CME as SRST Error: returning empty string !!!!

Hi All:

 

When I configure CME as SRST I get a constant scrolling error: returning empty string !!!!  

I cannot find one instance of this error anywhere on the internet. Has anyone ever seen the aforementioned error. Please see the SRST configurationand 'show ephone' output below. 

 

Thanks,

 

Ron

 

==================================================================================================

 

telephony-service

 srst mode auto-provision all

 srst dn line-mode octo

 max-ephones 10

 max-dn 10

 ip source-address 177.1.254.3 port 2000

 max-conferences 8 gain -6

 transfer-system full-consult

 secondary-dialtone 0

 create cnf-files version-stamp 7960 Oct 30 2013 16:31:32

!

!

ephone-dn  1  octo-line

 number 3002

 description 0207033002

 name 0207033002

!

!         

ephone-dn  2  octo-line

 number 3010

 description 0207033010

 name 0207033010

!

!

ephone-dn  3  octo-line

 number 3001

 description 0207033001

 name 0207033001

!

!

ephone  2

 device-security-mode none

 mac-address 001B.D4C6.9372

 button  1:1 2:2

!

!

!

ephone  6

 device-security-mode none

 mac-address D4A0.2A88.0613

 button  1:3 2:2

 

 

=============================================================================================

 

 

Branch2#show ephone

 

 

ephone-1[0] Mac:-711.-MTP. TCP socket:[19] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=1

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:4 

IP:177.1.254.3 61941 Unknown 83  keepalive 0 max_line 0 available_line 0

Preferred Codec: g711ulaw 

 

 

 

 

ephone-2[1] Mac:001B.D4C6.9372 TCP socket:[4] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 11/9 max_streams=0

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:1

IP:177.3.11.17 50124 Telecaster 7960  keepalive 6 max_line 6 available_line 4

button 1: dn 1  number 3002  CM Fallback CH1   IDLE         CH2   IDLE         CH3   IDLE         CH4   IDLE         CH5   IDLE         CH6   IDLE         CH7   IDLE         CH8   IDLE         

button 2: dn 2  number 3010  CM Fallback CH1   IDLE         CH2   IDLE         CH3   IDLE         CH4   IDLE         CH5   IDLE         CH6   IDLE         CH7   IDLE         CH8   IDLE         shared 

Preferred Codec: g711ulaw 

 

 

 

 

ephone-3[2] Mac:PHQ-.729-.MTP TCP socket:[5] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=1

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:5 

IP:177.1.254.1 57397 Unknown 83  keepalive 6 max_line 0 available_line 0

Preferred Codec: g711ulaw 

 

 

 

 

ephone-4[3] Mac:PHQ-.711-.MTP TCP socket:[6] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=1

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:4 

IP:177.1.254.1 34789 Unknown 83  keepalive 6 max_line 0 available_line 0

Preferred Codec: g711ulaw 

 

 

 

 

ephone-5[4] Mac:-729.-MTP. TCP socket:[13] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=1

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_se

 returning empty string !!!!nt:0 paging 0 debug:0 caps:0 

IP:177.1.254.2 26066 Unknown 83  keepalive 0 max_line 0 available_line 0

Preferred Codec: g711ulaw 

 

 

 

ephone-6[5] Mac:D4A0.2A88.0613 TCP socket:[10] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=5

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:12 privacy:1

IP:177.3.11.18 19486 7962  keepalive 4 max_line 3 available_line 3

button 1: dn 3  number 3001  CM Fallback CH1   IDLE         CH2   IDLE         CH3   IDLE         CH4   IDLE         CH5   IDLE         CH6   IDLE         CH7   IDLE         CH8   IDLE         

button 2: dn 2  number 3010  CM Fallback CH1   IDLE         CH2   IDLE         CH3   IDLE         CH4   IDLE         CH5   IDLE         CH6   IDLE         CH7   IDLE         CH8   IDLE         shared 

Preferred Codec: g711ulaw 

Comments

  • it sounds like something didn't initialize right, any chance you saved and reloaded already?


    On Wed, Oct 30, 2013 at 9:42 AM, rjferriolo <[email protected]> wrote:

    Hi All:

     

    When I configure CME as SRST I get a constant scrolling error: returning empty string !!!!  

