Greetings, we have recently been provisioned 10 Sip trunks each with individual account usernames and passwords as well as 20 DDI numbers, these numbers are configured by the service provider to route all calls via one of the SIP trunk accounts.
Now i can make a call inbound and outbound but cant make multiple concurrent calls.
This is what i have so far
voice translation-rule 1
rule 1 /.........8158/ /6001/
rule 2 /.........8364/ /1000/
rule 3 /.........8365/ /1001/
rule 4 /.........8366/ /1002/
rule 5 /.........8367/ /1003/
rule 6 /.........8368/ /1004/
rule 7 /.........8369/ /1005/
rule 8 /.........8370/ /1006/
!
voice translation-rule 2
rule 1 /^999$/ /999/
rule 2 /^9\(.*\)$/ /\1/
!
voice translation-rule 3
rule 1 /.*/ /84413141/
!
!
voice translation-profile SIP_Inbound
translate called 1
!
voice translation-profile SIP_Trunk_1
translate calling 3
translate called 2
!
voice service voip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 900 min 900
transport switch udp tcp
localhost dns:*******************.co.uk
!
dial-peer voice 9000 voip
description ** cue voicemail pilot number **
destination-pattern 9000
session protocol sipv2
session target ipv4:10.10.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 9001 voip
description ** cue auto attendant number **
destination-pattern 9001
b2bua
session protocol sipv2
session target ipv4:10.10.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1 voip
description **Outbound call to SIP**
translation-profile outgoing SIP_Trunk_1
preference 1
destination-pattern 9T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip
description **Inbound call from SIP**
translation-profile incoming SIP_Inbound
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
no vad
!
sip-ua
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
credentials username 8888888 password 7 realm voiceflex
authentication username 84413141 password 7
nat symmetric role passive
nat symmetric check-media-src
no remote-party-id
retry invite 3
retry register 5
retry options 5
timers connect 100
timers register 100
registrar ipv4:***.**.***.*** expires 80
sip-server ipv4:***.**.***.***
host-registrar
I have a single incomming dial peer and basic translation rule and a single outgoing dial peer which maps the calling number to the SIP Account which all the numbers are connected too.
Has anyone got any examples of how to configure this for concurrent calls?
Any help would be much appreciated.
Regards
