<div>Hi Rob,</div>
<div> </div>
<div>I am very interested in your experience of MS Speech Server integrated with CCM thru SIP trunk. My environment is CCM 4.0(2) + MGCP gateway, and I tried to configure the sip function working with Microsoft Office Communicator (LCS server). Is there any special configuration needed in MS server & gateway? I'm using SIP trunk pointing to LCS server, and the server configure the SIP gateway too. But right now both the inbound call and outbound call from/to Office Communicator is not working. I can see the missed call log in IP Phone but I didn't hear it ringing. Below is my gateway configuration, please kindly help to check is anything wrong. Thanks a lot ~
<br> </div>
<div>voice-card 1<br>!<br>ip subnet-zero<br>ip tcp synwait-time 13<br>!<br>!<br>ip cef<br>!<br>!<br>!<br>isdn switch-type primary-qsig<br>!<br>!<br>voice call send-alert<br>voice rtp send-recv<br>!<br>voice service voip<br>
sip<br> session transport tcp<br>!<br>!<br>ccm-manager fallback-mgcp<br>ccm-manager mgcp<br>ccm-manager music-on-hold<br>ccm-manager config server <a href="http://10.1.254.250">10.1.254.250</a><br>ccm-manager config<br>
!<br>!<br>controller T1 1/0<br> framing esf<br> crc-threshold 320<br> clock source internal<br> linecode b8zs<br> pri-group timeslots 1-24 service mgcp<br>!<br>!<br>!<br>interface FastEthernet0/1<br> ip address <a href="http://10.8.128.250">
10.8.128.250</a> <a href="http://255.255.255.0">255.255.255.0</a><br> ip route-cache flow<br> speed 100<br> full-duplex<br> no cdp enable<br>!<br>interface Serial1/0:23<br> no ip address<br> no logging event link-status<br>
isdn switch-type primary-qsig<br> isdn protocol-emulate network<br> isdn incoming-voice voice<br> isdn bind-l3 ccm-manager<br> no cdp enable<br>!<br>ip classless<br>ip route <a href="http://10.0.0.0">10.0.0.0</a> <a href="http://255.0.0.0">
255.0.0.0</a> <a href="http://10.8.128.1">10.8.128.1</a><br>!<br>!<br>voice-port 1/0:23<br> cptone TW<br>!<br>voice-port 3/0/0<br>!<br>voice-port 3/0/1<br>!<br>voice-port 3/1/0<br>!<br>voice-port 3/1/1<br>!<br>mgcp<br>mgcp call-agent CM 2427 service-type mgcp version
0.1<br>mgcp rtp unreachable timeout 1000 action notify<br>mgcp package-capability rtp-package<br>mgcp package-capability sst-package<br>no mgcp package-capability fxr-package<br>no mgcp timer receive-rtcp<br>mgcp sdp simple
<br>mgcp fax t38 inhibit<br>mgcp rtp payload-type g726r16 static<br>!<br>mgcp profile default<br>!<br>!<br>!<br>!<br>dial-peer voice 1 pots<br> application session<br> destination-pattern 1T<br> port 1/0:23<br>!<br>dial-peer voice 9 pots
<br> application session<br> destination-pattern 9T<br> port 1/0:23<br>!<br>dial-peer voice 424 voip<br> application session<br> destination-pattern 2....<br> session protocol sipv2<br> session target ipv4:<a href="http://10.1.254.100">
10.1.254.100</a><br> session transport tcp<br>!<br>dial-peer voice 2 pots<br> application mgcpapp<br> direct-inward-dial<br> port 1/0:23<br>!<br>sip-ua<br> retry invite 2<br> retry response 2<br> retry bye 2<br> retry cancel 2
<br> timers trying 100<br> sip-server ipv4:<a href="http://10.1.254.100">10.1.254.100</a></div>
<div> </div>
<div> </div>
<div> </div>
<div>Regards,</div>
<div> </div>
<div>Alan</div>
<div><br> </div>
<div><span class="gmail_quote">2006/4/18, Leetun, Rob <<a href="mailto:rleetun@co.boulder.co.us">rleetun@co.boulder.co.us</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div style="DIRECTION: ltr">
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2">Hi Wes,</font></span></div>
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2">True. I have a Microsoft Speech Server connected to our CM environment (4.1.3) through a SIP trunk. It works pretty nice. We are looking at IPICS, which from what I understand we will be using the SIP trunking as well with this product.
