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Configuring CCME Concurrent Calls Using SIP

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

1 REPLY
Silver

Re: Configuring CCME Concurrent Calls Using SIP

The number of concurrent calls is limited by the number of MTP resources available. The limit of 48 refers to software MTP.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00808b6ca6.shtml

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