07-19-2009 11:42 AM - edited 03-15-2019 07:01 PM
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.
07-19-2009 02:44 PM
I am wondering if this issue is caused b/c I have an access-list to block just one phone and not the entire site including the voice gateway. Thoughts?
07-19-2009 05:48 PM
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.
07-19-2009 11:38 PM
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.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide