I asked this before, but never got any response, so hope someone can help me this time. Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls? I plan on installing FreeSwitch on a Mac and it would be really handy to be able to use Apple's USB modem in this way as otherwise there's no way to add such PSTN functionality on the Mac Mini. Although my requirement is Mac based, I guess this question applies to any FreeSwitch installation. Ken G i l l e t t _/_/_/_/_/_/_/_/
No, at the moment is not possible. -giovanni On 8/8/10, Ken Gillett <ken@ukgb.net> wrote:
I asked this before, but never got any response, so hope someone can help me this time.
Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls?
I plan on installing FreeSwitch on a Mac and it would be really handy to be able to use Apple's USB modem in this way as otherwise there's no way to add such PSTN functionality on the Mac Mini. Although my requirement is Mac based, I guess this question applies to any FreeSwitch installation.
Ken G i l l e t t
_/_/_/_/_/_/_/_/
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You can always use a standalone sip gateway (patton, audiocodes, etc.). On 8/8/10, Ken Gillett <ken@ukgb.net> wrote:
I asked this before, but never got any response, so hope someone can help me this time.
Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls?
I plan on installing FreeSwitch on a Mac and it would be really handy to be able to use Apple's USB modem in this way as otherwise there's no way to add such PSTN functionality on the Mac Mini. Although my requirement is Mac based, I guess this question applies to any FreeSwitch installation.
Ken G i l l e t t
_/_/_/_/_/_/_/_/
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Well that's the alternative, but I confess to being unclear on one aspect of such devices. In a simple scenario of several SIP devices registered to a SIP server plus a PSTN analog line, it is relatively simple to see how PSTN to VOIP bridging can be implemented with a gateway device that includes an FXS, FXO and ethernet port. An incoming call on the PSTN causes the gateway to initiate a call via the SIP server to SIP client devices registered on that ID and then putting the 2 ends of the call together so the PSTN call is 'bridged' onto VOIP and being handled by a SIP client device, i.e. a softphone or SIP telephone. But what about the reverse? My requirement is not only for the above basic ATA, but to be able to bridge VOIP to PSTN. IOW so that a SIP client can make an outgoing call onto the PSTN. Both the gateway and the SIP client are registered to a SIP server (same one in fact), how does the SIP client place a call onto the PSTN? The gateway is not a SIP server, what is the process by which a SIP client can tell such a gateway to place a call on the PSTN to the number that is somehow provided to the gateway by the SIP client? I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation. I'd be grateful for any advice about this. Thanks. On 8 Aug 2010, at 20:49, Tony Graziano wrote:
You can always use a standalone sip gateway (patton, audiocodes, etc.).
On 8/8/10, Ken Gillett <ken@ukgb.net> wrote:
I asked this before, but never got any response, so hope someone can help me this time.
Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls?
I plan on installing FreeSwitch on a Mac and it would be really handy to be able to use Apple's USB modem in this way as otherwise there's no way to add such PSTN functionality on the Mac Mini. Although my requirement is Mac based, I guess this question applies to any FreeSwitch installation.
Ken G i l l e t t _/_/_/_/_/_/_/_/
Check out the Linksys SPA-3102 -MC On Mon, Aug 9, 2010 at 11:49 AM, Ken Gillett <ken@ukgb.net> wrote:
Well that's the alternative, but I confess to being unclear on one aspect of such devices.
In a simple scenario of several SIP devices registered to a SIP server plus a PSTN analog line, it is relatively simple to see how PSTN to VOIP bridging can be implemented with a gateway device that includes an FXS, FXO and ethernet port. An incoming call on the PSTN causes the gateway to initiate a call via the SIP server to SIP client devices registered on that ID and then putting the 2 ends of the call together so the PSTN call is 'bridged' onto VOIP and being handled by a SIP client device, i.e. a softphone or SIP telephone.
But what about the reverse? My requirement is not only for the above basic ATA, but to be able to bridge VOIP to PSTN. IOW so that a SIP client can make an outgoing call onto the PSTN. Both the gateway and the SIP client are registered to a SIP server (same one in fact), how does the SIP client place a call onto the PSTN? The gateway is not a SIP server, what is the process by which a SIP client can tell such a gateway to place a call on the PSTN to the number that is somehow provided to the gateway by the SIP client?
I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation.
I'd be grateful for any advice about this. Thanks.
