I thought you could only use trunk groups on analog...?<br><br><br>Jonathan<br><br><div><span class="gmail_quote">On 4/20/07, <b class="gmail_sendername">Robert Kulagowski</b> <<a href="mailto:bob@smalltime.com">bob@smalltime.com
</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Voll, Scott wrote:<br>> Can someone send me a sample config with 2 or 3 pri's in use for an H323
<br>> Voice gateway.<br><br>I like using trunk groups, especially if you have a bunch of PRIs and/or<br>analogs. That way, you don't need to individually have a dial-peer for<br>each port. Take a look at dial-peer 8011; it shows it pretty well.
<br>Dial peer 10 shows how you can force calls out analogs without having to<br>have a DP for each analog port.<br><br>IOS 12.4.9(T)<br><br>I also have this as a Word Document; at the top of the Document I have<br>reminders like {SITEPREFIX} {IPSUBNET}, etc. Then when it comes to
<br>deploy I can just do a search / replace and get the boilerplate out of<br>the way and concentrate on dial-peers.<br><br>no service pad<br>service tcp-keepalives-in<br>service tcp-keepalives-out<br>no service config<br>
service timestamps debug datetime msec localtime<br>service timestamps log datetime msec localtime<br>service password-encryption<br>!<br>hostname {SITEPREFIX}VGW01<br>card type t1 0 1<br>! Need to tell card whether it is an E1 or T1
<br>!<br>logging buffered 51200 debugging<br>!<br>aaa new-model<br>!<br>!<br>aaa authentication login default group radius local<br>aaa authentication login console-in local<br>!<br>clock timezone {TIMEZONE} {TZOFFSET}<br>
clock summer-time EDT recurring 2 Sun Mar 2:00 1 Sun Nov 2:00<br>network-clock-participate wic 1<br>network-clock-select 1 T1 0/1/0<br>!<br>voice-card 0<br> dsp tdm pooling<br> dsp services dspfarm<br> no local-bypass<br>
<br>! Use this so that individual ports don't have to be enumerated in<br>! the dial peers<br>trunk group FXO<br>max-retry 5<br> voice-class cause-code 1<br> hunt-scheme round-robin<br><br>! Used to differentiate if there are multiple PRIs at a site
<br>trunk group Local<br><br>trunk group LongDistance<br><br>voice class cause-code 1<br> no-circuit<br><br><br><br>!<br>no ip dhcp use vrf connected<br>! Used to release IP address back in to the pool<br>no ip dhcp conflict logging
<br>! Next command puts a copy of the dhcp database off-site<br>ip dhcp database tftp://1.1.1.1/{SITEPREFIX}VGW01-dhcp<br>ip dhcp excluded-address 1.1.{IPSUBNET}.1 1.1.{IPSUBNET}.20<br>! Polycom 7936's statically allocated downward from .19
<br>!<br>! Generic "phones" is good enough ; makes other documentation easier,<br>! like firmware update doc<br>ip dhcp pool Phones<br> network 1.1.{IPSUBNET}.0 <a href="http://255.255.255.0">255.255.255.0</a>
<br> option 150 ip <a href="http://1.1.1.10">1.1.1.10</a> <a href="http://1.1.23.10">1.1.23.10</a><br> default-router 1.1.{IPSUBNET}.1<br> lease 7<br>!<br>no ip bootp server<br>no ip domain lookup<br>no ftp-server write-enable
<br>isdn switch-type primary-ni<br>isdn voice-call-failure 0<br>!<br>voice call send-alert<br>voice call carrier capacity active<br>voice rtp send-recv<br>!<br>voice service voip<br> h323<br> call preserve limit-media-detection
<br>!<br>voice class codec 1<br>! TO select which codec to use<br> codec preference 1 g711ulaw<br> codec preference 2 g729r8<br>!<br>!<br>voice class h323 1<br>! If first call manager is not available timeout switch to 2nd
<br> h225 timeout tcp establish 5<br> call preserve<br>!<br>!<br>!<br>!<br>!<br>voice vad-time 65536<br>!<br>voice translation-rule {SITECODE}<br>! for rule 1, we're taking inbound calls that start with 89 and changing
<br>! it to 3489. (Site prefix is 34)<br> rule 1 /^89/ /3489/<br>! for rule 2, we're taking any inbound call that comes in as 4141 and<br>! changing it to the operator number<br> rule 2 /^4141/ /348900/<br>! for rule 3, we're taking inbound call to 5921 and sending it to
<br>! operator (800 number has DNIS of 5921)<br> rule 3 /^5921/ /348900/<br>!<br>!<br>voice translation-profile Prepend_Sitecode<br> translate called {SITECODE}<br>!<br>!<br>! For AAR and SRST<br>application<br> service rdnis_save flash:rdnis_save.tcl
<br> param restore_did 81234567890<br> !<br> global<br> service alternate default<br> !<br>!<br>!<br>controller T1 0/1/0<br> framing esf<br> linecode b8zs<br> pri-group timeslots 1-24<br> description CIRCUIT ID
