SRST failure

Unanswered Question

I have a new site I am trying to setup SRST at, having a few issues.


I have put my phone into SRST failure over mode, however outbound and inbound calls are failing.


I show the phone registers with the gateway, however when I attempt to make an outbound call I never see it hit the gateway, the phone rings fast busy.


Translation of inbound numbers to ext number applied to voice port


voice translation-rule 2

rule 1 /^7....../ /1303XXXXXXX/

!

voice translation-rule 9

rule 1 /^713...\(....\)/ /715\1/

!

!


voice-port 0/0/0:23

translation-profile incoming Inbound

!

voice-port 0/1/0:23

translation-profile incoming Inbound

!

voice-port 0/0/1:23

translation-profile incoming Inbound

voice translation-profile Inbound

translate called 9

!

voice translation-profile Outbound

translate called 2




dial-peer voice 1 pots

translation-profile incoming inbound

translation-profile outgoing Outbound

preference 1

destination-pattern 9T

incoming called-number .

direct-inward-dial

port 0/0/1:23

forward-digits all



dial-peer voice 2 pots

translation-profile incoming inbound

translation-profile outgoing Outbound

preference 1

destination-pattern 9T

incoming called-number .

direct-inward-dial

port 0/1/0:23

forward-digits all

!

dial-peer voice 3 pots

translation-profile incoming inbound

translation-profile outgoing Outbound

preference 1

destination-pattern 9T

incoming called-number .

direct-inward-dial

port 0/0/0:23

forward-digits all



call-manager-fallback

secondary-dialtone 9

max-conferences 16 gain -6

transfer-system full-consult

timeouts interdigit 6

timeouts ringing 12

ip source-address 10.XX.XX.XX port 2000

max-ephones 730

max-dn 960

system message primary SRST Fallback

system message secondary SRST Fallback

transfer-pattern .T

voicemail 9XXXXXXXXXXX

no huntstop

moh SampleAudioSource.ULAW.wav

multicast moh 239.1.1.1 port 16384 route XX.XX.XX.XX


ephone-109[108] Mac:0022.90B9.BF55 TCP socket:[65] activeLine:0 whisperLine:0 REGISTERED

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0

IP:10.XX.XX.XX 7942 keepalive 132 music 0 1:1 CM Fallback


Any ideas? Please note this is a new install, so I am sure I am missing something.

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a.gooding Sun, 07/19/2009 - 17:48
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taking a shot here but normally id configure the ALIAS command to ensure that calls coming in from the PSTN would reach a phone. Also ill normally have another command, say ACCESS-CODE 9 FXO (Or in your case PRI) to make outgoing calls.

KonradStepniewski Sun, 07/19/2009 - 23:38
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When you block one phone and you have MGCP gateway then you can only test SRTS registration. You won't be able to make any calls as you need MGCP fallback set on gateway.


If you have problems with translations test it with debug isdn q931, so you will see what you send and receive from telco.

I see you have forward-digit-all in your dial-peers. You will send 9 to telco which is not good.

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