با سلام
دوستان من می خوام 5300 رو تبدیل کنم به یه Voice gateway .
تو اینترنت سرچ کردم به نتیجه مطلوبی نرسیدم .
فقط می دونم باید کارت vfc داشته باشه و ios که Voip رو ساپورت کنه ممنون میشم اگه کسی تو این ضمینه اطلاعاتی داره راهنمایی کنه.
Printable View
با سلام
دوستان من می خوام 5300 رو تبدیل کنم به یه Voice gateway .
تو اینترنت سرچ کردم به نتیجه مطلوبی نرسیدم .
فقط می دونم باید کارت vfc داشته باشه و ios که Voip رو ساپورت کنه ممنون میشم اگه کسی تو این ضمینه اطلاعاتی داره راهنمایی کنه.
فکر کنم همین باشه
[URL="http://davidlinmsn.spaces.live.com/blog/cns%2152BEC44D7E074AA6%21130.entry"]configure a Cisco SIP Gateway(AS5300 5300) - Windows Live[/URL]
[B]
How do I configure a Cisco SIP Gateway (AS5300 5300) to work with VOCAL
[/B]
The following demonstrates:
1) accessing the configuration of a Cisco AS5300.
2) the configuration of our internal trial system
/* The Cisco AS5300 is located behind a firewall and uses a private
* (non-routable) address. In the configuration shown, we have two T1
* lines (one PRI and one CAS), but only the PRI is being used. The T1
* is configured at the Central Office to serve DIDs 408.321.5100-5199.
* The called numbers on the PRI are presented to us as 11 digit
* numbers of the form 140832151xx.
* For calls we make, the called numbers does NOT need a '9'.
*/
[eckelcu@eckelcu-lnx ~/config] telnet vvs-sip5300-0
Trying 172.19.175.208...
Connected to vvs-sip5300-0.cisco.com.
Escape character is '^]'.
User Access Verification
Password:
vvs-sip5300-0>enable
Password:
vvs-sip5300-0#show running-config
Building configuration...
/* Several of the entries in the configuration are simply there by
* default. I have annotated only those which are in my opinion both
* important to us and non obvious.
*/
Current configuration : 5002 bytes
!
! No configuration change since last restart
!
version 12.1
no service single-slot-reload-enable
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname vvs-sip5300-0
!
no logging buffered
no logging buffered
logging rate-limit console 10 except errors
no logging console /* this was added to enable viewing logging messages
* telnet sessions instead of from the serial console
* port.
*/
enable secret 5 $1$a5Y1$.j0ueQVChOy0ZtAC4gozQ.
enable password **************
!
!
!
resource-pool disable
!
clock timezone GMT -8
clock calendar-valid
ip subnet-zero
no ip finger
ip domain-list cisco.com
ip domain-name cisco.com
ip name-server 171.69.2.133
!
ip dhcp-server 172.19.174.41
mgcp modem passthrough voaal2 mode nse
no mgcp timer receive-rtcp
isdn switch-type primary-5ess /* type of PRI lines provided by CO */
isdn voice-call-failure 0
call rsvp-sync
!
!
!
!
!
fax interface-type vfc
mta receive maximum-recipients 0
!
!
!
controller T1 0 /* configuration of PRI line, must match that from CO */
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
description T1 PRI: 408.526.9233: DIDs 408.321.51[00-99]
!
controller T1 1 /* not really used but shows how to set up a CAS T1 */
shutdown /* because we are not using it */
framing esf
clock source line secondary 1
linecode b8zs
ds0-group 1 timeslots 1-24 type e&m-fgb
cas-custom 1
description T1 CAS: 408.232.6380-6399
!
controller T1 2
shutdown /* because we are not using it */
framing esf
linecode b8zs
pri-group timeslots 1-24
!
controller T1 3
shutdown /* because we are not using it */
framing esf
linecode b8zs
pri-group timeslots 1-24
!
/* this rule is needed for outgoing calls (internal SIP phone to PSTN)
* to change the calling number we give to the CO
* for ourselves from the 4 digit extension we use interanlly to a 10
* digit number used for caller-id services.
* For example, it changes ids of the form 51xx to 40832151xx.
* It also changes the type of number from 'unknown' to 'national'
*/
translation-rule 408321
Rule 1 51% 40832151 unknown national
!
/* this rule is needed for outgoing calls to change the called number
* from international to national if the called number begins with a 1.
*/
translation-rule 1
Rule 1 1% 1 international national
!
/* this rule is needed for outgoing calls to change the called number
* from unknown to international if the called number begins with 011.
