For this first post on the Cisco UC I want to share you the configuration that made me sweat ! I have a customer that has a centralized UC infrastructure with only two CUCM, one UCCX and remote sites with 2901 gateway (IOS 15.4) and some 7821 IP phones. The PSTN is linked to the gateway with an E1 on the local sites and the gateway uses SIP Trunks to the CUCM. There is only one DID on the PSTN acces on each site for the call to the UCCX.
So my challange was to configure the SRST for SIP Phones and find a way to redirect all incoming calls to all the phones (hunt with broadcast) if the WAN link is down and phones registers to SRST. Easy no ? No. (and now if you want the solution jump to the diagram below)
Why ? Because if you configure a hunt group with the same number as the UCCX DN but with lower preference in dial-peer, the hunt group is still local to the gateway and will be preferred than the trunk to CUCM. Worst, if you configure the hunt group after the trunk it would work but after a reload of the gateway it doesn’t work anymore .
I could configure the gateway in MGCP that would be privilegied in normal mode with the hunt group for the SRST, but using MGCP for a PSTN gateway when SIP is almost used every where, I don’t think it’s the best idea ! So what ? Shared-line ? Not supported in SRST, if you try it you would have a dial-peer for each phones with this line and with the dial-peer hunting, the gateway still would select the same phone.
Almost every post on the Net about SRST + Hunt would say to use the magic alias command, but this works perfectly with… SCCP Phones ! And the alias on the voice register pool section seems not working as well. What are the remaining options ? Mhmm maybe a custom TCL script or using the EEM to create dynamically a hunt group when the CUCM Trunk dial-peer is busied-out… yuck !
And finally during a peaceful night the solution appeared, A Cisco gateway does voice routing (yes it seems that it was a strange night). So why not use the hunt group solution not only in SRST but always ? Let’s imagin a hunt-group that would send the call to the UCCX first and then if no answer or error (no WAN) send the call to all phones. And “all phones” is a second hunt group ! Yes it works ! YES !
Here is the solution :
It could explain more in details,but I think that almost everything is in there ! So here is the basic voice configuration of the gateway :
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version 15.4 ! hostname BASIQUEMENT ! card type e1 0 0 ! network-clock-participate wic 0 network-clock-select 1 E1 0/0/0 ! isdn switch-type primary-net5 ! voice-card 0 dspfarm dsp services dspfarm ! voice call send-alert voice call disc-pi-off voice call convert-discpi-to-prog ! voice service voip ip address trusted list ipv4 172.30.100.227 ipv4 172.30.100.228 allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw sip registrar server expires max 600 min 60 ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 ! voice register global mode srst timeouts interdigit 5 system message SRST Mode is Active max-dn 50 max-pool 10 timezone 26 ! voice register pool 1 id network 172.25.236.0 mask 255.255.255.0 dtmf-relay rtp-nte sip-kpml sip-notify voice-class codec 1 no vad ! voice hunt-group 1 sequential phone-display final 5204 list 5202,5203 timeout 20 pilot 5201 description MainToUCCXorSRST name MainToUCCXorSRST preference 3 secondary 9 ! voice hunt-group 2 parallel phone-display final 5201 list 5211,5212,5213,5214,5215,5216,5217,5218,5219,5220,5221,5222,5223,5224,5225 timeout 60 pilot 5204 description SRST-CallCenter name SRST-CallCenter preference 3 secondary 9 ! voice translation-rule 10 ! voice translation-rule 20 rule 3 /8888/ /5201/ ! voice translation-rule 30 rule 5 /^00\(.*\)/ /\1/ type unknown international plan unknown isdn rule 6 /^0\(.*\)/ /\1/ type unknown national plan unknown isdn ! voice translation-rule 40 ! voice translation-profile INCOMING-PRI translate calling 10 translate called 20 ! voice translation-profile OUTGOING-PRI translate calling 30 translate called 40 ! license udi pid CISCO2901/K9 sn SNSNSNSNSN call-history-mib retain-timer 500 call-history-mib max-size 500 hw-module pvdm 0/0 ! controller E1 0/0/0 pri-group timeslots 1-31 ! interface GigabitEthernet0/0 description Local Network ip address 172.25.236.253 255.255.255.0 duplex auto speed auto ! interface Serial0/0/0:15 description ISDN Access to PSTN no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn send-alerting isdn sending-complete no cdp enable ! ip route 0.0.0.0 0.0.0.0 172.25.236.1 ! control-plane ! voice-port 0/0/0:15 cptone TR bearer-cap Speech ! dial-peer voice 100 voip description Inbound from CUCM session protocol sipv2 incoming called-number 0T voice-class codec 1 dtmf-relay rtp-nte no vad ! dial-peer voice 110 voip description Outboud to CUCM Sub preference 5 destination-pattern 5... session protocol sipv2 session target ipv4:172.30.100.228 voice-class codec 1 voice-class sip options-keepalive dtmf-relay rtp-nte no vad ! dial-peer voice 120 voip description Outboud to CUCM Pub preference 6 destination-pattern 5... session protocol sipv2 session target ipv4:172.30.100.227 voice-class codec 1 voice-class sip options-keepalive dtmf-relay rtp-nte no vad ! dial-peer voice 200 pots description Outbound E1 PRI translation-profile outgoing OUTGOING-PRI destination-pattern 0T progress_ind alert enable 8 progress_ind progress enable 8 progress_ind connect enable 8 port 0/0/0:15 forward-digits all ! dial-peer voice 210 pots description Inbound E1 PRI translation-profile incoming INCOMING-PRI incoming called-number .... direct-inward-dial port 0/0/0:15 ! dial-peer voice 300 pots description Outbound E1 PRI Emergency SRST translation-profile outgoing OUTGOING-PRI destination-pattern 1.. port 0/0/0:15 forward-digits all ! sip-ua registrar ipv4:172.25.236.253 expires 600 ! call-manager-fallback secondary-dialtone 0 max-conferences 4 gain -6 transfer-system full-consult timeouts interdigit 4 ip source-address 172.25.236.253 port 2000 max-ephones 10 max-dn 10 dual-line system message primary SRST Mode is Active transfer-pattern T no huntstop call-forward pattern .T moh enable-g711 "flash0:music-on-hold.wav" time-format 24 date-format dd-mm-yy |
And that’s it, I can now really have a good peaceful night.
I enjoyed reading this post.
Thanks a lot.
This is an awesome solution. Thanks for posting
I see what you did there, nice work mate. Loving the diagram as well.