linphone, but it seems not to support STUN, and I have
tried kphone which crashes regularly and I have no sound.

Is there another SIP client that works? Or should I try setup Asterisk
or SER to proxy calls from linphone?

Sorry, I'm new to VoIP and asking the _right question_ (TM) is
difficult. Any suggestions, directional pointers or references would be
greatly appreciated.

Thanks, Erik

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Re: VoIP World Leaders

[Gary Kline]

On Sat, Oct 02, 2004 at 03:38:02PM -0500, Jay Moore wrote:
> On Monday 27 September 2004 05:44 am, Spidey Knepscheld wrote:
> > Hi Guys
> >
> >
> > Can anyone perhaps inform me on the world leader in VoIP Solutions.We
> > were granted a license to supply VoIP in South Africa and we would like
> > to get in contact with the big guys in this field.
> >
> > Thank you
> >
> >
> > Spidey
> 
> This might be the lamest question ever asked on any mailing list or usenet 
> group in the history of the Internet... where's that Guinness Records book?
> 

Naah, I cam easily come up with something lamer: like:
"Why do we even need computers besides those of 
Microsoft's multi-billion-dollar expertise?"

gary



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Re: VoIP World Leaders

[Jay Moore]

On Monday 27 September 2004 05:44 am, Spidey Knepscheld wrote:
> Hi Guys
>
>
> Can anyone perhaps inform me on the world leader in VoIP Solutions.We
> were granted a license to supply VoIP in South Africa and we would like
> to get in contact with the big guys in this field.
>
> Thank you
>
>
> Spidey

This might be the lamest question ever asked on any mailing list or usenet 
group in the history of the Internet... where's that Guinness Records book?

Jay
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Re: VOIP

[Bill Moran]

"Peter Mussett" <[EMAIL PROTECTED]> wrote:
> Dear Sir/Madam
> 
> We are an import/export timber company in Australia who has many sites and
> suppliers around the world.
> Most important is our office and suppliers in P.N.G, we are looking to setup
> a VOIP server here in Australia to
> Manage and be in constant communication with our site and our suppliers in
> P.N.G.
> And is all goes well use this server to expand the technology so it can be
> available to other businesses/homes in P.N.G.
> Any information you can provide would be most appreciated.

http://www.asterisk.org

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VOIP

[Peter Mussett]

Dear Sir/Madam

We are an import/export timber company in Australia who has many sites and
suppliers around the world.
Most important is our office and suppliers in P.N.G, we are looking to setup
a VOIP server here in Australia to
Manage and be in constant communication with our site and our suppliers in
P.N.G.
And is all goes well use this server to expand the technology so it can be
available to other businesses/homes in P.N.G.
Any information you can provide would be most appreciated.

Thank you for your time

Kind Regards

Will Mussett
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Re: VOIP

[simon butsana]

Hi,
 
Yes.
 
I am currently working on deploying an H323 based VoIP network in FreeBSD 5.1
Do you have any specific question?
 
Simon

"Stanford .T. Mings Jr." <[EMAIL PROTECTED]> wrote:
Is anyone doing any work in VOIP in FreeBSD ?

stm
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Re: VOIP

[Gustaaf Wijnands]

Stanford .T. Mings Jr. wrote:
Is anyone doing any work in VOIP in FreeBSD ?
Did you have a look at Skype ( /usr/ports/net/skype ), www.skype.com
or do you mean something else?
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VOIP

[Stanford .T. Mings Jr.]

Is anyone doing any work in VOIP in FreeBSD ?

stm
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Re: Firewall for VoIP box

[Vivek Khera]

>>>>> "DM" == Dmitry Mishchenko <[EMAIL PROTECTED]> writes:

DM> - second card has another real IP (lets say 65.1.1.2) and connected to
DM> Quintum VoIP box.

Personally, I would *never* put my quintum on a public IP even with a
firewall in front of it  i run mine inside a NAT'd LAN, and let
remote sites access it via the VPN, so they also have inside-the-LAN
IP numbers.


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Firewall for VoIP box

[Dmitry Mishchenko]

Hello,

We have the following configuration: FreeBSD router with 2 network cards.
- first card has a real IP and connected to internet (lets say 
65.1.1.1). There is a  ipfw firewall which control traffic to this card.
- second card has another real IP (lets say 65.1.1.2) and connected to 
Quintum VoIP box.

VoIP box has also its own real IP (lets say 65.1.1.3).

How should firewall be configured for card #2 for allowing VoIP traffic?
What ports should be open for normal VoIP work?
Thanks,
Dmitry
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