    I cannot find one instance of this error anywhere on the internet. Has anyone ever seen the aforementioned error. Please see the SRST configurationand 'show ephone' output below. 

     

    Thanks,

     

    Ron

     

    ==================================================================================================

     

    telephony-service

     srst mode auto-provision all

     srst dn line-mode octo

     max-ephones 10

     max-dn 10

     ip source-address 177.1.254.3 port 2000

     max-conferences 8 gain -6

     transfer-system full-consult

     secondary-dialtone 0

     create cnf-files version-stamp 7960 Oct 30 2013 16:31:32

    !

    !

    ephone-dn  1  octo-line

     number 3002

     description 0207033002

     name 0207033002

    !

    !         

    ephone-dn  2  octo-line

     number 3010

     description 0207033010

     name 0207033010

    !

    !

    ephone-dn  3  octo-line

     number 3001

     description 0207033001

     name 0207033001

    !

    !

    ephone  2

     device-security-mode none

     mac-address 001B.D4C6.9372

     button  1:1 2:2

    !

    !

    !

    ephone  6

     device-security-mode none

     mac-address D4A0.2A88.0613

     button  1:3 2:2

     

     

    =============================================================================================

     

     

    Branch2#show ephone

     

     

    ephone-1[0] Mac:-711.-MTP. TCP socket:[19] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=1

    mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:4 

    IP:177.1.254.3 61941 Unknown 83  keepalive 0 max_line 0 available_line 0

    Preferred Codec: g711ulaw 

     

     

     

     

    ephone-2[1] Mac:001B.D4C6.9372 TCP socket:[4] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 11/9 max_streams=0

    mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:1

    IP:177.3.11.17 50124 Telecaster 7960  keepalive 6 max_line 6 available_line 4

    button 1: dn 1  number 3002  CM Fallback CH1   IDLE         CH2   IDLE         CH3   IDLE         CH4   IDLE         CH5   IDLE         CH6   IDLE         CH7   IDLE         CH8   IDLE         

    button 2: dn 2  number 3010  CM Fallback CH1   IDLE         CH2   IDLE         CH3   IDLE         CH4   IDLE         CH5   IDLE         CH6   IDLE         CH7   IDLE         CH8   IDLE         shared 

    Preferred Codec: g711ulaw 

     

     

     

     

    ephone-3[2] Mac:PHQ-.729-.MTP TCP socket:[5] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=1

    mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:5 

    IP:177.1.254.1 57397 Unknown 83  keepalive 6 max_line 0 available_line 0

    Preferred Codec: g711ulaw 

     

     

     

     

    ephone-4[3] Mac:PHQ-.711-.MTP TCP socket:Devil activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=1

    mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:4 

    IP:177.1.254.1 34789 Unknown 83  keepalive 6 max_line 0 available_line 0

    Preferred Codec: g711ulaw 

     

     

     

     

    ephone-5[4] Mac:-729.-MTP. TCP socket:[13] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=1

    mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_se

     returning empty string !!!!nt:0 paging 0 debug:0 caps:0 

    IP:177.1.254.2 26066 Unknown 83  keepalive 0 max_line 0 available_line 0

    Preferred Codec: g711ulaw 

     

     

     

    ephone-6[5] Mac:D4A0.2A88.0613 TCP socket:[10] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=5

    mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:12 privacy:1

    IP:177.3.11.18 19486 7962  keepalive 4 max_line 3 available_line 3

    button 1: dn 3  number 3001  CM Fallback CH1   IDLE         CH2   IDLE         CH3   IDLE         CH4   IDLE         CH5   IDLE         CH6   IDLE         CH7   IDLE         CH8   IDLE         

    button 2: dn 2  number 3010  CM Fallback CH1   IDLE         CH2   IDLE         CH3   IDLE         CH4   IDLE         CH5   IDLE         CH6   IDLE         CH7   IDLE         CH8   IDLE         shared 

    Preferred Codec: g711ulaw 





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  • Hi Michael:

    The phones were all reset and the router was subsequently rebooted. 

     

    Thanks,

  • just to clarify, reset / booted and no difference?


    On Wed, Oct 30, 2013 at 1:40 PM, rjferriolo <[email protected]> wrote:

    Hi Michael:

    The phones were all reset and the router was subsequently rebooted. 

