FOOD FOR THOUGHT Hosted IncrediblePBX, ATA inside local network

[kengetz]

Please forgive me in advance if this is something that's easily locatable--I haven't been able to find an answer in my searching, and I feel that I must be missing something simple.

I've set a hosted installation of IncrediblePBX (using the amazing Digital Ocean service), and all is well. I'm testing, and things seem to work great. My current setup uses a local Raspberry PI running IncrediblePBX, inside my local network. I have a single PSTN line, connected to my network using an AudioCodes device (it's a standard ATA device). I can point this ATA at the hosted PBX, and it registers just fine, and I can call the analog phone hanging off the ATA device.

The problem is setting up and registering the ATA to handle the connection to the PSTN phone line. Inside my network, my PBX has a trunk setup with a connection to the ATA, using an internal IP address to register with the ATA. From the hosted PBX, however, I don't see how to tell the PBX the address of my ATA so that I can create a similar trunk there. Am I missing something simple? It seems that the hosted PBX would need to be able to communicate with the device inside my local/private network, and I'm guessing this requires some sort of port forwarding, which I've been trying to avoid.

Any suggestions appreciated! Thanks -- Ken

 

[briankelly63]

Most people running a PBX in a cloud will connect to VOIP providers for their service. Some are virtually free. If you get a local DID you can just forward your PSTN line to the VOIP DID. You can do what you want but its a tedious configuration. I use an OBI 110 as an ATA for a local POTS line. The ATA in this case registers with Asterisk rather then the other way around. In a normal trunk situation the PEER details would include the IP address of the provider. In that case Asterisk registers with the provider OR some providers just use your external IP address and there is no registration string. Search on setting up the OBI or go to there site to see how its configured to "register". It's not really a registration but more like and authorized SIP connection.

Having said all that I don't know anyone who's using a cloud PBX with a local POTS line as a trunk. Its also not really a trunk since it only handles one call at a time. The trunk you get from a VOIP provider is more like a highway to your PBX where lots of calls can go over different DID's.

Under PEER details in the trunk I have:

username=OBITRUNK1
type=friend
secret=xxxxxxx
disallow=all
allow=ulaw
qualify=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
insecure=very
context=from-trunk

 

[kengetz]

Although I appreciate the answer, I really don't understand what you're telling me, or what I can do to make this work, or if it's just not going to work. My ATA does register with the PBX, whether it's local or remote. That works fine.

Can you (or someone) point me at specific instructions on how to either make this work, or clarify that it's not possible? Thanks -- Ken

Most people running a PBX in a cloud will connect to VOIP providers for their service. Some are virtually free. If you get a local DID you can just forward your PSTN line to the VOIP DID. You can do what you want but its a tedious configuration. I use an OBI 110 as an ATA for a local POTS line. The ATA in this case registers with Asterisk rather then the other way around. In a normal trunk situation the PEER details would include the IP address of the provider. In that case Asterisk registers with the provider OR some providers just use your external IP address and there is no registration string. Search on setting up the OBI or go to there site to see how its configured to "register". It's not really a registration but more like and authorized SIP connection.

Having said all that I don't know anyone who's using a cloud PBX with a local POTS line as a trunk. Its also not really a trunk since it only handles one call at a time. The trunk you get from a VOIP provider is more like a highway to your PBX where lots of calls can go over different DID's.

Under PEER details in the trunk I have:

username=OBITRUNK1
type=friend
secret=xxxxxxx
disallow=all
allow=ulaw
qualify=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
insecure=very
context=from-trunk

 

[rjaiswal]

Please forgive me in advance if this is something that's easily locatable--I haven't been able to find an answer in my searching, and I feel that I must be missing something simple.

I've set a hosted installation of IncrediblePBX (using the amazing Digital Ocean service), and all is well. I'm testing, and things seem to work great. My current setup uses a local Raspberry PI running IncrediblePBX, inside my local network. I have a single PSTN line, connected to my network using an AudioCodes device (it's a standard ATA device). I can point this ATA at the hosted PBX, and it registers just fine, and I can call the analog phone hanging off the ATA device.

