درخواست ios سیسکو 5300 جهت راه اندازی voip

[ms_804]

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.

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موضوعات مشابه:

• در خواست کمک در مورد راه اندازی Voip برای یک شرکت در 6 شهر ؟
• راه اندازی voip از صفر تا صد با تجهیزات سیسکو
• اینترنت هوشمند و سیسکو 5300 و کمپرس سازی پکت ها
• سوالاتی در مورد مشکل VOIP در 5300

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[mavrick]

فکر کنم همین باشه

configure a Cisco SIP Gateway(AS5300 5300) - Windows Live

How do I configure a Cisco SIP Gateway (AS5300 5300) to work with VOCAL


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