     

    Thanks,




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  • What telnet client you are using? I guess this if the console output issue only, nothing to do with SRST or CME.

  • secure CRT


    On Wed, Oct 30, 2013 at 3:44 PM, MijanurRahman <[email protected]> wrote:

    What telnet client you are using? I guess this if the console output issue only, nothing to do with SRST or CME.




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  • Please try with Putty once just to check if it's returning the same.

  • Yes, the phones were rebooted from telephony-service and the router was subsequently rebooted. 

    • I have multiple frames established between Branch2 & the PSTN
    • From Branch2, I can Ping the PSTN phone
    • The Ephones are REGISTERED
    • I can reach a local extension but I cannot dial inbound/outbound to the PSTN
    • When I dial OUTBOUND, the call is not even reaching the gateway using debug voip dialpeer
    • When dialing INBOUND from the PSTN, I receive the following debug isdn q931 output:

    PSTN#

    ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x4 0x1, Calling num 7037333

    ISDN Se0/0/0:15 Q931: Sending SETUP  callref = 0x0086 callID = 0x8007 switch = primary-net5 interface = Network 

    ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref = 0x0086 

            Bearer Capability i = 0x8090A3 

                    Standard = CCITT 

                    Transfer Capability = Speech  

                    Transfer Mode = Circuit 

                    Transfer Rate = 64 kbit/s 

            Channel ID i = 0xA18381 

                    Preferred, Channel 1 

            Progress Ind i = 0x8183 - Origination address is non-ISDN 

    PSTN# 

            Display i = 'Amsterdam NL Phone' 

            Calling Party Number i = 0x4180, '7037333' 

                    Plan:ISDN, Type:Subscriber(local) 

            Called Party Number i = 0xC1, '7033001' 

                    Plan:ISDN, Type:Subscriber(local)

    ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8  callref = 0x8086 

            Channel ID i = 0xA98381 

                    Exclusive, Channel 1

    ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x8086 

            Cause i = 0x80A6 - Network out of order

    ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8  callref = 0x0086

    ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x8086

     

  • HI,

    Are you not translating the incoming number 7033001 to 3001? Can I have a look at your VOIP dial-peers?

  • Here is the config. Unfortunately nothing is reaching Branch2 Inbound or Outbound, so I can' t provide any debug output...

    thanks,

     

    Branch2#sh run

    Building configuration...

    Current configuration : 6604 bytes

    !

    ! Last configuration change at 16:38:39 CEDT Thu Oct 31 2013

    !

    version 12.4

    no service pad

    no service timestamps debug uptime

    no service timestamps log uptime

    no service password-encryption

    !

    hostname Branch2

    !

    boot-start-marker

    boot system flash:c2800nm-advipservicesk9-mz.124-24.T7.bin

    boot system flash:

    boot-end-marker

    !

    card type e1 0 0

    logging message-counter syslog

    no logging console

    !

    no aaa new-model

    clock timezone CEST 1

    clock summer-time CEDT recurring

    network-clock-participate wic 0 

    no network-clock-participate aim 0 

    !

    dot11 syslog

    ip source-route

    !

    !

    ip cef

    !

    !

    no ip domain lookup

    no ipv6 cef

    !

    multilink bundle-name authenticated

    !

    !

    !

    !

    isdn switch-type primary-net5

    !

    !

    !         

    voice service voip 

     no supplementary-service h225-notify cid-update

     fax protocol cisco 

    !

    !

    !

    voice class codec 1

     codec preference 1 g711ulaw

     codec preference 2 g711alaw

     codec preference 3 g729r8

    !

    !

    !

    !

    !

    !

    !

    voice class custom-cptone LEAVE-TONE

     dualtone conference

      frequency 300 3600

      cadence 500 100 150

    !

    voice class custom-cptone JOIN-TONE

     dualtone conference

      frequency 300 3600

      cadence 150 100 500

    !

    !

    !

    !

    !

    !

    !

    !

    voice translation-rule 1

     rule 1 /^7033...$/ /020&/

     rule 2 /^020703/ //

     rule 3 /^703/ //

     rule 4 /^7033/ /3/

    !

    voice translation-rule 10

     rule 1 /^0/ /0&/

    !

    voice translation-rule 200

     rule 1 /^206501...$/ /1&/

    !         