The problem is setting up and registering the ATA to handle the connection to the PSTN phone line. Inside my network, my PBX has a trunk setup with a connection to the ATA, using an internal IP address to register with the ATA. From the hosted PBX, however, I don't see how to tell the PBX the address of my ATA so that I can create a similar trunk there. Am I missing something simple? It seems that the hosted PBX would need to be able to communicate with the device inside my local/private network, and I'm guessing this requires some sort of port forwarding, which I've been trying to avoid.

Any suggestions appreciated! Thanks -- Ken

If I'm reading this correctly, you want to connect an ATA, which has a PSTN line connected to it, to your hosted PBX. To do this, you would need some sort of port forwarding. The most reliable way of doing this is to have a static IP where the ATA is located. Do you have one there?

 

[kengetz]

Thanks. Yes, absolutely. Fixed IP address locally. Just unclear what port(s) to forward, and how to indicate to the ATA and to the hosted PBX what's going on. Thanks!

 

[rjaiswal]

Thanks. Yes, absolutely. Fixed IP address locally. Just unclear what port(s) to forward, and how to indicate to the ATA and to the hosted PBX what's going on. Thanks!

I am assuming you are going to have VOIP handsets at the local location, as well, as the ATA. In that case, I would use a different SIP port, say 5070 for the ATA, and leave the handsets on 5060. My only concern is that I don't think you can limit what RTP ports a trunk can use in Asterisk... If you forward the RTP ports to the ATA, you'll get silence when you try to dial out using a handset.

I would try just forwarding the SIP port from the ATA and see if the firewall will NAT the RTP ports for you. If not, then you might need another static IP, just for the ATA so you can forward both SIP and RTP ports to the ATA.

 

[kengetz]

Thanks. Will try it. What are the RTP ports? And yes, we do have VOIP handsets at the local location. Looks like this isn't something people have really done, so perhaps it's best that I continue to use a local PBX for now? Thanks.

 

[rjaiswal]

Thanks. Will try it. What are the RTP ports? And yes, we do have VOIP handsets at the local location. Looks like this isn't something people have really done, so perhaps it's best that I continue to use a local PBX for now? Thanks.

RTP is the protocol that transmits audio between end points, trunks, and the pbx.

If you are unfamiliar with RTP and SIP, I would HIGHLY recommend that you keep your pbx local for the time being.

I would keep the hosted server up and running, so you can use that as a test / learning system. Once you get comfortable with the protocols that asterisk uses, I would then switch to a hosted system.

Also, I should have been more specific with my previous statement. You would need a static PUBLIC IP address for the ata. If the address is dynamic, even with a fairly quick DDNS service, your PSTN trunk could go down for a split second. And knowing "Murphy", it would go down at the worst possible time.

 

[kengetz]

Yep, public fixed IP address. I'm comfortable with SIP and with networking, just didn't know which specific ports I need to map for RTP. Still don't, I guess... Thanks!

 

[jake372]

kengetz did you ever get this up and running? I have a similar situation. Installed Incredible PBX at RentPBX but I am having one way audio issues with a softphone at my local install (not a static IP). I have read through just about everything I can find and the set up is either over my head OR when I follow the instructions to the letter, still isn't working.

 

[kengetz]

Hi. Did get it working, without even opening any ports in my router--the Obi110 ATA set up a connection and kept it open, so everything just worked. I was very impressed. I had no problems with quality. I finally realized that it wasn't worth it for me to pay the hosting charges--just easier and cheaper to maintain a Raspberry PI system at home (this was a system for two mailboxes, so hardly worth the cost). Can't tell you why it's not working for you, but there are many reasons why it might fail. The simplest is to check your bandwidth, and see if it's appropriate for VOIP (voiptest.8x8.com is a good place to start). But probably best to open a new thread on this.