
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
Hi Michael:
The phones were all reset and the router was subsequently rebooted.
Thanks,
What telnet client you are using? I guess this if the console output issue only, nothing to do with SRST or CME.
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.
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
On Thu, Oct 31, 2013 at 1:15 PM, walt <[email protected]> wrote:
Yes its top down and ur top rule is expanding the incoming called num to 10 digits. Ur ephones are 4 digit.
voice translation-rule 1
rule 1 /^7033...$/ /020&/
rule 2 /^020703/ //
rule 3 /^703/ //
voice translation-rule 1
rule 1 /^703(3...$)/ /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
Didn't know VoIP was sexy. You learn something new every day.
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.
Any ideas?
Thanks,
Ron