    !

    voice translation-profile 7DigitDNIS-to-10Digit

     translate called 1

    !

    voice translation-profile Prefix0_InFrom_CUCM

     translate called 10

    !

    voice translation-profile Prefix1-toCorpHQ-ANI

     translate calling 200

    !

    !

    voice-card 0

     dsp services dspfarm

    !

    !

    !

    !

    !

    archive

     log config

      hidekeys

    !         

    !

    !

    !

    controller E1 0/0/0

     pri-group timeslots 1-3,16

     description == Voice Circuit to PSTN

    !

    controller E1 0/0/1

    !

    !

    !

    !

    !

    interface Loopback0

     ip address 177.1.254.3 255.255.255.255

     h323-gateway voip bind srcaddr 177.1.254.3

    !

    interface FastEthernet0/0

     no ip address

     duplex auto

     speed auto

    !

    interface FastEthernet0/0.11

     encapsulation dot1Q 11

     ip address 177.3.11.1 255.255.255.0

     ip helper-address 177.1.10.10

    !

    interface FastEthernet0/0.12

     encapsulation dot1Q 12

     ip address 177.3.12.1 255.255.255.0

    !

    interface FastEthernet0/1

     no ip address

     shutdown

     duplex auto

     speed auto

    !

    interface Serial0/0/0:15

     no ip address

     encapsulation hdlc

     isdn switch-type primary-net5

     isdn incoming-voice voice

     isdn bchan-number-order ascending 

     no cdp enable

    !

    interface Serial0/1/0

     description == Frame-Relay Circuit to WAN

     no ip address

     encapsulation frame-relay

     fair-queue 64 256 37

     cdp enable

     no frame-relay inverse-arp

     frame-relay lmi-type ansi

     ip rsvp bandwidth

    !

    interface Serial0/1/0.1 point-to-point

     description == FR To HQ

     ip address 177.0.201.2 255.255.255.0

     snmp trap link-status

     frame-relay interface-dlci 102   

     ip rsvp bandwidth 136

    !

    interface Serial0/1/1

     no ip address

     shutdown

     clock rate 2000000

    !

    interface Service-Engine1/0

     no ip address

     shutdown

    !

    router ospf 1

     log-adjacency-changes

     network 0.0.0.0 255.255.255.255 area 0

    !

    ip forward-protocol nd

    ip http server

    ip http authentication local

    no ip http secure-server

    ip http path flash:

    !

    !

    !

    !

    !

    !

    !

    !

    !

    control-plane

    !

    !         

    !

    voice-port 0/0/0:15

     translation-profile incoming 7DigitDNIS-to-10Digit

    !

    ccm-manager music-on-hold

    !

    !

    sccp local Loopback0

    sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1 

    sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1 

    sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1 

    sccp

    !

    sccp ccm group 1

     bind interface Loopback0

     associat

     returning empty string !!!!e ccm 2 priority 1

     associate ccm 1 priority 2

     associate ccm 3 priority 3

     associate profile 1 register Br2-729-MTP

     associate profile 2 register Br2-711-MTP

     associate profile 3 register Br2-HW-Xcode

     associate profile 4 register Br2-HW-Conf

    !

    dspfarm profile 3 transcode  

     codec g711ulaw

     codec g711alaw

     codec g729ar8

     codec g729abr8

     codec g729r8

     codec g729br8

     maximum sessions 2

     associate application SCCP

    !

    dspfarm profile 4 conference  

     codec g711ulaw

     codec g711alaw

     codec g729ar8

     codec g729abr8

     codec g729r8

     codec g729br8

     maximum sessions 1

     conference-join custom-cptone JOIN-TONE

     conference-leave custom-cptone LEAVE-TONE

     associate application SCCP

    !         

    dspfarm profile 1 mtp  

     codec g729ar8

     codec g729r8

     rsvp

     maximum sessions software 10

     associate application SCCP

    !

    dspfarm profile 2 mtp  

     codec g711ulaw

     rsvp

     maximum sessions software 10

     associate application SCCP

    !

