Hi All,
We're replacing our voip sytem from Asterisk to Cisco CME but we're experiencing problems to configure the sip trunk to use an external sip system do our internal calls.
in asterisk we have this configuration:
type=peer
host=201.22.170.156
username=856098845
secret=1010
fromuser=788430101
allow=alaw&ulaw
canreinvite=yes
context=from-trunk
insecure=very
856098845:1010@201.22.170.156/856098845
and everything is working fine... on cisco we did these configurations (for testing purposes):
Building configuration...
Current configuration : 7995 bytes
!
version 12.4
!
!
!
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
sip
registrar server expires max 3600 min 600
!
!
!
voice class codec 1
codec preference 1 g711ulaw
!
!
voice translation-rule 1
rule 1 /^89(.*)/ /\1/
!
!
voice translation-profile VOIP-Outgoing
translate calling 1
!
!
voice-card 0
!
dial-peer voice 1 voip
description Saida Tmais
translation-profile outgoing VOIP-Outgoing
destination-pattern .T
redirect ip2ip
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711alaw
no vad
!
!
sip-ua
credentials username 856098845 password 1010 realm sip.provider.com.br
authentication username 856098845 password 1010 realm sip.provider.com.br
no remote-party-id
retry invite 5
retry response 3
retry bye 5
retry cancel 5
retry prack 5
retry notify 4
retry register 5
retry options 5
timers connect 100
timers connection aging 30
timers register 600
registrar dns:sip.provider.com expires 3600
sip-server dns:sip.provider.com:5060
notify telephone-event max-duration 3000
host-registrar
!
!
!
telephony-service
em logout 0:0 0:0 0:0
max-ephones 20
max-dn 20
ip source-address 192.168.0.3 port 2000
calling-number initiator
timeouts interdigit 2
system message NetConnection
load 7960-7940 term42
load 7965 term65.default
max-conferences 4 gain -6
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Dec 10 2010 17:21:35
!
!
ephone-dn 1
number 230
label User 1
!
ephone 1
device-security-mode none
mac-address ECC7.9875.E65A
button 1:1
We're not being able to call using the external sip service, I think I missed something on configuration.
Does anyone know what am I doing wrong?
thanks,
Ricardo
We're replacing our voip sytem from Asterisk to Cisco CME but we're experiencing problems to configure the sip trunk to use an external sip system do our internal calls.
in asterisk we have this configuration:
type=peer
host=201.22.170.156
username=856098845
secret=1010
fromuser=788430101
allow=alaw&ulaw
canreinvite=yes
context=from-trunk
insecure=very
856098845:1010@201.22.170.156/856098845
and everything is working fine... on cisco we did these configurations (for testing purposes):
Building configuration...
Current configuration : 7995 bytes
!
version 12.4
!
!
!
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
sip
registrar server expires max 3600 min 600
!
!
!
voice class codec 1
codec preference 1 g711ulaw
!
!
voice translation-rule 1
rule 1 /^89(.*)/ /\1/
!
!
voice translation-profile VOIP-Outgoing
translate calling 1
!
!
voice-card 0
!
dial-peer voice 1 voip
description Saida Tmais
translation-profile outgoing VOIP-Outgoing
destination-pattern .T
redirect ip2ip
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711alaw
no vad
!
!
sip-ua
credentials username 856098845 password 1010 realm sip.provider.com.br
authentication username 856098845 password 1010 realm sip.provider.com.br
no remote-party-id
retry invite 5
retry response 3
retry bye 5
retry cancel 5
retry prack 5
retry notify 4
retry register 5
retry options 5
timers connect 100
timers connection aging 30
timers register 600
registrar dns:sip.provider.com expires 3600
sip-server dns:sip.provider.com:5060
notify telephone-event max-duration 3000
host-registrar
!
!
!
telephony-service
em logout 0:0 0:0 0:0
max-ephones 20
max-dn 20
ip source-address 192.168.0.3 port 2000
calling-number initiator
timeouts interdigit 2
system message NetConnection
load 7960-7940 term42
load 7965 term65.default
max-conferences 4 gain -6
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Dec 10 2010 17:21:35
!
!
ephone-dn 1
number 230
label User 1
!
ephone 1
device-security-mode none
mac-address ECC7.9875.E65A
button 1:1
We're not being able to call using the external sip service, I think I missed something on configuration.
Does anyone know what am I doing wrong?
thanks,
Ricardo