Re: VoIP and SSH
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 ___ 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 - something (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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED] ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP and SSH
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 - something (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? ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP and SSH
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP and SSH
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP and SSH
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] ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP and SSH
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 - something (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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP problems
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 both/leftvol[:rightvol] (currently 68:68)
rear both/leftvol[:rightvol] (currently 48:48)
rear.rec ON|OFF (currently OFF)
rec off
center both/leftvol[:rightvol] (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 monovol (currently 128)
ext.recordsrc MIC|LINE (currently MIC)
vmix0-src Fast|Low|Medium|High|High+|Production|OFF (currently
Fast)
vmix0-vol monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm5 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm6 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm7 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm8 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-in leftVU:rightVU] (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
_
{Beto|Norberto|Numard} Meijome
If you find a solution and become attached to it, the solution may become your
next problem.
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.
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Re: VoIP problems
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 both/leftvol[:rightvol] (currently 68:68)
rear both/leftvol[:rightvol] (currently 48:48)
rear.rec ON|OFF (currently OFF)
rec off
center both/leftvol[:rightvol] (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 monovol (currently 128)
ext.recordsrc MIC|LINE (currently MIC)
vmix0-src Fast|Low|Medium|High|High+|Production|OFF (currently
Fast)
vmix0-vol monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm5 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm6 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm7 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm8 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-in leftVU:rightVU] (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
_
{Beto|Norberto|Numard} Meijome
If you find a solution and become attached to it, the solution may become your
next problem.
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.
___
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Re: VoIP problems
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. ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-multimedia To unsubscribe, send any mail to [EMAIL PROTECTED] ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP problems
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 both/leftvol[:rightvol] (currently 68:68)
rear both/leftvol[:rightvol] (currently 48:48)
rear.rec ON|OFF (currently OFF)
center both/leftvol[:rightvol] (currently 48:48)
center.rec ON|OFF (currently OFF)
ext.spread ON|OFF (currently OFF)
ext.loopback ON|OFF (currently OFF)
ext.recordvol monovol (currently 128)
ext.recordsrc MIC|LINE (currently MIC)
vmix0-src Fast|Low|Medium|High|High+|Production|OFF (currently
Fast)
vmix0-vol monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm5 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm6 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm7 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-out.pcm8 monovol (currently 25.0 dB)
vmix0-out leftVU:rightVU] (currently 0:0)
vmix0-in leftVU:rightVU] (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
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP behind NAT and FreeBSD
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP behind NAT and FreeBSD
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP behind NAT and FreeBSD
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP behind NAT and FreeBSD
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED] ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP behind NAT and FreeBSD
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP behind NAT and FreeBSD
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP behind NAT and FreeBSD
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. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: Voip - solutions!
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: Voip - solutions!
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 ___ freebsd-questions@freebsd.org mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP: sip client
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 -- Ph: +34.666334818 web: www.locolomo.org S/MIME Certificate: http://www.locolomo.org/crt/2004071206.crt Subject ID: A9:76:7A:ED:06:95:2B:8D:48:97:CE:F2:3F:42:C8:F2:22:DE:4C:B9 Fingerprint: 4A:E8:63:38:46:F6:9A:5D:B4:DC:29:41:3F:62:D3:0A:73:25:67:C2 ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP: sip client
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. ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP World Leaders
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 ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VoIP World Leaders
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 -- Gary Kline [EMAIL PROTECTED] www.thought.org Public service Unix ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VOIP
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 -- Bill Moran Potential Technologies http://www.potentialtech.com ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VOIP
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 ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED] - Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis.Téléchargez GRATUITEMENT ici ! ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
Re: VOIP
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? -- Gustaaf ___ [EMAIL PROTECTED] mailing list http://lists.freebsd.org/mailman/listinfo/freebsd-questions To unsubscribe, send any mail to [EMAIL PROTECTED]
