08-09-2015 06:21 AM - edited 03-17-2019 03:55 AM
ATA 190 not working in SRST . Unable to send fax
CCM: 10.5.2
SRST MGCP Gateway: 2091
boot-start-marker
boot-end-marker
!
!
card type e1 0 0
!
no aaa new-model
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
!
ip dhcp pool ipphones
network 192.168.210.0 255.255.255.0
default-router 192.168.210.1
option 150 ip 192.168.210.10
domain-name wr
!
!
!
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
isdn switch-type primary-net5
!
cts logging verbose
!
!
voice-card 0
dspfarm
dsp services dspfarm
!
!
!
!
!
!
voice translation-rule 7
rule 1 /^2/ /043652/
!
voice translation-rule 9
rule 1 /^1/ /91/
rule 2 /^2/ /902/
rule 3 /^3/ /903/
rule 4 /^4/ /904/
rule 5 /^5/ /905/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /98/
rule 9 /^9/ /9009/
rule 10 /^0/ /90/
!
voice translation-rule 10
rule 1 /^1/ /83041/
rule 2 /^2/ /83042/
rule 3 /^3/ /83043/
rule 4 /^4/ /83044/
rule 5 /^5/ /83045/
rule 6 /^6/ /83046/
rule 7 /^7/ /83047/
rule 8 /^8/ /83048/
rule 9 /^9/ /83049/
rule 10 /^0/ /83040/
!
voice translation-rule 11
rule 1 /^1/ /83651/
rule 2 /^2/ /83652/
rule 3 /^3/ /83653/
rule 4 /^4/ /83654/
rule 5 /^5/ /83655/
rule 6 /^6/ /83656/
rule 7 /^7/ /83657/
rule 8 /^8/ /83658/
rule 9 /^9/ /83659/
rule 10 /^0/ /83650/
!
!
voice translation-profile Add9
translate calling 9
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voice translation-profile CALL-KWT
translate calling 10
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voice translation-profile CALL-NIBBranches
translate calling 10
!
voice translation-profile change_calling
translate calling 7
!
!
!
!
application
global
service alternate default
!
!
license udi pid CISCO2921/K9
hw-module pvdm 0/0
!
!
!
!
redundancy
!
!
controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-31 service mgcp
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address 192.168.210.11 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/2
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 outgoing display-ie
isdn outgoing ie redirecting-number
no cdp enable
!
ip forward-protocol nd
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no ip http server
no ip http secure-server
!
!
!
!
control-plane
!
!
voice-port 0/0/0:15
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
voice-port 0/2/0
!
voice-port 0/2/1
!
voice-port 0/2/2
!
voice-port 0/2/3
!
!
!
!
!
mgcp
mgcp call-agent 192.168.210.10 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp bind control source-interface GigabitEthernet0/0
mgcp bind media source-interface GigabitEthernet0/0
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm 192.168.210.10 identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register HQ-XCODE
associate profile 1 register HQ-CONF
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ccm-manager music-on-hold
!
ccm-manager switchback immediate
ccm-manager fallback-mgcp
ccm-manager mgcp
ccm-manager config
!
dspfarm profile 2 transcode universal
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g729br8
codec g729r8
codec ilbc
codec pass-through
maximum sessions 3
associate application SCCP
!
dspfarm profile 1 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec ilbc
maximum sessions 2
associate application SCCP
!
dial-peer voice 1 pots
translation-profile incoming Add9
preference 7
destination-pattern 9.T
incoming called-number .
direct-inward-dial
port 0/0/0:15
forward-digits all
!
dial-peer voice 101 pots
translation-profile outgoing change_calling
preference 3
destination-pattern 9(050.......)
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 7
direct-inward-dial
port 0/0/0:15
!
dial-peer voice 102 pots
translation-profile outgoing change_calling
preference 2
destination-pattern 9([2,3,4,5]......)
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 7
direct-inward-dial
port 0/0/0:15
!
dial-peer voice 103 pots
translation-profile outgoing change_calling
preference 1
destination-pattern 9(0[2,3,6,7].......)
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 7
direct-inward-dial
port 0/0/0:15
!
dial-peer voice 104 pots
translation-profile outgoing change_calling
destination-pattern 9T
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 7
direct-inward-dial
port 0/0/0:15
!
dial-peer voice 165 pots
translation-profile outgoing change_calling
preference 3
destination-pattern 9(055.......)
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 7
direct-inward-dial
port 0/0/0:15
!
!
dial-peer voice 500 voip
destination-pattern 5...
session protocol sipv2
session target ipv4:192.168.210.11
dtmf-relay sip-notify
codec g711ulaw
no vad
!
!
!
!
num-exp 2800 2824
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 192.168.210.11 port 2000
max-ephones 75
max-dn 152 dual-line
keepalive 20
!
!
!
08-09-2015 09:07 AM
Hi,
You are using ephones SRST and ATA190 uses SIP. Try to configure voice register global in SRST mode.
08-09-2015 11:28 AM
Following commands are Ok ?
Whats the IP id network ?
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
h323
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 600 min 60
no silent-discard untrusted
!
!
voice register global
mode srst
timeouts interdigit 7
max-dn 10
max-pool 40
!
voice register pool 1
registration-timer max 120 min 60
id network XX.XXX.XXX.XX mask 255.255.255.0
dtmf-relay rtp-nte sip-notify
voice-class codec 1
!
Can you help me?
08-09-2015 08:05 PM
Waqas,
Whats the IOS version?
Secondly On the ATA-190 page on the CUCM./ Check the network configuration and see if you have a ip-address of the sip-srst gateway.
-Regards,
Kevin
08-09-2015 08:09 PM
Waqas,
On the gateway, just for testing purpose make the following change and remove the following " no registration-timer max 120 min 60".
Under "voice service voip" do the following:
voice service voip
no ip address turst authenticate
voice register pool 1
no registration-timer max 120 min 60
id network 0.0.0.0 mask 0.0.0.0
dtmf-relay rtp-nte sip-notify
voice-class codec 1
-Kevin
08-09-2015 09:54 PM
Hi Waqas,
Your config is missing the following:
1. source interface IP under voice register global
2. the id network should be the subnet of your SIP Phones/ATA which is going to use SRST
3. In you CUCM, create an SRST reference and assign it to the device pool used by SIP Phones/ATA. Once the SIP Phones/ATA Register with CUCM they will get the SRST IP and they can fallback.
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