On 12/13/19 9:28 PM, Greg Troxel wrote:> sean darcy <seandarcy2 at gmail.com> writes: > >> I'm moving asterisk to a laptop, so can't use the dahdi board. Is >> there any supported USB dahdi device ? I see the Sangoma USBfxo >> device, but the dahdi driver no longer supports it. Anything else ? > > There is also the ObiHai OBi202 with an OBiLine, which provides an FXO > port remoted over SIP. (I am not sure if this is discontinued.) >"FXO port remoted over SIP"? I have an analog phone system. I can use the obi202 to connect the system to asterisk ? That is, the 202 will connect a outgoing call from one of its phone ports to asterisk, and connect an incoming call from asterisk to that same phone port ? BTW, the 202 is one sale today at Amazon ! sean
sean darcy <seandarcy2 at gmail.com> writes:>> There is also the ObiHai OBi202 with an OBiLine, which provides an FXO >> port remoted over SIP. (I am not sure if this is discontinued.) > > "FXO port remoted over SIP"? > > I have an analog phone system. I can use the obi202 to connect the > system to asterisk ? That is, the 202 will connect a outgoing call > from one of its phone ports to asterisk, and connect an incoming call > from asterisk to that same phone port ?The obi202 is sort of a mini pbx itself. It has 4 slots for programming SIP service providers (ITSP and SP, split, but four of each). Then there are call routing strings you can program to tell it what to do. On mine, I have three Service Providers configured for SIP, all with ITSP profile A. Basically, PH (first FXS) has a sip login to asterisk via SP2, and a "primary service" of the SP2. So while I can dial **N to make it do other things (an obi thing), if I pick it up and dial my hello world extension it works. PH2 is similar but on SP3. Then, I have another sip login on SP1, for the FXO port, and an inbound call route to sp1(NNNNNNNNNN) - just to have a dialplan destination for incoming calls. When the POTS line rings, asterisk gets an INVITE and can route it back out to PH, or whatever. I have set the 'spoof callerid" flag on the obi, so that the callerid from POTS is passed to asterisk as the origin. On SP1, I have inbound call route set to "li", so that calls from asterisk go out the POTS line. I asked on the list earlier, I think, about the wisdom of separate SIP logins to logically separate these, vs trying to mulitplex based on digitstrings. I concluded that especially given that I had no need for more service providers in the obi, that I might as well assign SP1/2/3 to LI/PH/PH2, and be able to tell the obi "calls arriving on SP3 just go to ph2" rather than having to sort inbound calls by destination (which I suspect is doable). There is more complexity already inherent in the system than seems fun, so I am tending to reduce it when it doesn't cost me funtionality. I have encountered the dreaded spurious touch tone problem, in a way where they keep going. I believe this is due to confusion about signaling mode and talk-off, and have set the obi explicitly to RFC2833. That seems to work, but I'm not sure.> BTW, the 202 is one sale today at Amazon !Keep in mind that the 202 provides two FXS ports. You need the OBiLINE also (plugs into USB port on 202) to get an FXO port.
On 12/14/19 11:29 AM, Greg Troxel wrote:> sean darcy <seandarcy2 at gmail.com> writes: > >>> There is also the ObiHai OBi202 with an OBiLine, which provides an FXO >>> port remoted over SIP. (I am not sure if this is discontinued.) >> >> "FXO port remoted over SIP"? >> >> I have an analog phone system. I can use the obi202 to connect the >> system to asterisk ? That is, the 202 will connect a outgoing call >> from one of its phone ports to asterisk, and connect an incoming call >> from asterisk to that same phone port ? > > The obi202 is sort of a mini pbx itself. It has 4 slots for programming > SIP service providers (ITSP and SP, split, but four of each). Then > there are call routing strings you can program to tell it what to do. > > On mine, I have three Service Providers configured for SIP, all with > ITSP profile A. Basically, PH (first FXS) has a sip login to asterisk > via SP2, and a "primary service" of the SP2. So while I can dial **N to > make it do other things (an obi thing), if I pick it up and dial my > hello world extension it works. PH2 is similar but on SP3. > > Then, I have another sip login on SP1, for the FXO port, and an inbound > call route to sp1(NNNNNNNNNN) - just to have a dialplan destination for > incoming calls. When the POTS line rings, asterisk gets an INVITE and > can route it back out to PH, or whatever. I have set the 'spoof > callerid" flag on the obi, so that the callerid from POTS is passed to > asterisk as the origin. > > On SP1, I have inbound call route set to "li", so that calls from > asterisk go out the POTS line. > > I asked on the list earlier, I think, about the wisdom of separate SIP > logins to logically separate these, vs trying to mulitplex based on > digitstrings. I concluded that especially given that I had no need for > more service providers in the obi, that I might as well assign SP1/2/3 > to LI/PH/PH2, and be able to tell the obi "calls arriving on SP3 just go > to ph2" rather than having to sort inbound calls by destination (which I > suspect is doable). There is more complexity already inherent in the > system than seems fun, so I am tending to reduce it when it doesn't cost > me funtionality. > > I have encountered the dreaded spurious touch tone problem, in a way > where they keep going. I believe this is due to confusion about > signaling mode and talk-off, and have set the obi explicitly to RFC2833. > That seems to work, but I'm not sure. > >> BTW, the 202 is one sale today at Amazon ! > > Keep in mind that the 202 provides two FXS ports. You need the OBiLINE > also (plugs into USB port on 202) to get an FXO port.The FXS >This is spectacular. Thanks for all the info. But the dreaded FXO/FXS issue. It's like trying to remember linear algebra. If I'm plugging the line from analog phone system into the 202, which then routes it to asterisk, I'm plugging the line into an FXS port , correct? The line from the phone company is plugged into the FXO port. And since I'm not connecting the 202 to the phone company , I don't need one. Correct? Sigh. sean