hola
el flujo de llamada seria CUCM<---> SIP TRUNK<--->IOS gateway<-->SIP TRUNK PSTN
en el CUCM solo tengo configurado la troncal SIP apuntando a la interfaz de telefonoia del IOS Gateway,asi mismo tengo un route pattern con una coincidencia a mi celular y seleccionando la troncal SIP configurada.
sigo pensando que lo que esta mal es mi config. en el IOS gateway
Building configuration...
Current configuration : 32721 bytes
!
! Last configuration change at 10:35:56 Mexico Wed Feb 21 2018 by fvicente
! NVRAM config last updated at 17:24:06 Mexico Tue Feb 20 2018 by jamorales
! NVRAM config last updated at 17:24:06 Mexico Tue Feb 20 2018 by jamorales
version 15.2
service nagle
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service internal
service sequence-numbers
!
hostname Rt-Main_Mexico
!
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.152-4.M3.bin
boot-end-marker
!
!
!
aaa new-model
!
!
aaa group server radius iRadius
server-private 192.168.39.52 timeout 1
server-private 192.168.9.52 timeout 1
!
aaa authentication password-prompt Password:
aaa authentication username-prompt User:
aaa authentication login default group iRadius local
aaa authorization exec default local group iRadius if-authenticated
!
!
!
!
!
aaa session-id common
clock timezone Mexico -6 0
clock summer-time Mexico recurring 1 Sat Apr 22:00 last Sat Oct 22:00
network-clock-participate wic 1
!
no ip source-route
ip cef
!
!
!
!
!
!
voice-card 0
codec complexity high
dspfarm
dsp services dspfarm
!
!
voice rtp send-recv
!
voice service voip
ip address trusted list
ipv4 200.94.59.xxx
ipv4 200.94.26.xxx
ipv4 200.94.26.xxx
ipv4 200.94.26.xxx
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
modem passthrough nse codec g711ulaw
sip
bind control source-interface Loopback246
bind media source-interface Loopback246
session transport tcp
registrar server
outbound-proxy ipv4:200.94.59.xxx:5060
associate registered-number 4421538xxx
registration passthrough
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
voice class h323 1
h225 timeout tcp establish 3
call start fast
call preserve
!
voice class sip-profiles 1
request REINVITE sdp-header Audio-Attribute modify "sendonly" "sendrecv"
request REINVITE sdp-header Audio-Attribute modify "sendonly" "sendrecv"
!
!
voice iec syslog
voice register global
system message SIP SRST
max-dn 45
max-pool 45
!
voice register pool 1
id network 192.168.7.0 mask 255.255.255.0
codec g711ulaw
!
!
!
!
!
!
application
service clid_authen_collect
param uid-len 4
param pin-len 4
!
!
license udi pid CISCO2921/K9 sn FTX1630AJHL
license accept end user agreement
license boot module c2900 technology-package securityk9
license boot module c2900 technology-package uck9
hw-module ism 0
!
hw-module pvdm 0/0
!
hw-module sm 1
!
!
!
!
controller E1 0/1/0
framing NO-CRC4
line-termination 75-ohm
ds0-group 1 timeslots 1-11 type r2-digital r2-compelled ani
cas-custom 1
country telmex
category 2
answer-signal group-b 1
caller-digits 4
dnis-digits min 1 max 16
ani-digits min 0 max 16
groupa-callerid-end
description Link to PSTN
!
vlan 3,41-43,60-64,80
!
vlan 333
name Native
!
!
!
!
!
interface Loopback222
ip address 192.168.222.1 255.255.255.0
!
interface Loopback246
description GW-Mex-Srst
ip address 192.168.246.5 255.255.255.252
ip access-group firewall in
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.246.5
!
!
voice-port 0/1/0:1
echo-cancel coverage 32
cptone MX
timeouts call-disconnect 3
timeouts wait-release 2
!
voice-port 0/3/0
input gain 6
output attenuation 0
timing hookflash-out 500
connection plar 1144
description Tellular
!
voice-port 0/3/1
!
!
!
!
!
!
mgcp profile default
!
sccp local Loopback246
sccp ccm 192.168.7.3 identifier 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register transcoder
associate profile 1 register conference
!
dspfarm profile 2 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g729br8
codec g729r8
maximum sessions 2
associate application SCCP
!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
!
!
dial-peer voice 5000 voip
description CUCM
destination-pattern 5...
session target ipv4:192.168.7.3
dtmf-relay h245-alphanumeric
!
dial-peer voice 5010 voip
description CUCM
destination-pattern 1...
session target ipv4:192.168.7.3
codec g711alaw
!
dial-peer voice 20000 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind control source-interface Loopback246
voice-class sip bind media source-interface Loopback246
dtmf-relay rtp-nte
no vad
!
dial-peer voice 5011 voip
description CUCM
destination-pattern 3...
session target ipv4:192.168.7.3
dtmf-relay h245-alphanumeric
!
dial-peer voice 5012 voip
description CUCM
destination-pattern 1144
session protocol sipv2
session target ipv4:192.168.7.3
dtmf-relay h245-alphanumeric
!
dial-peer voice 30000 voip
description **Incoming Call from CUCM**
session protocol sipv2
session target ipv4:192.168.7.3
incoming called-number .
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 15 voip
description MARCACION CELULAR NACIONAL SIP
destination-pattern 0455528077xxx
session protocol sipv2
session target sip-server
incoming called-number .
no voice-class sip associate registered-number
no voice-class sip registration passthrough
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 10000 pots
description incoming calls from PSTN
incoming called-number .
direct-inward-dial
!
!
num-exp 8xxx 1144
num-exp 4421538xxx 1144
gateway
timer receive-rtp 1200
!
sip-ua
credentials username 4421538xxx password 7 xxx realm ascs.alestravoip.com
authentication username st.01467645.01 password 7 xxx
no remote-party-id
retry invite 2
retry register 10
retry options 2
timers connect 100
registrar ipv4:200.94.26.xxx:5060 expires 3600 auth-realm ascs.alestravoip.com
sip-server ipv4:200.94.26.xxx:5060
host-registrar
!
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 192.168.246.5 port 2000
max-ephones 50
max-dn 50 octo-line
system message primary SRST Fallback Active
date-format dd-mm-yy
!