<br>controller T1 0/2/0<br> framing esf<br> linecode b8zs<br> pri-group timeslots 1-24<br> description CIRCUIT ID<br>controller T1 0/2/1<br> framing esf<br> linecode b8zs<br> pri-group timeslots 1-24<br> description CIRCUIT ID
<br>!<br>!<br>interface Loopback0<br> description For music-on-hold<br> ip address <a href="http://6.6.6.6">6.6.6.6</a> <a href="http://255.255.255.255">255.255.255.255</a><br>!<br>interface GigabitEthernet0/0<br> description To {SITEPREFIX}VSW01 Port g1/0/1
<br> ip address 1.1.{IPSUBNET}.1 <a href="http://255.255.255.0">255.255.255.0</a><br> no ip unreachables<br> no ip proxy-arp<br> duplex auto<br> speed auto<br> no mop enabled<br>! Best Practices security<br> h323-gateway voip bind srcaddr
10.1.{IPSUBNET}.1<br>no shut<br>!<br>interface GigabitEthernet0/1<br> description To Production<br> ip address <a href="http://1.1.1.5">1.1.1.5</a> <a href="http://255.255.0.0">255.255.0.0</a><br> no ip unreachables<br>
no ip proxy-arp<br> duplex full<br> speed 100<br> no mop enabled<br>no shut<br>!<br>interface Serial0/1/0:23<br> no ip address<br> isdn switch-type primary-ni<br> isdn incoming-voice voice<br> isdn map address . plan isdn type national
<br> no cdp enable<br>! Set to appropriate trunk group; Local or LongDistance<br> trunk-group Local<br><br>interface Serial0/2/0:23<br> no ip address<br> isdn switch-type primary-ni<br> isdn incoming-voice voice<br> isdn map address . plan isdn type national
<br> no cdp enable<br>! Set to appropriate trunk group; Local or LongDistance<br> trunk-group Local<br><br>interface Serial0/2/1:23<br> no ip address<br> isdn switch-type primary-ni<br> isdn incoming-voice voice<br> isdn map address . plan isdn type national
<br> no cdp enable<br>! Set to appropriate trunk group; Local or LongDistance<br> trunk-group LongDistance<br><br><br>!<br>ip classless<br>ip route <a href="http://0.0.0.0">0.0.0.0</a> <a href="http://0.0.0.0">0.0.0.0</a>
1.1.{IPSUBNET}.1.1<br>!<br>no ip http server<br>ip http authentication local<br>!<br>!<br>! Dump syslog to off-site server<br>! Use the IP address of the voice interface, not the one facing<br>! production<br>logging source-interface GigabitEthernet0/0
<br>logging <a href="http://1.1.23.9">1.1.23.9</a><br>!<br>!<br>control-plane<br><br>voice-port 0/0/0<br> trunk-group FXO<br> timing hookflash-out 500<br>connection plar {OperatorNumber}<br>!<br>voice-port 0/0/1<br> trunk-group FXO
<br> timing hookflash-out 500<br>connection plar {OperatorNumber}<br><br>!<br>voice-port 0/0/2<br> trunk-group FXO<br> timing hookflash-out 500<br>connection plar {OperatorNumber}<br>!<br>voice-port 0/0/3<br> trunk-group FXO
<br> timing hookflash-out 500<br>connection plar {OperatorNumber}<br>!<br>!<br>voice-port 0/1/0:23<br> translation-profile incoming Prepend_Sitecode<br> input gain -3<br> echo-cancel coverage 24<br> echo-cancel erl worst-case 0
<br> playout-delay minimum low<br> no comfort-noise<br><br>voice-port 0/2/0:23<br> translation-profile incoming Prepend_Sitecode<br> input gain -3<br> echo-cancel coverage 24<br> echo-cancel erl worst-case 0<br> playout-delay minimum low
<br> no comfort-noise<br><br>voice-port 0/2/1:23<br> translation-profile incoming Prepend_Sitecode<br> input gain -3<br> echo-cancel coverage 24<br> echo-cancel erl worst-case 0<br> playout-delay minimum low<br> no comfort-noise
<br>!<br>sccp local GigabitEthernet0/0<br>sccp ccm <a href="http://10.1.23.10">10.1.23.10</a> identifier 3 version 4.1<br>sccp ccm <a href="http://10.1.1.12">10.1.1.12</a> identifier 2 version 4.1<br>sccp ccm <a href="http://10.1.1.11">
10.1.1.11</a> identifier 1 version 4.1<br>sccp<br>!<br>sccp ccm group 1<br>associate ccm 2 priority 2<br> associate ccm 1 priority 1<br> associate profile 10 register {SITEPREFIX}CFB01<br>!<br>dspfarm profile 10 conference
<br> codec g711ulaw<br> codec g711alaw<br> codec g729ar8<br> codec g729abr8<br> codec g729r8<br> codec g729br8<br>associate application SCCP<br>no shut<br>!<br>!<br>! Dial-peer sequence and numbering pattern<br>! For incoming, use sitecode + a number
<br><br>dial-peer voice {SITECODE}1 pots<br>description<br>incoming called-number {SITECODE}....<br> direct-inward-dial<br> port 0/1/0:15<br><br>dial-peer voice {SITECODE}2 pots<br>description<br>incoming called-number {SITECODE}....
