Hi,
ì`m a little bit new to the cme world. I have purchased a nice 2811 Router and want now to
test this in my lab. I have a external voice provider. This is what i have configured at the moment and incoming calls work but outgoing not:
voice translation-rule 3
rule 1 /^00/ /+/ type any international
!
!
voice translation-profile SIP-Outgoing
translate called 3
!
!
dial-peer voice 1 voip
description "INCOMING"
destination-pattern 004912345678912 <my international number>
voice-class codec 1
session protocol sipv2
session target dns:sip.provider.de
incoming called-number .T
no vad
!
dial-peer voice 4 voip
description "OUTGOING"
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target dns:sip.provider.de
dtmf-relay rtp-nte
no vad
!
voice call send-alert
voice call convert-discpi-to-prog
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol pass-through g711ulaw
h323
modem passthrough nse codec g711ulaw
sip
bind control source-interface FastEthernet0/0
session transport tcp
registrar server expires max 600 min 60
redirect contact order best-match
!
voice class codec 1
codec preference 1 g711ulaw
sip-ua
authentication username
<username>
password 7 <password>
no remote-party-id
retry invite 2
retry register 10
retry options 1
timers connect 100
registrar dns:<sip registrar> expires 3600
sip-server dns:<sip server>
host-registrar
dial-peer voice 1 voip
description SIP TRUNK TO PROVIDER
destination-pattern <my number in internataional format> extension-length 4 extension-pattern 1001
session protocol sipv2
session target dns:<sip server>
incoming called-number .T
voice-class codec 1
no vad
telephony-service
no auto-reg-ephone
max-ephones 20
max-dn 40
max-conferences 8 gain -6
transfer-system full-consult
---
incoming (works)
---
The Call Setup Information is:
Call Control Block (CCB) : 0x4A63D688
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +49xxxxxxxxxxx (my mobile, 11 digits)
Called Number : 0049xxxxxxxxxxx (my home, 11 digits)
Source IP Address (Sig ): 192.168.5.99
Destn SIP Req Addr:Port : <sip provider>:5060
Destn SIP Resp Addr:Port : <sip provieder>:5060
Destination Name : <sip provider>
*Sep 30 22:01:34.571: //3/962E68E78003/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.5.99
Source IP Port (Media): 18846
Destn IP Address (Media): <provider ip>
Destn IP Port (Media): 40064
Orig Destn IP Address:Port (Media): [ - ]:0
---
outgoing (works not)
---
The Call Setup Information is:
Call Control Block (CCB) : 0x4A63D688
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : +49xxxxxxxxxxx (my home, 11 digits)
Called Number : +49xxxxxxxxxxx (my mobile, 11 digits)
Source IP Address (Sig ): 192.168.5.99
Destn SIP Req Addr:Port : <sip provider>:5060
Destn SIP Resp Addr:Port : <sip provider>:5060
Destination Name : <sip provider>
Sep 30 22:13:21.864: //20/33C2F5638033/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.5.99
Source IP Port (Media): 18180
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
i have tried different outgoing formats (translation rules). I don`t know why i can`t dial out. Any suggestions?
