Hi Friends ,
i have installed SIP trunk & incoming and outgoing calls are working fine but i am facing the issue on incoming fax call.
let says
Service provider------SIP Trunk----------CISCO R3845 -------H323----Rightfax .
ISSUE: Incoming call for FAX is connected and give FAX tone and after 3 sec calls disconnect but outgoign is working fine .
Appreciated if some one help me!!!
debug is attached
calling number 4791313
called number 8749063 then translate to 9063
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voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
modem passthrough nse codec g711alaw
sip
early-offer forced
!
voice class codec 1
codec preference 1 g729r8
codec preference 3 g711ulaw
codec preference 4 g711alaw
!
!
!
voice translation-rule 1
rule 1 /^56\(..\)/ /87456\1/
rule 2 /^57\(..\)/ /87457\1/
rule 3 /^90\(..\)/ /87490\1/
rule 4 /^93\(..\)/ /87493\1/
rule 5 /^91\(..\)/ /87491\1/
!
voice translation-rule 2
rule 1 /^87456\(..\)/ /56\1/
rule 2 /^87457\(..\)/ /57\1/
rule 3 /^87490\(..\)/ /90\1/
rule 4 /^87493\(..\)/ /93\1/
rule 5 /^87491\(..\)/ /91\1/
!
voice translation-rule 10
rule 1 /\(5........\)/ /00\1/
rule 2 /1\(.......$\)/ /0\1/
rule 3 /\(2.......$\)/ /00\1/
rule 4 /\(4.......$\)/ /00\1/
rule 5 /\(3.......$\)/ /00\1/
rule 6 /\(6.......$\)/ /00\1/
rule 7 /\(7.......$\)/ /00\1/
!
!
voice translation-profile IN
translate calling 10
translate called 2
!
voice translation-profile OUT
translate calling 1
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voice translation-profile addprefix
translate calling 10
!
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voice translation-profile test
translate called 3
!
!
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ip route 0.0.0.0 0.0.0.0 10.1.5.1
ip route 10.200.7.157 255.255.255.255 172.29.17.37
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dial-peer voice 39101 voip
description *** Right fax ***
huntstop
destination-pattern 9101
session protocol sipv2
session target ipv4:172.16.0.27
session transport udp
incoming called-number 9101
voice-class sip early-offer forced
dtmf-relay rtp-nte
codec g711alaw
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
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dial-peer voice 1 voip
description *** SIP Incoming Dial-peer ***
translation-profile incoming IN
session protocol sipv2
session target sip-server
session transport udp
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
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