<br> direct-inward-dial<br> port 0/2/0:15<br><br><br>! Dial-peer for voip is the IP address without dots. Use the right<br>! ones!<br>dial-peer voice 11111 voip<br> destination-pattern {SITECODE}....<br> progress_ind setup enable 3
<br> progress_ind progress enable 8<br> voice-class codec 1<br> voice-class h323 1<br> session target ipv4:<a href="http://1.1.1.11">1.1.1.11</a><br> dtmf-relay h245-alphanumeric<br> ip qos dscp cs3 signaling<br> no vad
<br> fax-relay ecm disable<br>!<br>dial-peer voice 111112 voip<br> preference 1<br>destination-pattern {SITECODE}....<br> progress_ind setup enable 3<br> progress_ind progress enable 8<br> voice-class codec 1<br> voice-class h323 1
<br> session target ipv4:<a href="http://1.1.1.12">1.1.1.12</a><br> dtmf-relay h245-alphanumeric<br> ip qos dscp cs3 signaling<br> no vad<br> fax-relay ecm disable<br>!<br>! Dial-peer for pots is "8" – as the escape code
<br>! Next is the number of digits that we're dealing with, so 7, 10, 11,<br>! etc<br>! Last is the preference number so analog can be less preferred than PRI<br>! with the same destination<br>! pattern<br><br>dial-peer voice 871 pots
<br>trunkgroup Local 1<br>trunkgroup FXO 3<br>description Local Calling pattern – use Local PRI first, then FXO if PRI<br>is full or busy<br>destination-pattern 8.......<br>clid restrict<br><br>dial-peer voice 8101 pots<br>
preference 1<br>trunkgroup LongDistance<br>description LD Calling pattern<br>destination-pattern 81..........<br>clid restrict<br>! LD calls out the LD PRI don't need a 1<br><br>dial-peer voice 8102 pots<br>! We only use this dial peer if the one above is down / full
<br>preference 2<br>trunkgroup Local 1<br>trunkgroup FXO 3<br>description LD Calling pattern – use Local PRI first, then FXO if PRI<br>is full or busy<br>destination-pattern 81..........<br>clid restrict<br>! LD calls out the local PRI or FXO need the "1" in front.
<br>prefix 1<br><br>dial-peer voice 8011 pots<br>! One dial-peer covers all of our bases.<br> trunkgroup LongDistance 1<br> trunkgroup Local 3<br> trunkgroup FXO 5<br> description International call<br> destination-pattern 8011T
<br> progress_ind setup enable 3<br> prefix 011<br><br>dial-peer voice 911 pots<br>description Used with CER<br> destination-pattern 6911<br> progress_ind setup enable 3<br> trunkgroup Local<br> forward-digits 3<br>
!<br>dial-peer voice 8411 pots<br>description x11 Services<br> destination-pattern 8[2-9]11<br> trunkgroup Local 1<br> trunkgroup FXO 3<br> forward-digits 3<br>!<br>dial-peer voice 9101 pots<br> description Used to send caller ID for inter-office calls
<br> destination-pattern 91[2-9].........<br> progress_ind setup enable 3<br>trunkgroup LongDistance<br>! Note that we don't have clid restrict on this dial peer, and it has a<br>! "9" - we stuff the 9 on the route pattern in CM; users can't dial "9"
<br>! directly<br> forward-digits 10<br><br>dial-peer voice 9011 pots<br> trunkgroup LongDistance<br> description Used to send caller ID for inter-office calls internationally<br> destination-pattern 9011T<br> progress_ind setup enable 3
<br> prefix 011<br><br>dial-peer voice 10 pots<br>description Different escape pattern to force a call to analog<br>trunkgroup FXO<br>destination-pattern *8T<br>!<br><br>! Convert to U.S.-based 911 for our foreign guests; do the same for
<br>! "110" and "112" in case of SRST mode<br>num-exp 911 8911<br>num-exp 110 8911<br>num-exp 8110 8911<br>num-exp 112 8911<br>num-exp 8112 8911<br><br><br>gateway<br> media-inactivity-criteria rtp<br>
timer receive-rtcp 1000<br>ip rtcp report interval 65535<br><br><br>!<br>line con 0<br> login authentication console-in<br>line vty 0 15<br> session-timeout 120<br> exec-timeout 60<br>autoselect during-login<br>!<br>
ntp server <a href="http://1.1.1.26">1.1.1.26</a><br>ntp update-calendar<br>! If there's a core dump, then send it via TFTP somewhere else for analysis<br>exception dump <a href="http://1.1.23.9">1.1.23.9</a><br><br>! archive command provides CLI history of who entered a command
<br>archive<br> log config<br> logging enable<br> notify syslog<br> hidekeys<br>end<br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net
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