我正在尝试将我公司的电话系统从 Cisco 的 CUCM 转移到星号解决方案。
我已经设置了 PBXinaflash 服务器，并在 www.vitelity.com 上设置了一个帐户。我能够让它工作。我可以拨打/接听电话，并且它连接到 IVR。
接下来，我购买了带有 PRI 卡的 Cisco 2821 ISR，作为我们当前 VOIP 网关的备份。我能够在其上加载我们当前的配置，并且在插入 PRI 线路后，它就可以与我们的 Cisco CUCM 配合使用。
之后，我用谷歌搜索了如何重新配置 2821。我下载了思科的文档，并在谷歌上找到了一些示例。第一次尝试后，我能够拨打电话。我能够呼叫 IVR 并让它识别我拨打的数字。其他电话的来电 ID 显示了正确的来电者 ID。问题是无法接通来电。到目前为止，我已经让它发出咔嗒声、拨号音、全线忙音，并且没有任何真正的电话响铃。
我的目标是让我们的 Aterisk IVR 接听任何电话号码的所有来电。我们有大约 150 个 DID，但只有 4 个真正用于任何事情。
我的问题是，有人能用这种设置吗？有可能吗？你能和我分享一下你的配置吗？
我认为问题出在 2821 的配置或星号中的中继配置上。
[思科 2821 ISR ~ 10.1.10.252]
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname asterisk
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
clock timezone GMT -6
network-clock-participate wic 1
network-clock-select 1 T1 0/1/0
!
dot11 syslog
ip source-route
!
ip cef
!
no ipv6 cef
!
multilink bundle-name authenticated
!
isdn switch-type primary-ni
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
signaling forward unconditional
fax protocol cisco
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
!
voice-card 0
dspfarm
!
archive
log config
hidekeys
!
controller T1 0/1/0
pri-group timeslots 1-24
!
interface GigabitEthernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/1
ip address 10.1.10.252 255.255.255.0
duplex auto
speed auto
!
interface Serial0/1/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn calling-number 3167945555
isdn supp-service name calling
no cdp enable
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.1.10.253
no ip http server
no ip http secure-server
!
control-plane
!
voice-port 0/1/0:23
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
no mgcp explicit hookstate
!
dial-peer voice 1 pots
numbering-type unknown
destination-pattern .
direct-inward-dial
port 0/1/0:23
forward-digits all
!
dial-peer voice 2 voip
numbering-type unknown
destination-pattern .
session protocol sipv2
session target ipv4:10.1.10.7:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
!
gateway
timer receive-rtp 1200
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:10.1.10.7
!
line con 0
transport output telnet
line aux 0
transport output telnet
line vty 0 4
login
transport input telnet
line vty 5 15
login
transport input telnet
!
scheduler allocate 20000 1000
end
[星号主干 ~ 10.1.10.7]
Outgoing Settings
Trunk Name:
cisco2821
PEER Details:
type=friend
qualify=yes
nat=no
insecure=very
host=10.1.10.252
dtmfmode=rfc2833
disallow=all
context=from-internal
allow=ulaw&alaw
Incoming Settings
USER Context:
from-internal
USER Details:
type=friend
qualify=yes
nat=no
host=10.1.10.252
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw&alaw
答案1
好的，我让它工作了，我需要学习如何使用 2821 ISR 上的调试日志。感谢 freepbx.org 的 SkykingOH 对这个项目的帮助。
debug isdn q931
debug voip dialpeer inout
debug voip dialpeer detail
debug ccsip states
debug ccsip info
debug ccsip calls
debug ccsip events
debug ccsip messages
我也简化了 [asterisk trunk ~ 10.1.10.7]
PEER Details:
type=friend
qualify=yes
nat=no
insecure=port,invite
host=10.1.10.252
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw
2821 的配置
version 12.4
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
no service password-encryption
!
hostname asterisk
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
enable password ********
!
no aaa new-model
clock timezone GMT -6
network-clock-participate wic 1
network-clock-select 1 T1 0/1/0
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
ip domain name *********.com
ip name-server 10.0.0.21
ip name-server 10.0.0.22
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-ni
!
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
signaling forward unconditional
fax protocol cisco
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
!
voice-card 0
dspfarm
!
archive
log config
hidekeys
!
controller T1 0/1/0
pri-group timeslots 1-24
!
interface GigabitEthernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/1
ip address 10.1.10.252 255.255.255.0
duplex auto
speed auto
!
interface Serial0/1/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn calling-number 316********
isdn supp-service name calling
no cdp enable
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.1.10.253
no ip http server
no ip http secure-server
!
!
ip dns server
!
logging trap warnings
logging 10.0.0.14
!
control-plane
!
voice-port 0/1/0:23
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
no mgcp explicit hookstate
!
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
port 0/1/0:23
forward-digits all
!
dial-peer voice 2 voip
destination-pattern ^[2-9]......$
session protocol sipv2
session target ipv4:10.1.10.7:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 9001 pots
preference 1
destination-pattern .
port 0/1/0:23
forward-digits all
!
!
gateway
timer receive-rtp 1200
!
sip-ua
no remote-party-id
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:10.1.10.7
!
!
!
line con 0
transport output telnet
line aux 0
transport output telnet
line vty 0 4
login
transport input telnet
line vty 5 15
login
transport input telnet
!
scheduler allocate 20000 1000
end
使用调试后，我了解到我的电信公司（COX）正在向我发送 7 位中继号码。