Help setting up Dial-Peer in the CME

Hello experts,

I tried to set up my network voice for 2 days but no luck.

Here is my scenario:

CUCM

CME

IRIS PBX

+++ Posted inside Config.

CME and PBX are PRI resources; I can see active layer 1 and Layer 2 multi_Frame_Established in 'display the ISDN State '.

SIP trunk of CUCM to established CMF.

Cannot communicate registered CUCM to analog phones in PBX Voip phones.

Several times analog voip calls work and then some changes in the PBX, which in fact shit once again. (Those that have been made by PBX technician)

VoIP ext: 1003

Analog position: 270

!! for voip analog calls

In CUCM:

Routing model: 9XXX--> to the trunk of the CME

In the CME:

The destination model: 9... --> E1 card (with auto stripping)

!! for analog / voip calls

In PBX access code 4 and stripped when sending guys.

that the means to post analogue for reach 1003 user consist 41003 and 4 will be stripped.

In the CME:

The destination model: 1... --> session target ipv4:

With this configuration in the 'debugging ccapi inout"see"transfer number is NULL"and

in the "show history of last calls 1" am see reason code "1 c" - invalid number.

After some research, this are my ideas but not tested:

1. I have incoming dial Exchange files corresponds to the movement of e1 card:

voice pots Dial-peer X

incoming called-number. T

(2) I have to configure dtmf-relay-notify sip and dtmf-relay rtp - nte to accept the numbers of telephone service.

(3) need to create any DIDs for incoming calls to pbx?

Please help me on this point, how could I get from analog to voip calls and vice versa, which I think is fair? If this is not the case, what would be the alternatives? where we should use the dtmf-relay command. ?

Thanks to all in advance.

Current config:

type 0 1 e1 card
network-clock-participate wic 1
network-clock-select 1 E1 1/0/1
primary-qsig ISDN switch type
voip phone service
h323 connections allow h323
allow connections h323 to SIP
allow connections sip h323
allow sip to sip connections
no service additional h450.2
no service additional h450.3
Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
SIP
Registration Server expires max 600 min 60
no update-callerid
!
voice class codec 1
preferably 1 codec g729br8
codec preference 2 g729r8
g723ar63 preferably 3 codec
g711ulaw codec preference 4
g711alaw preferably 5 codec
!
controller E1 1/0/1
No.-CRC4 framing
time intervals PRI - Group 1-31
!
Voice-port 1/0/1:15
Capitalization in the carrier speech
MGCP
MGCP behavior considered range tgcp only
MGCP comedia-role behavior no
disable the behavior MGCP comedia-check-media-src
disable the behavior of MGCP comedia-sdp-force
!
profile MGCP default
!
!
!
!
Dial-peer voice 3 pots
destination-style 9...
port 0/1/1:15
!
Dial-peer voice 1 voip
destination-model 1...
session target ipv4:10.15.108.242
!
!
!
!
access controller
Shutdown
!
!
phone service
MAX conferences 8-6 win
transfer full-consult system
!
!

OK, lets delete all of your peers-Dial and start from scratch.

For testing purposes, we will test with specific numbers.  We can get more general thereafter.  Try the following tips for incoming traffic.

voice pots Dial-peer 10

/ / DESC # incoming Iris PBX #.

incoming called-number.

direct line to inside
Forward-digits all the

Then lets try this for your Call Manager Dial-Peer

Dial-peer voice voip 20

destination-model 501

session protocol sipv2

session target ipv4: CUCM_IP

If please retest and propose some outputs fresh showing.

debugging ccapi inout voice

ccsip debug messages

Debug ISDN q931

Tags: Cisco Support

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