SIP inbound and outboung not working CME

[yamikani2g2]

Good day Experts

I am not getting any luck with my inbound and outbound calls on my SIP trunk. Internally phones work.

They can call each other thats a piece of cake i know...

Firslty when i do show dialplan incall  i cal see my dialplans being hit.

My Debug ccsip its either erro 508 to 404 below is a snippit.

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.9.6.18:5060;branch=z9hG4bK5o9345o9n45malrg5a4gc3n4y;Role=3;Hpt=8e68_16
From: "977466567"<sip:977466567@192.168.10.1;transport=udp;user=phone>;tag=9208srtv-CC-1002-OFC-29
To: "760918270"<sip:760918270@192.168.10.1;transport=udp;user=pho
VOIP-RTR#ne>;tag=156497CC-E60
Date: Tue, 05 Nov 2019 16:37:18 GMT
Call-ID: isbcrcphcumla2tlplhrjv0uj2mm2tupajjm@10.18.5.64
CSeq: 1 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.10.1 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-15.7.3.M
Reason: Q.850;cause=1
Session-ID: 5e35775caebb5f949423957d1404b072;remote=966171d16fa350b3a19e9d565c78c371
Content-Length: 0

Attached is my relevant configs for this SIP trunk i have omitted other parts.

[TONY SMITH]

Looks like it's not finding an outbound dial peer, assuming that the SIP in your debug is between the phone and the CME.  Can you show your dial peers?   Also I've a bit puzzled by the ptime=5, but that might be a red herring as I don't usually look at SIP phone (rather than trunk) traffic. 

[yamikani2g2]

I don't think you analysed the show run I will add the running configurations here maybe that can assist.

dial-peer voice 4000 voip
translation-profile incoming direct-line
session protocol sipv2
session target ipv4:10.9.6.18:5060
incoming called-number .
dtmf-relay rtp-nte
codec g711alaw

voice translation-rule 4000
rule 1 /^760918260$/ /4013/

voice translation-profile direct-line
translate called 4000

dial-peer voice 4001 voip
description SIP TRUNK TO ITSP
destination-pattern 0T
session protocol sipv2
session target ipv4:10.9.6.18:5060
dtmf-relay rtp-nte
codec g711alaw

VOIP-RTR#test voice translation-rule 4000 760918260
Matched with rule 1
Original number: 760918260 Translated number: 4013
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none

[TONY SMITH]

I don't see a full "show run" in either of your posts.   However with a bit of guess work it appear you are dialling 760918270 which may or may not be translated to 4013 depending on how that inbound dial peer relates to the rest of the configuration.  Outbound you only show one dial peer with destination-pattern 0T.   That won't match either 760918270 or 4013, it will only match a dialled number starting with 0.

Is there anything else configured that would match?  What do you get from "show dialplan number 760918270"  and "show dialplan number 4013" 

Can you paste in your full configuration, or if you don't want to do then then at least "show dial-peer voice sum"

[yamikani2g2 · Accepted solution]

Solution below is what worked i now can call in and out i only have issue with one way audio.

dial-peer voice 4001 voip
description SIP TRUNK TO ITSP
translation-profile outgoing OUTSIDECALLING
destination-pattern 0T
session protocol sipv2
session target ipv4:10.9.6.18:5060
dtmf-relay rtp-nte
codec g711alaw
!
dial-peer voice 4000 voip
translation-profile incoming direct-line
session protocol sipv2
session target ipv4:10.9.6.18:5060
incoming called-number .%
dtmf-relay rtp-nte
codec g711alaw
!

voice translation-rule 4000
rule 1 /760918260/ /4013/
rule 2 /760918270/ /4000/
!
voice translation-rule 4001
rule 1 /^4013$/ /760918270/
rule 2 /^4000$/ /760918260/
!
!
voice translation-profile OUTSIDECALLING
translate calling 4001
!
voice translation-profile direct-line
translate called 4000
!