SIP Account

Unanswered Question
Jun 25th, 2009
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Hi,

I have a CME 7.0 on a Cisco 2811 router.

I configured an SIP account with more numer on the same username and password and URI.

The configuration is:


voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

redirect ip2ip

h323

no h225 timeout keepalive

call preserve limit-media-detection

modem passthrough nse codec g711alaw

sip

bind control source-interface FastEthernet0/0

bind media source-interface FastEthernet0/0

registrar server


voice translation-rule 2

rule 1 /^0\(.*\)/ /\1/

!

voice translation-rule 3

rule 1 /^.*/ /5353188/

!

voice translation-rule 4

rule 1 /5949030890/ /999/

rule 2 /5353188/ /999/

rule 15 /^.*/ /999/

!

voice translation-rule 5

rule 1 /5949030891/ /999/

rule 2 /5949030890/ /100/

rule 3 /5353188/ /100/

rule 15 /^.*/ /100/


voice translation-profile PSTN_Incomming_Acetaia

translate called 5

!

voice translation-profile PSTN_Incomming_GCREM

translate called 4

!

voice translation-profile PSTN_Outgoing

translate calling 3

translate called 2


dial-peer voice 20 voip

description Chiamate verso Sip

translation-profile outgoing PSTN_Outgoing

destination-pattern 0T

redirect ip2ip

rtp payload-type cisco-codec-fax-ind 102

rtp payload-type nte 96

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

session transport udp

dtmf-relay rtp-nte

no vad

!

dial-peer voice 21 voip

description Chiamate entranti da Sip per GCREM

translation-profile incoming PSTN_Incomming_GCREM

redirect ip2ip

rtp payload-type cisco-codec-fax-ind 102

rtp payload-type nte 96

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

session transport udp

incoming called-number 5949030890

dtmf-relay rtp-nte

no vad

!

dial-peer voice 22 voip

description Chiamate entranti da Sip per Acetaia

translation-profile incoming PSTN_Incomming_Acetaia

redirect ip2ip

rtp payload-type cisco-codec-fax-ind 102

rtp payload-type nte 96

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

session transport udp

incoming called-number 5949030891

dtmf-relay rtp-nte

no vad

!

!

sip-ua

credentials username 5353188 password 7 15151E180124222D realm sip.messagenet.it

authentication username 535XXXX password 7 10491C0D00191B02

retry invite 4

retry response 3

retry bye 2

retry cancel 2

registrar ipv4:212.97.59.76:5060 expires 3600 sip-server ipv4:212.97.59.76:5060




show sip-ua register status

Line peer expires(sec) registered

============ ============= ============ ===========

5353188 -1 2115 yes



I can call with both the numbers on the same SIP account.

I need to ring one Ip-phone in the case the number called is 5949030890 an other IP-Phone in case the number called is 5949030891.


In attach you can find the debug output of:

debug voice dialpeer detail

debug voice dialpeer inout

debug voice translation



How can I make this work?



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bwalchez Wed, 07/01/2009 - 13:25
User Badges:

Try to use the voice translation rule:

voice translation-rule xx

rule 1 /^[+]\(.*\)/ /\1/


alex.sintoni Wed, 07/01/2009 - 23:39
User Badges:

Thank you for your reply, but this translation-rule didn't match anything.

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