On 8 Aug 2010, at 20:49, Tony Graziano wrote:
You can always use a standalone sip gateway (patton, audiocodes, etc.).
On 8/8/10, Ken Gillett <ken@ukgb.net> wrote:
I asked this before, but never got any response, so hope someone can help me this time.
Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls?
I plan on installing FreeSwitch on a Mac and it would be really handy to be able to use Apple's USB modem in this way as otherwise there's no way to add such PSTN functionality on the Mac Mini. Although my requirement is Mac based, I guess this question applies to any FreeSwitch installation.
Ken G i l l e t t
_/_/_/_/_/_/_/_/
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Am 10.08.2010 02:39, schrieb Michael Collins:
Check out the Linksys SPA-3102 -MC
Do you know if this device forward the caller id (respectively caller phone number) transferred from the PSTN to the SIP-switch (FreeSwitch). Normally it should, because it may be necessary for routing. Kind Regards, Holger
On Mon, Sep 20, 2010 at 1:50 AM, Holger von Rhein <lortas@freenet.de> wrote:
Am 10.08.2010 02:39, schrieb Michael Collins:
Check out the Linksys SPA-3102 -MC
Do you know if this device forward the caller id (respectively caller phone number) transferred from the PSTN to the SIP-switch (FreeSwitch). Normally it should, because it may be necessary for routing.
In my experience it does forward the CID information - assuming that the phone line has caller ID capabilities. -MC
I'm pretty sure the zoom does not support sip originated calls to the FXO port. It's FXO port is strictly used as failover or selectable via dialplan when the call originates from the FXS port (eg: dial 9 first to get FXO). Try: cisco 3102 audiocodes grandstream for atas that support full FXS/FXO operation. On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett <ken@ukgb.net> wrote:
I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation.
-- -Rupa
Well I did try a Linksys/Cisco/Sipura 3102 but it's a configuration nightmare. The problem mainly is as I said, I cannot figure how VOIP -> PSTN bridging can be achieved whatever the gateway device. A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the call. From that piece of data it gets the SIP domain and the ID of the user. So how is the gateway able to be inserted in this process. The SIP client has no knowledge of this device and no way to include that third piece of data into its process. Obviously such gateways do exist, but I do not yet understand how the process works. Some advice on the basic process flow would therefore assist me to set up what I need irrespective of what devices I am using. Sorry to be so ignorant about this, but anyone able and willing to help with an explanation? On 10 Aug 2010, at 04:23, Rupa Schomaker wrote:
I'm pretty sure the zoom does not support sip originated calls to the FXO port. It's FXO port is strictly used as failover or selectable via dialplan when the call originates from the FXS port (eg: dial 9 first to get FXO).
Try:
cisco 3102 audiocodes grandstream
for atas that support full FXS/FXO operation.
On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett <ken@ukgb.net> wrote: I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation.
Ken G i l l e t t _/_/_/_/_/_/_/_/
http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo On Tue, Aug 10, 2010 at 2:17 AM, Ken Gillett <ken@ukgb.net> wrote:
Well I did try a Linksys/Cisco/Sipura 3102 but it's a configuration nightmare. The problem mainly is as I said, I cannot figure how VOIP -> PSTN bridging can be achieved whatever the gateway device.
A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the call. From that piece of data it gets the SIP domain and the ID of the user. So how is the gateway able to be inserted in this process. The SIP client has no knowledge of this device and no way to include that third piece of data into its process.
Obviously such gateways do exist, but I do not yet understand how the process works. Some advice on the basic process flow would therefore assist me to set up what I need irrespective of what devices I am using. Sorry to be so ignorant about this, but anyone able and willing to help with an explanation?
On 10 Aug 2010, at 04:23, Rupa Schomaker wrote:
I'm pretty sure the zoom does not support sip originated calls to the FXO port. It's FXO port is strictly used as failover or selectable via dialplan when the call originates from the FXS port (eg: dial 9 first to get FXO).
Try:
cisco 3102 audiocodes grandstream
for atas that support full FXS/FXO operation.
On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett <ken@ukgb.net> wrote: I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation.
