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ATA 190 not working in SRST

waqas sardar
Level 1
Level 1

 

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
!
voice translation-profile CALL-KWT
 translate calling 10
!
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
!
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
!
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
!
!
!

5 Replies 5

Hi,

 

You are using ephones SRST and ATA190 uses SIP. Try to configure voice register global in SRST mode.

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?


 

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

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

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.