    !

    dial-peer voice 1 pots

     incoming called-number .

     direct-inward-dial

    !

    dial-peer voice 10 pots

     destination-pattern 112

     no digit-strip

     port 0/0/0:15

    !         

    dial-peer voice 11 pots

     destination-pattern 00[1-9]T

     port 0/0/0:15

     prefix 0

    !

    dial-peer voice 12 pots

     translation-profile outgoing Prefix1-toCorpHQ-ANI

     destination-pattern 000T

     port 0/0/0:15

     prefix 00

    !

    dial-peer voice 100 voip

     description == Inbound/Outbound H323 PSTN GW From/To GK and CUCM Pub

     translation-profile incoming Prefix0_InFrom_CUCM

     destination-pattern 0207033...$

     voice-class codec 1

     session target ipv4:177.1.10.10

     incoming called-number .

     ip qos dscp cs3 signaling

    !

    dial-peer voice 101 voip

     description == Outbound H323 PSTN GW To CUCM Sub

     translation-profile incoming Prefix0_InFrom_CUCM

     destination-pattern 0207033...$

     voice-class codec 1

     session target ipv4:177.1.10.20

     ip qos dscp cs3 signaling

    !

    !

    dial-peer hunt 1

    !

    !

    telephony-service

     srst mode auto-provision all

     srst dn line-mode octo

     max-ephones 10

     max-dn 10

     ip source-address 177.1.254.3 port 2000

     system message CM Fallback Service Operating

     max-conferences 8 gain -6

     transfer-system full-consult

     secondary-dialtone 0

     create cnf-files version-stamp 7960 Oct 30 2013 16:31:32

    !

    !

    ephone-dn  1  octo-line

     number 3002

     description 0207033002

     name 0207033002

    !

    !

    ephone-dn  2  octo-line

     number 3010

     description 0207033010

     name 0207033010

    !

    !

    ephone-dn  3  octo-line

     number 3001

     description 0207033001

     name 0207033001

    !

    !

    ephone  2

     device-security-mode none

     mac-address 001B.D4C6.9372

     button  1:1 2:2

    !

    !         

    !

    ephone  6

     device-security-mode none

     mac-address D4A0.2A88.0613

     button  1:3 2:2

    !

    !

    !

    line con 0

     exec-timeout 0 0

     privilege level 15

     logging synchronous level 0 limit 20

    line aux 0

    line 66

     no activation-character

     no exec

     transport preferred none

     transport input all

     transport output pad telnet rlogin lapb-ta mop udptn v120 ssh

    line vty 0 4

     exec-timeout 0 0

     privilege level 15

     logging synchronous

     no login

    line vty 5 15

     exec-timeout 0 0

     privilege level 15

     logging synchronous

     no login

    !

    scheduler allocate 20000 1000

    ntp source Loopback0

    ntp update-calendar

    ntp server 177.1.254.1

    end

  • Hi Mijanur:

    I was able to fix the OUTBOUND Local and National call routing by adding the following dial-peers, however I am still cannot dial INBOUND locally. Also, for some reason I am not getting any debug output for debug isdn q931 and debug voip dialpeer on Branch2 even for calls that are routing properly. I'm also still getting the error: returning empty string !!!! but it doesn't appear to be affecting phone registration or call routing. Please see the dial-peers that were added...

    Thanks,

    Ron

    ==========================================================================

    dial-peer voice 13 pots 

    destination-pattern 0[1-8]......$

    port 0/0/0:15

    forward-digits 7

    !

    dial-peer voice 14 pots 

    destination-pattern 00[1-8]........$

    port 0/0/0:15

    forward-digits 10

  • Hi ron on ur translation rules are not going to work. Delete translation rule 1 and replace with

    Voice translation-r 1

    Rule 1 /^703/ //

    Rule 2 /^20703/ //

     

    Regards

  • Is that because of the order in which they are processed?



    On Thu, Oct 31, 2013 at 1:15 PM, walt <[email protected]> wrote:

    Hi ron on ur translation rules are not going to work. Delete translation rule 1 and replace with

    Voice translation-r 1

    Rule 1 /^703/ //

    Rule 2 /^20703/ //

     

    Regards




    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



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  • Yes its top down and ur top rule is expanding the incoming called num to 10 digits. Ur ephones are 4 digit.

