Voice Over IP
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I have an AdTran 924E Total Access Integrated Access Device (IAD). Â I want to use it to take the PRI from my carrier (voice T1) and break it up into SIP trunks for my IP-PBX. Â I have been able to find many example configurations of taking in a SIP trunk from a SIP provider and delivering PRI connections to the PBX on the customer side, but I need to do the reverse:
PRI from carrier --> AdTran 924E --> Local LAN --> SIP Trunks to IP-PBX
Can anyone help with this device? Â I can't find any example configurations.
thanks!
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T1 type of a connection between the adtran and the IPPBX.
What timing, encapsulation does your IP-PBX support? What IP PBX is in use?
The Ad-Tran will have to accept those settings inbound. Â We have a 100 number DID block that is available through the PRI as well.
I know that on PRI handoff from carrier, I have read that we need to specify USER on the T1/PRI. Â Other than this, I am really not sure how to set this up.






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The dsx1(t1 5) is a point to point to your T1 interface on your IP PBX.
If your SIP connection established between the allworx and the remote, that all you need is the T1 for WAN and then the T1 to allworks.
PRI handoff to me means that the 924e handles everything with the provider it is terminating the WAN and the SIP trunk.
The DSX1 PRI handoff will mean that you will have 24 d channel interfaces that will be brought up as neede when calls comng in or going out.
Is. The 924e part of the provider's required equipment?

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The connection between the adtran dsx1 and a t1 port on the allworks will use the ni sip.
Check what dlci the frame has and whether you need IPSec tunnels through which the frame will go.
Usually the provider of the T1 will have the T1 frame configured.
What's better, giving pri to the allworx or giving sip to the allworx and the other pbx? Does  the delivery options change anything?






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Not sure whether you can use one of the four T1s as a PRI handoff to the other PBX.
You do not need to split the SIP trunk. Â You can split DID.
and then on the PBX's you will limit how many connection each can handle.
The 1-24 rule deals with how many SIP channels you are allocating on the connection.
You possible can use one of the four T1 to setup a PRI 2 to the second PBX
Why have multiple PBXs? you run into complexity of cross PBX communication.
It would be great to be able to hand-off to the PSTN on the bench, and that's what we're aiming for. Â For our own office, we only have one production PBX, the Allworx.
I think it would be best to have the incoming PRI hand off to the AdTran 924e. Â I want to configure it to serve SIP TRUNKS to the local LAN and allow the Allworx to use them. Â I'd also like OTHER PBX's on the LAN to be able to use the SIP trunks as well...and possibly, since it has come up, plug the DSX1 into the bench/lab PBX to test, if that PBX has PRI going into it. Â We usually can't test this until we're onsite at the customer premis.
Will this device be able to have that kind of configuration?
- Accept PRI in
- Create SIP trunks for use by misc. devices
- Create DSX1 port for PBX use (probably NOT the Allworx, but being able to use it would be great)
With this configuration, would I be able to :
- Set the Allworx to use the SIP trunks for outbound and set inbound calls to go here as well?
- Set the other PBXs in the office to use the SIP trunks for outbound calling? Â Would I also be able to map DID's to the other PBX that uses the SIP trunks?
I'd love an "A-Z manual" for the 924e but what I've found is how to configure it for certain situations without explanation. Â I guess I really want an "AdTran 924e unleashed" guide or something...in absence of that, a config to do what I'm asking would also be great!
As for support from AdTran, their gear is supposedly lifetime warranty, so I'm sure they'll talk to me if I called during business hours, but I haven't tried that yet.
You have many options. Â The 924e has 4 T1 inbound DSX1 (voice Port T1 to T1 on PBX) and 24 Analog ports
You need to check with them whether one of the four T1's can be used in a similar capacity as a DSX1 which seems possible.
You can provide them with a scenario and see what they say.
The difficulty I see in your situation is that you do not have all the necessary information from the current provider.
The example I provided for the interface configuration only deal with an integrated T1 from a Provider using esf, b8zs 24 channel.
The Eth0 port is the LAN side.
The DSX1 is the PRI handoff to a PBX
the first T1 and the Frame deal with Data channel on the T1 and VOIP/SIP trunk within the frame
The frame might have to go through an IPSEC tunnel.
On site, you may have to first establish/configure the T1 WAN /eth0 LAN.
You would then need to establish the ipsec connections if any.
You would then need to configure/setup the frame interface to establish the SIP trunk.
At this point you will handoff the PRI to the PBX.
Configure the PBX to handle the DIDs
and then see whether the calls flow through and back.

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Thanks!
not sure how you would overlay the T1 data with a frame otherwise.
The SIP data could be flowing through the IPSEC tunnel/s
the DSX1 is just another T1 interface.
in your case you should have a total of 5 T1 interfaces with the last one being the DSX1
You can use one of the four to be a PRI handoff to another PBX
You then should also have 24 fxs ports. (plenum to a patch block)
If you have the equipment in house to mimic a provider T1/SIP channel you can configure the backend PRI handoff to PBX.
The provider to WAN side, I think has to be done on site.
Voice Over IP
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Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN). Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).