Need a VoIP Guru - bonded T1s with reserved channels

A

archivalbackup

Gawd
Joined
Oct 12, 2007
Messages
643
I need some help parsing some configs to make sure this will work from a VoIP standpoint. I have a client that has two private T1's between the central office, and an office out of state. They want to bond the T1's into a single multilink to get better bandwidth for some heavy hitting database traffic, but they have their PBX tied into one of the T1 cards and I am not sure how to go about doing this on a bonded link. Here is the relevant configuration on the T1 card that has some voice channels carved out:

Code: 
controller T1 0/1/0
 framing esf
 linecode b8zs
 channel-group 0 timeslots 1-24
!
controller T1 0/1/1
 framing esf
 clock source internal
 linecode b8zs
 ds0-group 0 timeslots 1-6 type e&m-wink-start
!
!
interface Serial0/1/0:0
 bandwidth 1544
 ip address x.x.x.x 255.255.255.252
 ip directed-broadcast
 ip nbar protocol-discovery
 encapsulation ppp
 ip tcp header-compression iphc-format
 max-reserved-bandwidth 100
 service-policy output PtP-Edge
 ip rtp header-compression iphc-format
!
!
voice-port 0/1/1:0
!
dial-peer voice 200 voip
 max-conn 7
 destination-pattern 2...
 session target ipv4: x.x.x.x
 ip qos dscp cs3 signaling
!
dial-peer voice 100 pots
 destination-pattern 3...
 port 0/1/1:0
!

The other T1 Card (currently in another router for redundancy) is just a straight up T1 card
Code: 
interface Serial0/1/0
 description Point-to-Point T1 link to  
 bandwidth 1544
 ip address x.x.x.x 255.255.255.252
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip nbar protocol-discovery
 encapsulation ppp
 ip tcp header-compression iphc-format
 service-module t1 clock source internal
 service-module t1 timeslots 1-24
 max-reserved-bandwidth 100
 service-policy output PtP-Edge
 ip rtp header-compression iphc-format
!

The inventory display for the voice card:
Code: 
NAME: "Two port T1 voice interface daughtercard with Drop and Insert on Slot 0 SubSlot 1", DESCR: "Two port T1 voice interface daughtercard with Drop and Insert"
PID: VWIC-2MFT-T1-DI=  , VID: 1.0, SN:  

NAME: "PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 4", DESCR: "PVDMII DSP SIMM with one DSP"
PID: PVDM2-16          , VID: NA  , SN:

It seems to me that everything is being sent VoIP between the two sites, but they think there is some reserved channel setup with the phone company. I lack the VoIP depth to be sure.
 
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What type of PBX is it?

What is the trunk between the two offices? In a true VoIP situation, both of those T1's should act as data. If they have two T1's going inbetween the sites and one is voice, the other is data, they more than likely have a T1 voice trunk (not VoIP, aka DS1) tieing the two systems together. If it is a two data trunks, then all you need to worry about is QoS and making sure the data can traverse it. This is usually a SIP or H323 trunk.
 
randyc said:
What type of PBX is it?

What is the trunk between the two offices? In a true VoIP situation, both of those T1's should act as data. If they have two T1's going inbetween the sites and one is voice, the other is data, they more than likely have a T1 voice trunk (not VoIP, aka DS1) tieing the two systems together. If it is a two data trunks, then all you need to worry about is QoS and making sure the data can traverse it. This is usually a SIP or H323 trunk.
Click to expand...

The problem is that no one knows much, it is a Toshiba DX system.

From what I can tell, the PBX plugs into the dual-port T1 card at each site, then it is pure data in between, and then converted back at the other end. Both T1's carry data, no question about that, but they seemed to think they had channels reserved for voice which makes no sense because when the primary T1 is down and the traffic re-routes over the secondary T1 voice comes back online.

I just wanted a second set of eyes from someone that deals with VoIP regularly to make sure I am not missing something. The only thing in the config is a 'Dial Peer voip'.
 
Ok, the T1 card in question has this interesting feature

--
Drop and Insert

*

The Drop and Insert (D&I) feature allows DS0 timeslots to be taken off one T1 interface and inserted into time slots of the other T1 interface. This feature is available in VIC and WIC applications.
o

Drop and Insert functionality does not support different framing and line coding on the two ports. Therefore, when a tdm-group is configured on the controller T1 or E1, the framing type between the two controllers must be the same. This is only for the tdm-group functionality of the VWIC card.

Note: If you do configure two different framing types, this is the error message that the IOS sends to the console of the router:

Voice_Router (config)#connect TDM t1 0/1 t1 0/2 %CONN TDM:
Framing type mismatch %CONN TDM: Endpoints are incompatible
%CONN: Invalid Command

o

Drop and Insert timeslots do not need to be contiguous.
o

Drop and Insert of timeslots must be on the T1 controllers on the same 2-port VWIC, unless the gateway is Multiservice Interchange (MIX) enabled. When a gateway is MIX-enabled and the proper TDM network clock participation is configured, Drop and Insert of timeslots between T1 controllers on different VWICs is possible. Refer to Multiservice Interchange (MIX) for Cisco 3600 Series Multiservice Platforms for more details.
o

Drop and Insert uses tdm-groups.
o

Drop and Insert on a PRI can only be done if the entire PRI, including its D-channel is D&I'd. If individual channels must be D&I'd and other channel terminated, you cannot use PRI, you need to use CAS.
 
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I'm not 100% sure if this will be relevant, but its good to know: Can you confirm that there is a D channel on both the T1s (NFAS) ? or do they share D channels (FAS)?
 
I would double check with the account rep for the T1's...


actually, I would ask your account rep if (s)he has any notes on your setup.
 
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