  • Looking at the current rule set:

    voice translation-rule 1

     rule 1 /^7033...$/ /020&/

     rule 2 /^020703/ //

     rule 3 /^703/ //

     rule 4 /^7033/ /3/


    rule 1 will prepend 020 to any number starting with 7033XXX (potentially longer)
    rule 2 will see a 6 digit number of 020703 and dump those digits (passing nothing I expect)...MAYBE passing the last 4 of the expected number

    rule 3 will pretty much mirror 2 in its actions...
    rule 4 won't get hit because 3 would trigger first.

    If what you're trying to do is (catch either 7033XXX and prepend 020 and pass it along) and (catch specifically 0207033xxx and pass it along as 4 digits) then I would modify the second a little to /^020703(3...$)/ /1/.  That second rule says any number matching 0207033XXX pass on the values within the FIRST set of parentheses (in this case...the only set) which is 3XXX.


    If we change it to:

    voice translation-rule 1

     rule 1 /^703(3...$)/ /020&/

     rule 2 /^020703(3...$)/ /1/


    then it's more in line with what they would expect during an IE exam.  It's more specific and you know exactly when it will trigger...and let's face...it just looks sexier :)





    On Thu, Oct 31, 2013 at 2:48 PM, rjferriolo <[email protected]> wrote:
    Is that because of the order in which they are processed?




    On Thu, Oct 31, 2013 at 1:15 PM, walt <[email protected]> wrote:

    Hi ron on ur translation rules are not going to work. Delete translation rule 1 and replace with

    Voice translation-r 1

    Rule 1 /^703/ //

    Rule 2 /^20703/ //

     

    Regards




    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx




    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx

  • Thanks for the info Michael. I have only foundational knowledge of voice translation-rules. I haven't had an opportunity to hit the Deep Dives yet. 



    Didn't know VoIP was sexy. You learn something new every day. 


    I was able to get rid of the returning empty string!!! 

    error by re-activating the Callmanager service and shutting down the Serial WAN interfaces between Branch2 & CorpHQ instead. When it came back on line in SRST mode, the error went away. 
    Everything is routing, however, in SRST mode when I make a call from Branch2 (CME-as-SRST), I get no debug output on any of the three routers even though the call is routing properly. 



    Any ideas? 


    Thanks,


    Ron



    On Thu, Oct 31, 2013 at 3:39 PM, michael.thompson <[email protected]> wrote:
    Looking at the current rule set:

    voice translation-rule 1

     rule 1 /^7033...$/ /020&/

     rule 2 /^020703/ //

     rule 3 /^703/ //

     rule 4 /^7033/ /3/



    rule 1 will prepend 020 to any number starting with 7033XXX (potentially longer)
    rule 2 will see a 6 digit number of 020703 and dump those digits (passing nothing I expect)...MAYBE passing the last 4 of the expected number


    rule 3 will pretty much mirror 2 in its actions...
    rule 4 won't get hit because 3 would trigger first.

    If what you're trying to do is (catch either 7033XXX and prepend 020 and pass it along) and (catch specifically 0207033xxx and pass it along as 4 digits) then I would modify the second a little to /^020703(3...$)/ /1/.  That second rule says any number matching 0207033XXX pass on the values within the FIRST set of parentheses (in this case...the only set) which is 3XXX.



    If we change it to:

    voice translation-rule 1

     rule 1 /^703(3...$)/ /020&/

     rule 2 /^020703(3...$)/ /1/


    then it's more in line with what they would expect during an IE exam.  It's more specific and you know exactly when it will trigger...and let's face...it just looks sexier :)





    On Thu, Oct 31, 2013 at 2:48 PM, rjferriolo <[email protected]> wrote:
    Is that because of the order in which they are processed?





    On Thu, Oct 31, 2013 at 1:15 PM, walt <[email protected]> wrote:

    Hi ron on ur translation rules are not going to work. Delete translation rule 1 and replace with

    Voice translation-r 1

    Rule 1 /^703/ //

    Rule 2 /^20703/ //

     

    Regards




    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx





    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx




    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx

  • Ron,
        once you get used to voice translation-rule / voice translation-profile builds, they're really are crazy easy...it's mostly about getting in the right mind set.  Once you start getting into the digit replace options, !,+, etc., it requires a little more thought and practice.  Mark is excellent at simplifying the thought process required to analyze how the router will see the digits.  