</font></span></div>
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font face="Arial" color="#0000ff" size="2">Rob</font></span></div><br>
<div lang="en-us" dir="ltr" align="left">
<hr>
<font face="Tahoma" size="2"><b>From:</b> Wes Sisk [mailto:<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:wsisk@cisco.com" target="_blank">wsisk@cisco.com</a>] <br><b>Sent:</b> Tuesday, April 18, 2006 7:44 AM
<br><b>To:</b> Alan Su<br><b>Cc:</b> Leetun, Rob; <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> Re: [cisco-voip] SIP Trunks config in CCM in relation to Vail Systems SIP TIM
<br></font><br> </div></div>
<div style="DIRECTION: ltr"><span class="e" id="q_10aad4571322fd90_1">
<div></div>Alan,<br><br>SIP trunk will only work to a proxy server. it will not work for endpoints. CM5.0 adds SIP lineside support for to allow direct registration of SIP clients and adds the ability for CM to function as a SIP back to back user agent.
<br><br>/Wes<br><br>Alan Su wrote:
<blockquote cite="http://mid8f9f19f00604170824u80b6dc5nee1320d9b18ef618@mail.gmail.com" type="cite">
<div>Hi Wes,</div>
<div> </div>
<div>I just have a question regarding CCM 4.0 intergrated with SIP, I see lots of document say that CCM5.0 is fully compatible with SIP, but can I do it in CCM 4.0 by using SIP trunk?</div>
<div>My customer is looking for a soultion that some of the users in organization will use SIP phone and others will use Cisco IP Phone, is it possible to deploy this solution in CCM 4.0? I have tried to configure CCM working with MS Office Communicator (act like SIP softphone), not worked well, I can see incoming call from Office Communicator but can't answer it.
</div>
<div> </div>
<div>Thanks so much.</div>
<div> </div>
<div> </div>
<div>Regards,</div>
<div> </div>
<div>Alan<br><br> </div>
<div>2006/4/3, Wes Sisk <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:wsisk@cisco.com" target="_blank">wsisk@cisco.com</a>>:
<blockquote class="gmail_quote">I have not used the Vail Systems SIP TIM but the CM SIP trunk is pretty<br>basic. Configure it with a destination address = IP address of your <br>destination device, set the port number (5060 by default), select TCP or
<br>UDP, check the box for MTP required (note default CM MTP resources are<br>g711 only.. you will need hardware DSP resources for g729)<br><br>then create a route pattern pointing to the SIP trunk. give it a<br>partition if desired. give your originating device a calling search
<br>space that includes that partition.<br><br>I have been testing the SIP trunk last few weeks and it seems to work well. <br><br>/Wes<br><br><br>Leetun, Rob wrote:<br>> Good afternoon,<br>><br>> Anyone have experience in the configuration of Callmanager
4.1.3 for a<br>> SIP trunk to connect to a Vail Systems SIP TIM system? I am trying to <br>> figure out what are the requirements for the configuration of the<br>> CallManager environment. I configured a SIP trunk, but wonder how it is
<br>> going to connect to the SIP TIM system through port 5060? <br>><br>> Thanks.<br>><br>> Rob<br>><br>> _______________________________________________<br>> cisco-voip mailing list<br>> <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:cisco-voip@puck.nether.net" target="_blank">
cisco-voip@puck.nether.net</a> <br>> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://portal.mxlogic.com/redir/?atXKnsvshKrjLtYQsFIcTpdAVPmEBCbdSaY3ivNU6U9GX33VkDa3JsJaBGBPdpb6XZuZQrFTjLsTsSoFaxk5mBiRundI6zBVYSyyM-OUeupKr73zobZ8Qg6BKQGmGncRAIqnjh0c1Emd409yT0HgQgmH1Sh4Qgiwq818i6Ph1a1EwzV-PBm1EwDkQg1Dp7Cy0obkDWJ3h0c1Emd4093hITfM-u0USyrjdFTpopvjuvhdFkVV" target="_blank">
https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>><br><br>_______________________________________________<br>cisco-voip mailing list<br><a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:cisco-voip@puck.nether.net" target="_blank">
cisco-voip@puck.nether.net</a><br><a onclick="return top.js.OpenExtLink(window,event,this)" href="http://portal.mxlogic.com/redir/?atXKnsvshKrjLtYQsFIcTpdAVPmEBCbdSaY3ivNU6U9GX33VkDa3JsJaBGBPdpb6XZuZQrFTjLsTsSoFaxk5mBiRundI6zBVYSyyM-OUeupKr73zobZ8Qg6BKQGmGncRAIqnjh0c1Emd409yT0HgQgmH1Sh4Qgiwq818i6Ph1a1EwzV-PBm1EwDkQg1Dp7Cy0obkDWJ3h0c1Emd4093hITfM-u0USCrjdFTpopvjuvhdFkVV" target="_blank">
https://puck.nether.net/mailman/listinfo/cisco-voip</a><br></blockquote></div><br></blockquote></span></div></blockquote></div><br>