[cisco-voip] recommendations for handing off sip trunks as pri for legacy

[Joel Perez]

Anthony put it more eloquently than i ever could have. All his points are
right on and those are the rules we try to follow as often as possible.

Joel P.


On Sun, Mar 31, 2013 at 10:43 PM, Anthony Holloway <
avholloway+cisco-voip at gmail.com> wrote:

> Unless I missed part of your requirements, I don't see where you'll need a
> CUBE license.  You are not doing IP-to-IP call legs (E.g., SIP-to-SIP or
> H323-to-SIP or H323-to-H323), but instead doing SIP-to-POTS.
>
> The config is very simple, but just like configuring a PRI for PSTN
> connectivity, you need to sync up your settings with the telco for SIP.
>  I.e., Call control ip address and port, proxy settings, expected diversion
> header, authentication mechanisms, early offer, etc.
>
> The complete SIP guide on a router can be found in the CVOICE book/course
> material.  I highly recommend you read this book.
>
>
> http://www.amazon.com/Implementing-Unified-Communications-Foundation-Learning/dp/1587204193/ref=sr_1_1?ie=UTF8&qid=1364782855&sr=8-1&keywords=cvoice+642
>
> Below are some of my personal notes on SIP on a voice gateway.
>
> ! SIP Gateway Configurations
> ! ==============================================================================
>
> ! SIP gateways have two things:
> ! 1. SIP-UA (Optionally the destination can be set at the DP level)
> ! 2. VoIP (SIP) Dial Peers
>
> ! To configure the SIP UA which contains:
> ! Authentication (Optional)
> ! SIP Servers (Registrar and Proxy)
> sip-ua
>  ! To specify a sip-server (you can also type this at the DP level)
>  ! This is how you configure the SIP proxy you point at
>  sip-server ipv4:192.168.3.1:5060
> !
>
> ! SIP VoIP Dial Peers have these two things:
> ! 1. Session Protocol defined as SIP
> ! 2. A session target pointing at the SIP UA
>
> ! The first thing you need to do is change the default call control protocol
> ! from H.323 to SIP, then you need to add a session target
> dial-peer voice 10 voip
>  session protocol sipv2
>  session target ipv4:192.168.3.1:5060
>  ! OR if you specified the server in the sip-ua section
>  session target sip-server
> !
>
> ! SIP uses UDP for outbound signaling by default, but you can change it if you
> ! want to or if you have to (ITSP standard)
> voice service voip
>  sip
>   session transport tcp
>   ! Or if you want to UDP it
>   ! session transport udp
> !
>
> ! Or at the DP level
> dial-peer voice 10 voip
>  session transport tcp
> !
>
> ! To bind to a source address globally (only way)
> voice service voip
>  sip
>   bind control source-interface Loopback0
>   bind media source-interface Loopback0
>   ! Or bind both in a single command (think of this like a macro for the above two commands)
>   ! bind all source-interface Loopback0
> !
>
> ! The only way you can tune SIP timers is in SIP-UA mode
> sip-ua
>  ! To cut the default timer in half for how long an INVITE is valid
>  timers expires 90000
> !
>
> ! If you have an ISDN->SIP gateway, and you would like Calling Name display
> voice service voip
>  signaling forward unconditional
> !
> inteface Serial0/0/0:23
>  isdn supp-service name calling
> !
>
> ! If you wish to enable CLID privacy (aka blocking)
> dial-peer voice 10 voip
>  clid strip pi-restrict
> !
>
> ! If you wish to enable mapping the calling number into the display name field
> voice service voip
>  clid substitute name
> !
>
> ! DTMF will be inband with the audio stream unless you say otherwise
>
> dial-peer voice 10 voip
>  ! This will try to use SIP NOTIFY first, then RTP-NTE second and prevent
>  ! any DTMF from ever being inserted into the voice stream
>  dtmf-relay sip-notify rtp-nte digit-drop
> !
>
> ! Fax support is on by default and is cisco fax relay
> ! To change this to another type, you could
> voice service voip
>  ! This configures t38 followed by pass-through globally
>  fax protocol t38 fallback pass-through g711ulaw
>  ! You could also disable ECM
>  fax-relay ecm disable
>  ! Or even prevent SG3 faxing (downgrades to G3)
>  fax-relay sg3-to-g3
> !
>
> ! By default the fax rate is "voice" which means "as fast a possible"
> ! If you wanted to slow it down, you have to do it at the DP level
> dial-peer voice 10 voip
>  ! This slows down the transmission to 14.4kbps
>  fax rate 14400
>  ! If you want the dial-peer to take on the fax-relay config from global
>  fax-relay system
> !
>
> ! Modem passthrough is a breakthrough
> voice service voip
>  modem passthrough nse codec g711ulaw
> !
>
> ! If you want a dial-peer to use the global settings you need to...
> dial-peer voice 10 voip
>  modem passthrough system
> !
>
> ! Modem relay is better, and will auto fallback to passthrough.
> voice service voip
>  modem relay nse codec g711ulaw
> !
>
> ! If you want a dial-peer to use the global settings...oh wait, they can't
> dial-peer voice 10 voip
>  modem relay nse codec g711ulaw
> !
>
> ! If you're looking to change the codec of a voip call, you can do it globally
> ! with the added benefit of having a prioritized list...
> voice class codec 1
>  codec preference 1 g711ulaw
>  codec preference 2 g729r8
> !
> dial-peer voice 10 voip
>  voice-class codec 1
> !
>
> ! Or directly on the dial-peer, but at the cost of only a single codec
> dial-peer voice 10 voip
>  codec g711ulaw
> !
>
> ! For CUBE, you can actually just tell the DP to pass whatever it was given
> voice class codec 1
>  codec preference 1 transparent
>  codec preference 2 g711ulaw
>  codec preference 3 g729r8
> !
> dial-peer voice 10 voip
>  voice-class codec 1
> !
>
> ! Verifying SIP
>
> ! ==============================================================================
>
> ! Verify that the SIP feature is UP
>
> show sip-ua service ! SIP Service is up
>
> ! debug the SIP messages
> debug ccsip messages
>
>
>
> On Sun, Mar 31, 2013 at 3:34 PM, Mike <mikeeo at msn.com> wrote:
>
>> Chris,
>>
>> I've done this several times. You will need CUBE licenses and a multi-flex
>> T1 card (MFT), but you easily convert the SIP handoff to PRI.
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of chris
>> Sent: Sunday, March 31, 2013 9:28 AM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] recommendations for handing off sip trunks as pri
>> for
>> legacy
>>
>> We have recently acquired a new location which has a legacy analog pbx
>> and a
>> carrier who was providing service as a PRI.
>>
>> The carrier has now stated they will no longer be supporting PRI and are
>> recommending a switch to SIP trunks, which would be fine if they provided
>> adtran or similar to handle the conversion.
>>
>> The carrier says they do not get involved in this situation and its up to
>> the customer, so we are left to fend for ourselves :)
>>
>> This loction already has a 2851 with ipvoice, and I was thinking I should
>> be
>> able to do everything I need with that?
>>
>> Googling seems to turn up lots of configs which simply terminate local
>> calls
>> to a physically connected PRI, when in actuality I want to do the inverse
>> and use the cisco to hand off a traditional PRI to the analog system.
>>
>> Anyone gone down this path before? Have any config examples of what I
>> described that I can reference?
>>
>> Also what type of linecard would I need to handoff the PRI? Does it need
>> to
>> be a MFT T1 card or plain t1 dsu/csu?
>>
>> thanks in advance
>> chris
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>
>
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