    VoIP is sexy because smart is sexy...at least for the love of God I hope so...otherwise I'm in real trouble and somehow I shot WELL above my pay grade with my wife :)

    shutting down CCM services should be sufficient to flip in / out of SRST / CME as SRST albeit a heavy handed approach.  One of the things that INE does with their racks and Mark points out in one of the deep dives that does this process VERY well.  The key thing is that it 'kills' call control without killing your access to things like the http page of the IP phones.  It's an ACL that sits on the IP Subnet of the CUCM servers that blocks call control (SCCP / 2000, h323 / 1720, mgcp / 2428, 2427, etc.) as necessary.  This way things like CFUR will trip (CUCM CCM services need to be up to trigger CFUR).  Keep in mind, things like AAR are tripped by low bandwidth (locations / RSVP CAC triggers) and not by a down issue.


    HTH and keep trying strange things in your lab...I learn so much more from the 'why the hell won't this work' that turn out to be an oversight on my part than I do from straight lecture and practice stuff.  A balance between those two things is critical to hit all your bases.


    MT


    On Sat, Nov 2, 2013 at 11:51 AM, rjferriolo <[email protected]> wrote:
    Thanks for the info Michael. I have only foundational knowledge of voice translation-rules. I haven't had an opportunity to hit the Deep Dives yet. 





    Didn't know VoIP was sexy. You learn something new every day. 



    I was able to get rid of the returning empty string!!! 


    error by re-activating the Callmanager service and shutting down the Serial WAN interfaces between Branch2 & CorpHQ instead. When it came back on line in SRST mode, the error went away. 
    Everything is routing, however, in SRST mode when I make a call from Branch2 (CME-as-SRST), I get no debug output on any of the three routers even though the call is routing properly. 




    Any ideas? 



    Thanks,



    Ron



    On Thu, Oct 31, 2013 at 3:39 PM, michael.thompson <[email protected]> wrote:
    Looking at the current rule set:

    voice translation-rule 1

     rule 1 /^7033...$/ /020&/

     rule 2 /^020703/ //

     rule 3 /^703/ //

     rule 4 /^7033/ /3/




    rule 1 will prepend 020 to any number starting with 7033XXX (potentially longer)
    rule 2 will see a 6 digit number of 020703 and dump those digits (passing nothing I expect)...MAYBE passing the last 4 of the expected number



    rule 3 will pretty much mirror 2 in its actions...
    rule 4 won't get hit because 3 would trigger first.

    If what you're trying to do is (catch either 7033XXX and prepend 020 and pass it along) and (catch specifically 0207033xxx and pass it along as 4 digits) then I would modify the second a little to /^020703(3...$)/ /1/.  That second rule says any number matching 0207033XXX pass on the values within the FIRST set of parentheses (in this case...the only set) which is 3XXX.




    If we change it to:

    voice translation-rule 1

     rule 1 /^703(3...$)/ /020&/

     rule 2 /^020703(3...$)/ /1/


    then it's more in line with what they would expect during an IE exam.  It's more specific and you know exactly when it will trigger...and let's face...it just looks sexier :)





    On Thu, Oct 31, 2013 at 2:48 PM, rjferriolo <[email protected]> wrote:

    Is that because of the order in which they are processed?






    On Thu, Oct 31, 2013 at 1:15 PM, walt <[email protected]> wrote:


    Hi ron on ur translation rules are not going to work. Delete translation rule 1 and replace with

    Voice translation-r 1

    Rule 1 /^703/ //

    Rule 2 /^20703/ //

     

    Regards




    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx





    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx




    INE - The Industry Leader in CCIE Preparation


    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx




    INE - The Industry Leader in CCIE Preparation


    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx

  • Michael: 


    Your're absolutely right about the frustrations of a feature not working eventually becoming a blessing because of the troubleshooting you're forced to go through even though it certainly doesn't seem like it in the moment. 


    In the home lab I was able to dial between Branch1 (running on Callmanager Fallback) and Branch2 (running on CME-as-SRST) over the PSTN. This is a good thing to learn how to configure in the event of a catastrophic weather condition such as we had here in New York with the hurricane last year.