*/
translation-rule 20
Rule 1 011% 011 unknown international
!
!
!
interface Ethernet0 /* 10MB ethernet connection is not used */
description Lab System Connection
no ip address
no ip mroute-cache
shutdown
!
interface Serial0:23 /* D-channel for the PRI on controller T1 0 */
no ip address
ip mroute-cache
dialer-group 1
isdn switch-type primary-5ess
isdn incoming-voice modem
isdn disconnect-cause 1
no cdp enable
!
interface Serial2:23 /* D-channel of PRI on controller T1 2, not used */
no ip address
ip mroute-cache
dialer-group 1
isdn switch-type primary-5ess
isdn incoming-voice modem
isdn guard-timer 3000
isdn T203 10000
no cdp enable
!
interface Serial3:23 /* D-channel of PRI on controller T1 3, not used */
no ip address
ip mroute-cache
dialer-group 1
isdn switch-type primary-5ess
isdn incoming-voice modem
isdn guard-timer 3000
isdn T203 10000
no cdp enable
!
interface FastEthernet0 /* 100MB ethernet interface */
description Internal Trial System Ethernet Connection
ip address 172.19.175.208 255.255.254.0
no ip mroute-cache
duplex auto
speed auto
!
ip default-gateway 172.19.174.1
ip nat translation timeout never
ip nat translation tcp-timeout never
ip nat translation udp-timeout never
ip nat translation finrst-timeout never
ip nat translation syn-timeout never
ip nat translation dns-timeout never
ip nat translation icmp-timeout never
ip classless
ip route 0.0.0.0 0.0.0.0 172.19.174.1 /* needed for routing RTP packets,
* also need to turn on ip
* routing, although it is not
* specifically shown in the
* configuration file.
*/
no ip http server
!
no logging trap
dialer-list 1 protocol ip permit
dialer-list 1 protocol ipx permit
!
!
voice-port 0
!
voice-port 1:1
timeouts interdigit 4
!
voice-port 2
!
voice-port 3
!
/* dial peer for our SIP phones with extensions 51xx, primary */
dial-peer voice 51 voip
application session.t.old /* need to use session.t.old. instead of
* session in order for PRI to SIP mapping to
* work correcly.
*/
destination-pattern 51..$
progress_ind setup enable 3 /* specifies that ringback should be
* generated by the gateway when no
* progress indicator is specified by the
* caller (CO).
*/
session protocol sipv2
session target sip-server /* primary PSTN gateway marshal */
codec g711ulaw
no vad
!
/* dial peer for our SIP phones with extensions 51xx, backup */
dial-peer voice 511 voip
preference 1 /* preference is from 0-9 with 0 being the highest and the
* default, set to 1 to give this dial-peer lower
* precedence than dial-peer 51 (preference 0)
*/
application session.t.old
destination-pattern 51..$
progress_ind setup enable 3
session protocol sipv2
session target ipv4:172.19.175.201 /* backup PSTN gateway marshal */
codec g711ulaw
no vad
!
/* this dial-peer exists to handle calls in which the caller provides no
* caller id information. In this case, whatever is specified as the
* 'destination-pattern for the first dial-peer for the port (in this
* case port 0) is used as the callers id.
*/
dial-peer voice 99 pots
preference 1
application session.t.old
destination-pattern CallerId-Blocked
translate-outgoing calling 408321
no digit-strip
direct-inward-dial
port 0
!
/* dial-peer for international calls */
dial-peer voice 11 pots
application session.t.old
destination-pattern 011.
translate-outgoing called 20
direct-inward-dial
port 0
!
/* dial-peer for long distance (national) calls */
dial-peer voice 10 pots
application session.t.old
destination-pattern 1..........$
translate-outgoing called 1
no digit-strip
direct-inward-dial
port 0
!
/* dial-peer for local calls */
dial-peer voice 7 pots
application session.t.old
destination-pattern .......$
translate-outgoing calling 408321
no digit-strip
direct-inward-dial
port 0
!
/* dial-peer for calling 311,411,511,611,711,811 */
dial-peer voice 311 pots
preference 1
application session.t.old
destination-pattern [3-8]11$
translate-outgoing calling 408321
no digit-strip
direct-inward-dial
port 0
!
/* dial-peer for calling local operator */
dial-peer voice 1 pots
application session.t.old
destination-pattern 0$
translate-outgoing calling 408321
direct-inward-dial
port 0
!
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
timers invite-wait-100 1000
sip-server ipv4:172.19.175.204
!
!
line con 0
logging synchronous
transport input none
line aux 0
line vty 0 4
password *********************
login
!
scheduler interval 1000