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Route SIP Traffic using Dedicated SIP Trunk with Adtran Handoff
I need assistance with SIP Routing. We are deploying a new network for a client ( Both data and phone ) the current configuration is Public IP -- Cisco ASA5505 -- Cisco 2960 POE Switch ( VLANS on Switch) with Server 2012 R2 handling DHCP in an ADDS. So currently all traffic is using the ETH0 Default port for all traffic in and out.
Our carrier has provided a dedicated SIP trunk ( No Proxy ) using a small Adtran Device for handoff - So what we get is a RJ45 for the SIP traffic.
Constraint 1 - We must use the SwitchVox 80 for VOIP. This device has only a single Ethernet port to connect to the LAN.
Constraint 2 - Data connections are daisy-chained through Digium 50 VOIP Phones.
Thoughts for clarification and Input.-
Question 1 - how would we route the SIP using a Second WAN ETH1 port on the ASA 5505, Typically we would use a UTM device however with the cisco we have not locked down how to do this.
Question 2 - Is there an alternative, perhaps dropping a device behind the ASA such as a Peplink Balance 20?
Additional information - Has anyone overcome this caveat with the SwitchVox - We are a switchvox partner and their answer was ( Just plug it into something ) so obviously that was not a lot of help. How have others overcome this constraint?
Our carrier has provided a dedicated SIP trunk ( No Proxy ) using a small Adtran Device for handoff - So what we get is a RJ45 for the SIP traffic.
Constraint 1 - We must use the SwitchVox 80 for VOIP. This device has only a single Ethernet port to connect to the LAN.
Constraint 2 - Data connections are daisy-chained through Digium 50 VOIP Phones.
Thoughts for clarification and Input.-
Question 1 - how would we route the SIP using a Second WAN ETH1 port on the ASA 5505, Typically we would use a UTM device however with the cisco we have not locked down how to do this.
Question 2 - Is there an alternative, perhaps dropping a device behind the ASA such as a Peplink Balance 20?
Additional information - Has anyone overcome this caveat with the SwitchVox - We are a switchvox partner and their answer was ( Just plug it into something ) so obviously that was not a lot of help. How have others overcome this constraint?
ASKER
Thanks, yes that was my thought, so configuring the SwitchVox should I use the Adtrans Private IP ( Dedicated SIP ) as the gateway address, as I want all voice traffic out of that device.
Not necessarily -- You would probably be better adding a static route than changing the default gateway, as you will want to be able to get to Switchvox and other sites for updates.
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On the switchvox unit --
ASKER
I did not see that as an option in the SwitchVox GUI, IS it a CLI option?
I am sorry, but you should be able to SSH in and set it I think..
But I have not touched SwitchVox for a while so I think I will have to hope you get another response from someone who is supporting a current SV unit.
But I have not touched SwitchVox for a while so I think I will have to hope you get another response from someone who is supporting a current SV unit.
ASKER
Good Answer and the ultimate solution was a route that was anything going to XXX>XXX.XXX>XXX ( SIP Cloud ) to XXX.XXX.XXX.XXX ( Trunk)
Not sure I understand your concern, or problem with Switchvox's answer. If it's a dedicated, AKA no Internet presence, SIP connection than why not just put it into the 2960 on an appropriately configured VLAN and configure the SIP trunk and routing in the Switchvox correctly.