    On Sun, Nov 3, 2013 at 1:30 AM, michael.thompson <[email protected]> wrote:
    Ron,
        once you get used to voice translation-rule / voice translation-profile builds, they're really are crazy easy...it's mostly about getting in the right mind set.  Once you start getting into the digit replace options, !,+, etc., it requires a little more thought and practice.  Mark is excellent at simplifying the thought process required to analyze how the router will see the digits.  



    VoIP is sexy because smart is sexy...at least for the love of God I hope so...otherwise I'm in real trouble and somehow I shot WELL above my pay grade with my wife :)

    shutting down CCM services should be sufficient to flip in / out of SRST / CME as SRST albeit a heavy handed approach.  One of the things that INE does with their racks and Mark points out in one of the deep dives that does this process VERY well.  The key thing is that it 'kills' call control without killing your access to things like the http page of the IP phones.  It's an ACL that sits on the IP Subnet of the CUCM servers that blocks call control (SCCP / 2000, h323 / 1720, mgcp / 2428, 2427, etc.) as necessary.  This way things like CFUR will trip (CUCM CCM services need to be up to trigger CFUR).  Keep in mind, things like AAR are tripped by low bandwidth (locations / RSVP CAC triggers) and not by a down issue.



    HTH and keep trying strange things in your lab...I learn so much more from the 'why the hell won't this work' that turn out to be an oversight on my part than I do from straight lecture and practice stuff.  A balance between those two things is critical to hit all your bases.



    MT


    On Sat, Nov 2, 2013 at 11:51 AM, rjferriolo <[email protected]> wrote:
    Thanks for the info Michael. I have only foundational knowledge of voice translation-rules. I haven't had an opportunity to hit the Deep Dives yet. 







    Didn't know VoIP was sexy. You learn something new every day. 




    I was able to get rid of the returning empty string!!! 



    error by re-activating the Callmanager service and shutting down the Serial WAN interfaces between Branch2 & CorpHQ instead. When it came back on line in SRST mode, the error went away. 
    Everything is routing, however, in SRST mode when I make a call from Branch2 (CME-as-SRST), I get no debug output on any of the three routers even though the call is routing properly. 





    Any ideas? 




    Thanks,




    Ron



    On Thu, Oct 31, 2013 at 3:39 PM, michael.thompson <[email protected]> wrote:

    Looking at the current rule set:

    voice translation-rule 1

     rule 1 /^7033...$/ /020&/

     rule 2 /^020703/ //

     rule 3 /^703/ //

     rule 4 /^7033/ /3/





    rule 1 will prepend 020 to any number starting with 7033XXX (potentially longer)
    rule 2 will see a 6 digit number of 020703 and dump those digits (passing nothing I expect)...MAYBE passing the last 4 of the expected number




    rule 3 will pretty much mirror 2 in its actions...
    rule 4 won't get hit because 3 would trigger first.

    If what you're trying to do is (catch either 7033XXX and prepend 020 and pass it along) and (catch specifically 0207033xxx and pass it along as 4 digits) then I would modify the second a little to /^020703(3...$)/ /1/.  That second rule says any number matching 0207033XXX pass on the values within the FIRST set of parentheses (in this case...the only set) which is 3XXX.





    If we change it to:

    voice translation-rule 1

     rule 1 /^703(3...$)/ /020&/

     rule 2 /^020703(3...$)/ /1/


    then it's more in line with what they would expect during an IE exam.  It's more specific and you know exactly when it will trigger...and let's face...it just looks sexier :)





    On Thu, Oct 31, 2013 at 2:48 PM, rjferriolo <[email protected]> wrote:


    Is that because of the order in which they are processed?







    On Thu, Oct 31, 2013 at 1:15 PM, walt <[email protected]> wrote:



    Hi ron on ur translation rules are not going to work. Delete translation rule 1 and replace with

    Voice translation-r 1

    Rule 1 /^703/ //

    Rule 2 /^20703/ //

     

    Regards




    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx





    INE - The Industry Leader in CCIE Preparation

    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx




    INE - The Industry Leader in CCIE Preparation



    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx




    INE - The Industry Leader in CCIE Preparation



    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx




    INE - The Industry Leader in CCIE Preparation


    http://www.INE.com



    Subscription information may be found at:

    http://www.ieoc.com/forums/ForumSubscriptions.aspx

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