I am trying to implement the Switchboard function that would work during SRST mode.
The configuration/requirement in moment (when CUCM is operating/reachable) :
External callers dial on the main number or main number + 0.
The Called Party Number is then translated to 111 (on the gateway), which is a Hunt Pilot in CUCM.
In SRST mode, the Hunt Pilot is not reachable and the main number or the main number + 0 or the 111 has to be translated to a DN which is configured on a Cisco 9971 Phone.
In our environment we have SCCP and SIP Phones. The forwarding is working with call-manager-fallback for the SCCP Phones (with alias). It is also working when using cme-srst (telephony service) for SCCP Phones, but we were never able to forward the call to the Switchboard phone (Cisco 9971).
The Cisco 9971 registers on the gateway in SRST and it can be reached when we dial the full number (main number + DN of the 9971). We tried to do a translation from 111 to the DN with alias in the voice register pool, but that did not work.
In our test lab, we have both SCCP and SIP phones, but it should work in an environment with only SIP Phones.
Here is the configuration of the gateway :
router01.cucm.lab#sh run Building configuration...
Current configuration : 7496 bytes ! ! Last configuration change at 11:04:32 UTC Thu Jun 5 2014 ! NVRAM config last updated at 13:48:09 UTC Tue Jun 3 2014 ! version 15.1 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname router01.cucm.lab ! boot-start-marker boot-end-marker ! ! logging buffered 10000 enable password cisco ! aaa new-model ! ! ! ! ! ! ! aaa session-id common network-clock-participate wic 0 ! no ipv6 cef ip source-route ip cef ! ! ! ip dhcp excluded-address 192.168.2.0 192.168.2.10 ip dhcp excluded-address 192.168.2.200 192.168.2.255 ! ip dhcp pool voip ! ! no ip domain lookup ip domain name cucm.lab ip name-server 192.168.2.3 multilink bundle-name authenticated ! ! ! ! isdn switch-type basic-net3 ! crypto pki token default removal timeout 0 ! crypto pki trustpoint TP-self-signed-432774010 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-432774010 revocation-check none ! ! voice-card 0 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service h225-notify cid-update fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none h323 modem passthrough nse codec g711ulaw sip bind control source-interface GigabitEthernet0/0 bind media source-interface GigabitEthernet0/0 registrar server expires max 600 min 60 ! voice class codec 10 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 ! voice class h323 1 h225 timeout tcp establish 3 ! ! voice register global max-dn 10 max-pool 1 ! voice register pool 1 translation-profile outgoing outgoingcallSRST id network 192.168.2.0 mask 255.255.255.0 preference 1 proxy 192.168.2.253 preference 1 dtmf-relay sip-notify voice-class codec 10 ! ! ! voice translation-rule 10 rule 1 /^$\(.*\)/ /111/ rule 2 /^0/ /111/ ! voice translation-rule 41 rule 1 /111/ /104/ ! ! voice translation-profile incomingcall translate called 10 ! voice translation-profile outgoingcallSRST translate called 41 ! ! license udi pid CISCO2911/K9 sn FCZ153820CM license accept end user agreement hw-module pvdm 0/0 ! ! ! archive log config hidekeys username cisco privilege 15 secret 5 $1$qzcD$xhvbO9lcVgSW.CMSGdRwz0 username V62 password 0 12345 ! redundancy ! ! ip ssh time-out 60 ip ssh authentication-retries 2 ! ! ! ! interface GigabitEthernet0/0 ip address 192.168.2.253 255.255.255.0 duplex auto speed auto ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface GigabitEthernet0/1.2 encapsulation dot1Q 2 ip address 184.108.40.206 255.255.254.0 ! interface GigabitEthernet0/1.32 description ### fuer ntp server ### encapsulation dot1Q 32 ip address 220.127.116.11 255.255.254.0 ! interface GigabitEthernet0/1.202 encapsulation dot1Q 202 ip address 18.104.22.168 255.255.254.0 ! interface GigabitEthernet0/1.208 description ### fuer ntp server ### encapsulation dot1Q 208 ip address 22.214.171.124 255.255.254.0 ! interface GigabitEthernet0/2 no ip address shutdown duplex auto speed auto ! interface BRI0/0/0 no ip address isdn switch-type basic-net3 isdn overlap-receiving T302 3000 isdn point-to-point-setup isdn incoming-voice voice isdn send-alerting isdn sending-complete isdn static-tei 0 ! interface BRI0/0/1 no ip address isdn switch-type basic-net3 isdn point-to-point-setup isdn incoming-voice voice ! ip forward-protocol nd ! ip http server ip http access-class 23 ip http authentication local no ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ! ! ! ! ! ! control-plane ! ! voice-port 0/0/0 translation-profile incoming incomingcall translation-profile outgoing outgoingcall no vad compand-type a-law cptone DE timeouts interdigit 5 bearer-cap Speech ! voice-port 0/0/1 ! ! mgcp fax t38 ecm ! ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! dial-peer voice 1100 voip description zum Callmanager-PUB preference 5 destination-pattern [1-9]T session target ipv4:192.168.2.240 voice-class codec 10 voice-class h323 1 dtmf-relay h245-alphanumeric fax-relay ecm disable ! dial-peer voice 2000 pots description *****Local Via BRI0/0/0***** preference 5 destination-pattern 0T progress_ind alert enable 8 progress_ind progress enable 8 progress_ind connect enable 8 fax rate disable direct-inward-dial port 0/0/0 ! ! ! ! gatekeeper shutdown ! ! telephony-service srst mode auto-provision all srst ephone template 5 srst ephone description srst fallback auto-provision phone : Jul 07 2005 17:45:08 : Jun 05 2014 08:37:20 srst dn line-mode dual max-ephones 5 max-dn 10 ip source-address 192.168.2.253 port 2000 max-redirect 20 system message "SRST Mode: Cisco Unified CME keepalive 10 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jun 05 2014 08:37:38 ! ! ephone-dn 1 dual-line number 105 label 105 description +43732316262105 name +43732316262105 ! ! ephone-dn 2 dual-line number 106 description 106 name 106 ! ! ephone-dn 3 dual-line number 101 label 101 description +43732316262101 name +43732316262101 ! ! ephone 1 description srst fallback auto-provision phone : Jul 07 2005 17:45:08 : Jun 05 2014 08:37:20 mac-address 5897.1E29.955F ephone-template 5 button 1:1 2:2 ! ! ! ephone 2 description srst fallback auto-provision phone : Jul 07 2005 17:45:08 : Jun 05 2014 08:37:20 mac-address A456.3040.3F2D ephone-template 5 button 1:3 ! ! ! line con 0 password cisco logging synchronous line aux 0 line vty 0 4 privilege level 15 password cisco transport input all line vty 5 15 access-class 23 in privilege level 15 password cisco transport input telnet ssh ! scheduler allocate 20000 1000 ntp master 3 end
Thanks in advance for your help, it is really appreciated (been battling with this for few days).
1 - To begin you need to review your voice translation-rules. For example, rule 40 is translating the call into a number you don't have and rule 10 is configured as a set that you're not using.
2 - You need to create a "voice hunt-group" for "111". Please see example below:
To create a hunt group for phones in a Cisco Unified Communications Manager Express (Cisco Unified CME) or Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) system, use the voice hunt-group command in global configuration mode. To delete a hunt group, use the no form of this command.
Unique sequence number that identifies the hunt group. Range is 1 to 100.
Allows an incoming call to go first to the number that has been idle the longest for the number of hops specified when the hunt group was defined. The longest-idle time is determined from the last time that a phone registered, reregistered, or went on-hook.
Allows an incoming call to simultaneously ring all the numbers in the hunt group member list.
Allows a round-robin selection of the first extension to ring. Ringing proceeds in a circular manner from left to right. The round-robin selection starts with the number left of the number that answered when the hunt-group was last called.
Allows an incoming call to ring all the numbers in the left-to-right order in which they were listed when the hunt group was defined.
3 - You configured srst with an ephone-template that doesn't exist!
4 - dial-peer voice 2000 pots is not going to send the "0". Is "0" your site code or is the carrier expecting to see the "0"?
5 - Why does "dial-peer voice 1100 voip" and "dial-peer voice 2000" have the same preference?
6 - voice register pool 1 doesn't have a number assigned...
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