Ken G i l l e t t
_/_/_/_/_/_/_/_/
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-- -Rupa
That is very interesting and I will almost certainly be trying it out with an SPA3102, but I still don't have a complete grasp of VOIP -> PSTN bridging, whatever the devices. A SIP client normally makes a call by contacting a SIP server somewhere and telling it the number to be called. But how can it do this when it needs to be done through a gateway onto PSTN? The gateway is not a SIP server of any sort, just another client, so how can the calling SIP client/device contact the gateway and say "call this number"? It is this basic process flow that I want to get my head around. Once I have that basic understanding I am in a better position to try and configure the various devices. Without that, I might as well be pushing buttons at random. So I'd appreciate it if someone could please explain this process and enlighten me. On 10 Aug 2010, at 13:09, Rupa Schomaker wrote:
http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo
On Tue, Aug 10, 2010 at 2:17 AM, Ken Gillett <ken@ukgb.net> wrote: Well I did try a Linksys/Cisco/Sipura 3102 but it's a configuration nightmare. The problem mainly is as I said, I cannot figure how VOIP -> PSTN bridging can be achieved whatever the gateway device.
A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the call. From that piece of data it gets the SIP domain and the ID of the user. So how is the gateway able to be inserted in this process. The SIP client has no knowledge of this device and no way to include that third piece of data into its process.
Obviously such gateways do exist, but I do not yet understand how the process works. Some advice on the basic process flow would therefore assist me to set up what I need irrespective of what devices I am using. Sorry to be so ignorant about this, but anyone able and willing to help with an explanation?
On 10 Aug 2010, at 04:23, Rupa Schomaker wrote:
I'm pretty sure the zoom does not support sip originated calls to the FXO port. It's FXO port is strictly used as failover or selectable via dialplan when the call originates from the FXS port (eg: dial 9 first to get FXO).
Try:
cisco 3102 audiocodes grandstream
for atas that support full FXS/FXO operation.
On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett <ken@ukgb.net> wrote: I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation.
Ken G i l l e t t _/_/_/_/_/_/_/_/
The problem mainly is as I said, I cannot figure how VOIP -> PSTN bridging can be achieved whatever the gateway device.
A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the call. From that piece of data it gets the SIP domain and the ID of the user. So how is the gateway able to be inserted in this process. The SIP client has no knowledge of this device and no way to include that third piece of data into its process.
Obviously such gateways do exist, but I do not yet understand how the process works. Some advice on the basic process flow would therefore assist me to set up what I need irrespective of what devices I am using. Sorry to be so ignorant about this, but anyone able and willing to help with an explanation?
Here's a simple answer: Usually there is some sort of intermediary to help with routing, but purely from the gateway's perspective: The device listens for SIP signalling A call comes in to the domain or IP address of the gateway with the destination number as the user part of the SIP uri (the part before the @ sign). The gateway interprets the user part as the PSTN number, subject to any internal rules it has about adding or removing prefixes, etc. The gateway then picks up the line and places the call, sending back SIP messages as to the status. Here's a link that brings up a PDF document explaining SIP to PSTN call flow. http://www.eventhelix.com/EventStudio/telecom_message_sequence_charts/SIP_PS...
Ken Gillett wrote:
Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls? From a hardware point of view, I believe this is possible. For instance, take a closer look at some DSL modem/router and a VoIP Gateway/Router that is based on a TI AR7 platform. Chances are their hardware designs are the same and also use the same (Legerity) chipset, yet their functions are different due to their drivers and software applications. That said, if you can manage to find the source code for your hardware and are willing to invest your time on working on it, I am sure you will be able to succeed to make your DSL modem acting as a VoIP device.
----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Modems-tp5386739p5386969.html Sent from the freeswitch-users mailing list archive at Nabble.com.
On 08/09/2010 01:46 AM, Ken Gillett wrote:
I asked this before, but never got any response, so hope someone can help me this time.
Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls?
I plan on installing FreeSwitch on a Mac and it would be really handy to be able to use Apple's USB modem in this way as otherwise there's no way to add such PSTN functionality on the Mac Mini. Although my requirement is Mac based, I guess this question applies to any FreeSwitch installation.
Ken G i l l e t t People constantly ask this same question, but nobody wants to write the drivers. It shouldn't be that hard. There are packages to use many winmodems as modems on Linux machines. The DSP code is usually supplied as a binary object, but the interface to the hardware is supplied as source code. Modify that source, and you could turn it into, say, a DAHDI driver that would hook the device into Asterisk, Callweaver or Freeswitch.
Steve
participants (9)
-
Giovanni Maruzzelli
-
Holger von Rhein
-
Ken Gillett
-
mazilo
-
Michael Collins
-
Rupa Schomaker
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Steve Underwood
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Tim St. Pierre
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Tony Graziano