<div dir="ltr">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.<div>
<br></div><div style>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.</div>
<div style><br></div><div style>The complete SIP guide on a router can be found in the CVOICE book/course material. I highly recommend you read this book.</div><div style><br></div><div style><a href="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">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</a><br>
</div><div style><br></div><div style>Below are some of my personal notes on SIP on a voice gateway. </div><div style><pre style="color:rgb(0,0,0);word-wrap:break-word;white-space:pre-wrap">! 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</pre></div><div style><pre style="color:rgb(0,0,0);word-wrap:break-word;white-space:pre-wrap">! 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:<a href="http://192.168.3.1:5060">192.168.3.1:5060</a>
!
! 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:<a href="http://192.168.3.1:5060">192.168.3.1:5060</a>
! OR if you specified the server in the sip-ua section
session target sip-server
!</pre><pre style="color:rgb(0,0,0);word-wrap:break-word;white-space:pre-wrap"><pre style="word-wrap:break-word;white-space:pre-wrap">! 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
!</pre><pre style="word-wrap:break-word;white-space:pre-wrap"><span style="font-family:arial">! DTMF will be inband with the audio stream unless you say otherwise</span><br></pre><pre style="word-wrap:break-word;white-space:pre-wrap">
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
!</pre><pre style="word-wrap:break-word;white-space:pre-wrap"><pre style="word-wrap:break-word;white-space:pre-wrap">! Verifying SIP</pre><pre style="word-wrap:break-word;white-space:pre-wrap"><pre style="word-wrap:break-word;white-space:pre-wrap">
! ==============================================================================</pre><pre style="word-wrap:break-word;white-space:pre-wrap"><span style="font-family:arial">! Verify that the SIP feature is UP</span><br></pre>
show sip-ua service
! SIP Service is up
</pre><div style>! debug the SIP messages</div><div style>debug ccsip messages</div></pre></pre></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Sun, Mar 31, 2013 at 3:34 PM, Mike <span dir="ltr"><<a href="mailto:mikeeo@msn.com" target="_blank">mikeeo@msn.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Chris,<br>
<br>
I've done this several times. You will need CUBE licenses and a multi-flex<br>
T1 card (MFT), but you easily convert the SIP handoff to PRI.<br>
<div class="HOEnZb"><div class="h5"><br>
-----Original Message-----<br>
From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><br>
[mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of chris<br>
Sent: Sunday, March 31, 2013 9:28 AM<br>
To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
Subject: [cisco-voip] recommendations for handing off sip trunks as pri for<br>
legacy<br>
<br>
We have recently acquired a new location which has a legacy analog pbx and a<br>
carrier who was providing service as a PRI.<br>
<br>
The carrier has now stated they will no longer be supporting PRI and are<br>
recommending a switch to SIP trunks, which would be fine if they provided<br>
adtran or similar to handle the conversion.<br>
<br>
The carrier says they do not get involved in this situation and its up to<br>
the customer, so we are left to fend for ourselves :)<br>
<br>
This loction already has a 2851 with ipvoice, and I was thinking I should be<br>
able to do everything I need with that?<br>
<br>
Googling seems to turn up lots of configs which simply terminate local calls<br>
to a physically connected PRI, when in actuality I want to do the inverse<br>
and use the cisco to hand off a traditional PRI to the analog system.<br>
<br>
Anyone gone down this path before? Have any config examples of what I<br>
described that I can reference?<br>
<br>
Also what type of linecard would I need to handoff the PRI? Does it need to<br>
be a MFT T1 card or plain t1 dsu/csu?<br>
<br>
thanks in advance<br>
chris<br>
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