Hello Joe,
You will need a SIP trunk on each 6355 in order to send sip calls back and forth between locations.
Per your example this unit would route the dialed number "20" out the SIP trunk to the remote 6355. The accept 20 on the voice-grouped trunk allows the number out the trunk.
If a call from the PBX is sent to the ADTRAN via the FXO port, the ADTRAN can ring a voice user, ring-group or send the call out SIP depending on the "trunk-number" configured on the FXO voice trunk. For example, if you wanted "257" to ring an analog FXS user, then you would configure the trunk-number 257 on the voice trunk for the FXO. You would also need an FXS user to be called 257 or have a sip-identity of 257 on a voice user so the 6355 can properly route the call. You may or may not need blind-dial on the FXO voice trunk as well.
Just to show you what you can do, I added a grouped-trunk for the FXO port. Here the config would allow 911 to be switched out the FXS port to the PBX. The "$" with a high cost would also route calls to the FXO port if the sip trunk is down.
Here is a partial config:
Main - 192.168.10.1
voice trunk T01 type analog supervision loop-start
description "FXO Line"
did digits-transferred 4
trunk-number 257
connect fxo 0/1
!
voice trunk T02 type sip
description "SIP to Remote"
sip-server primary 192.168.20.1
!
voice grouped-trunk "FXO_to_PBX"
trunk T01
accept 911 cost 0
accept $ cost 100
!
voice grouped-trunk "SIP_to_Remote"
trunk T02
accept 20 cost 0
!
voice user 257
connect fxs 0/1
password "1234"
!
voice user 30
connect fxs 0/2
password "1234"
!
As an example I used "30" as a number that may be dialed from this site, destined to the Main location.
Remote - 192.168.20.1
voice trunk T01 type sip
description "SIP to Main"
sip-server primary 192.168.10.1
!
voice grouped-trunk "SIP_to_Main"
trunk T01
accept 30 cost 0
!
voice user 20
connect fxs 0/1
password "1234"
!
Again, this sample config will allow you to dial "30" from the Remote side and ring the analog user on FXS 0/2 at the Main site. You can dial 20 from the Main site and ring the analog user on FXS 0/1 at the Remote site. Also, if the PBX calls into the FXO port at the Main site, the ADTRAN will ring the analog user at 0/1.
Let me know if you have any questions.
Regards,
Geoff
Hello and thank you for posting to our forum.
Yes, sip does have to be globally enabled. While I am not quite sure the exact application from the Cisco config, maybe one of our other users will. In the meantime, can you tell me what you want to do with the FXS and FXO ports on the 6355? For example, do you want to use the FXS ports to dial out towards a SIP Trunk and use the FXO as a backup?
Also, I am going to reference an older post regarding FXS/FXO functionality for a similar product. Although the ADTRAN product is different, the config is applicable.
Regards,
Geoff
Geo,
On one of the 6355's I need to use the FXO ports for a Panasonic PBX to connect to. The Key system users dial 257 or 258 to get dial tone on a trunk and then from the 6355 main site if they dial 20 go to one FXS port at a remote office running a 6355, the main office router address is 192.168.10.1 and the remote office is 192.168.20.1. The offices are in a private VLAN given to us by the telco so we have no need for the firewall to restrict any access at this time.
Thanks,
Joe
Hello Joe,
You will need a SIP trunk on each 6355 in order to send sip calls back and forth between locations.
Per your example this unit would route the dialed number "20" out the SIP trunk to the remote 6355. The accept 20 on the voice-grouped trunk allows the number out the trunk.
If a call from the PBX is sent to the ADTRAN via the FXO port, the ADTRAN can ring a voice user, ring-group or send the call out SIP depending on the "trunk-number" configured on the FXO voice trunk. For example, if you wanted "257" to ring an analog FXS user, then you would configure the trunk-number 257 on the voice trunk for the FXO. You would also need an FXS user to be called 257 or have a sip-identity of 257 on a voice user so the 6355 can properly route the call. You may or may not need blind-dial on the FXO voice trunk as well.
Just to show you what you can do, I added a grouped-trunk for the FXO port. Here the config would allow 911 to be switched out the FXS port to the PBX. The "$" with a high cost would also route calls to the FXO port if the sip trunk is down.
Here is a partial config:
Main - 192.168.10.1
voice trunk T01 type analog supervision loop-start
description "FXO Line"
did digits-transferred 4
trunk-number 257
connect fxo 0/1
!
voice trunk T02 type sip
description "SIP to Remote"
sip-server primary 192.168.20.1
!
voice grouped-trunk "FXO_to_PBX"
trunk T01
accept 911 cost 0
accept $ cost 100
!
voice grouped-trunk "SIP_to_Remote"
trunk T02
accept 20 cost 0
!
voice user 257
connect fxs 0/1
password "1234"
!
voice user 30
connect fxs 0/2
password "1234"
!
As an example I used "30" as a number that may be dialed from this site, destined to the Main location.
Remote - 192.168.20.1
voice trunk T01 type sip
description "SIP to Main"
sip-server primary 192.168.10.1
!
voice grouped-trunk "SIP_to_Main"
trunk T01
accept 30 cost 0
!
voice user 20
connect fxs 0/1
password "1234"
!
Again, this sample config will allow you to dial "30" from the Remote side and ring the analog user on FXS 0/2 at the Main site. You can dial 20 from the Main site and ring the analog user on FXS 0/1 at the Remote site. Also, if the PBX calls into the FXO port at the Main site, the ADTRAN will ring the analog user at 0/1.
Let me know if you have any questions.
Regards,
Geoff
Geoff,
I am adding those suggestions and will do some testing tomorrow.
Thanks,
Joe
I am on site and working on this config today. I will have a sample soon. At the main site the PBX connects to FXO port on the cisco and I assume the same on the Adtran 6355. The remote sites use just FXS ports as they pick the line and get dial tone.
Hello Joe,
I was wondering if you were able to test this application out or get it up and running. Let me know if you have any questions.
Thanks,
Geoff
Hello Joe,
I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily. If you don't feel like the answer I marked was correct, feel free to come back to this post and unmark it and select another in its place with the applicable buttons. If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.
Thanks,
Geoff
Geo,
The Adtran solution did not work. We ended up having to use an Asterix PBX, so I would have to suggest if someone is using Cisco routers with FXO/FXS ports with a PBX they should just buy new Cisco gear and not use the FXP/FXS ports in the Adtran equipment.
Thanks,
Joe
Hello Joe,
I am sorry you were unable to get this working. Unless I am not understanding something about this application, it should work just the same with ADTRAN equipment. Did you collect any debugs when attempting to make this work or reach out to ADTRAN Technical Support? We would be happy to assist you with this if you would like to continue or will have the opportunity to try the ADTRAN equipment again.
Thanks,
Geoff