09-14-2015 09:36 PM - edited 03-17-2019 04:18 AM
Dear,
I am using 2800 router as gateway for PRI and Analogue trunks.
My topology is..
PRI---->Cisco 2800------>FreepBX
I have configured my router for ISDN PRI and incoming calls are coming to proper extensions as per DIDs.But when i dial outside from freepbx its log show no channel available and outgoing call dosnt makes
Kindly help
Current configuration : 3654 bytes
!
! Last configuration change at 12:19:04 UTC Mon Sep 14 2015
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$ml.W$gFgeC4R3cZwYLfDULvXx81
enable password
!
no aaa new-model
!
resource policy
!
network-clock-participate wic 1
network-clock-select 1 E1 0/1/0
ip subnet-zero
!
!
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.200.1 192.168.200.200
ip dhcp excluded-address 192.168.200.240 192.168.200.254
!
ip dhcp pool cisco
network 192.168.200.0 255.255.255.0
default-router 192.168.200.41
option 150 ip 192.168.200.41
!
!
ip name-server 192.168.200.170 //ip of Freepbx
isdn switch-type primary-net5
!
voice-card 0
no dspfarm
!
!
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
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 1200 min 300
!
!
voice register pool 1
!
!
controller E1 0/1/0 //connected to PRI
pri-group timeslots 1-31
!
controller E1 0/1/1
!
!
!
interface GigabitEthernet0/0
ip address 192.168.200.41 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/1/0:15
no ip address
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
!
ip classless
!
!
no ip http server
no ip http secure-server
!
!
!
tftp-server flash:apps70.9-2-1TH1-13.sbn
tftp-server flash:cnu70.9-2-1TH1-13.sbn
tftp-server flash:cvm70sip.9-2-1TH1-13.sbn
tftp-server flash:dsp70.9-2-1TH1-13.sbn
tftp-server flash:jar70sip.9-2-1TH1-13.sbn
tftp-server flash:SIP70.9-2-1S.loads
tftp-server flash:term70.default.loads
tftp-server flash:term71.default.loads
!
control-plane
!
!
voice-port 0/0/0
cptone PK
timing hookflash-out 50
connection plar opx 4000
!
voice-port 0/0/1
!
voice-port 0/1/0:15
!
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
no mgcp explicit hookstate
!
!
dial-peer voice 2 voip //voip dialpeer for freepbx
destination-pattern 4...
session protocol sipv2
session target ipv4:192.168.200.170
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 100 pots //PRi pots incoming
numbering-type unknown
incoming called-number .
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 3 voip
destination-pattern 6...
session target ipv4:192.168.200.41
codec g711ulaw
!
dial-peer voice 10 pots //PRi pots outgoing
destination-pattern 9T
port 0/1/0:15
forward-digits all
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:192.168.200.170
!
!
telephony-service
load 7970 term70.default
max-ephones 20
max-dn 30
ip source-address 192.168.200.41 port 2000
auto assign 1 to 10
auto assign 1 to 10 type 7970
create cnf-files version-stamp 7960 Sep 14 2015 09:43:40
max-conferences 8 gain -6
!
!
ephone-dn 1
number 6001
name shoaib
!
!
ephone-dn 2
number 6002
name hamza
!
!
ephone-dn 3
number 6003
name Amir
!
!
ephone 1
mac-address 18A9.05ED.99C0
type CIPC
button 1:1
!
!
!
ephone 2
mac-address 5CF9.DDE8.BC20
type CIPC
button 1:2
!
!
!
ephone 3
mac-address 001F.6C7F.5DF5
type 7970
button 1:3
!
!
line con 0
line aux 0
line vty 0 4
password *****
login
line vty 5 15
password *****
login
!
scheduler allocate 20000 1000
ntp clock-period 17179878
ntp server 192.168.200.11 key 0
!
end
Show isdn status is
Global ISDN Switchtype = primary-net5
ISDN Serial0/1/0:15 interface
dsl 0, interface ISDN Switchtype = primary-net5
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
CCB:callid=12, sapi=0, ces=0, B-chan=1, calltype=VOICE
Active dsl 0 CCBs = 1
The Free Channel Mask: 0xFFFF7FFE
Number of L2 Discards = 0, L2 Session ID = 21
Total Allocated ISDN CCBs = 1
show voice port summary
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
========= == ============ ===== ==== ======== ======== ==
0/0/0 -- fxo-ls up dorm idle on-hook y
0/0/1 -- fxo-ls up dorm idle on-hook y
0/1/0:15 01 isdn-voice up dorm none none y
0/1/0:15 02 isdn-voice up dorm none none y
0/1/0:15 03 isdn-voice up dorm none none y
0/1/0:15 04 isdn-voice up dorm none none y
0/1/0:15 05 isdn-voice up dorm none none y
0/1/0:15 06 isdn-voice up dorm none none y
0/1/0:15 07 isdn-voice up dorm none none y
0/1/0:15 08 isdn-voice up dorm none none y
0/1/0:15 09 isdn-voice up dorm none none y
0/1/0:15 10 isdn-voice up dorm none none y
0/1/0:15 11 isdn-voice up dorm none none y
0/1/0:15 12 isdn-voice up dorm none none y
0/1/0:15 13 isdn-voice up dorm none none y
0/1/0:15 14 isdn-voice up dorm none none y
0/1/0:15 15 isdn-voice up dorm none none y
0/1/0:15 17 isdn-voice up dorm none none y
0/1/0:15 18 isdn-voice up dorm none none y
0/1/0:15 19 isdn-voice up dorm none none y
0/1/0:15 21 isdn-voice up dorm none none y
0/1/0:15 22 isdn-voice up dorm none none y
0/1/0:15 23 isdn-voice up dorm none none y
0/1/0:15 24 isdn-voice up dorm none none y
0/1/0:15 25 isdn-voice up dorm none none y
0/1/0:15 26 isdn-voice up dorm none none y
0/1/0:15 27 isdn-voice up dorm none none y
0/1/0:15 28 isdn-voice up dorm none none y
0/1/0:15 29 isdn-voice up dorm none none y
0/1/0:15 30 isdn-voice up dorm none none y
0/1/0:15 31 isdn-voice up dorm none none y
50/0/1 1 efxs up up on-hook idle y
50/0/2 1 efxs up dorm on-hook idle y
50/0/3 1 efxs up up on-hook idle y
09-15-2015 04:35 AM
Hi,
First remove dialpeer 3 as it isn't needed. Then remove the command 'forward-digits all' from dialpeer 10.
After this make a test call from FreePBX and share the output of debug ccsip mess and debug isdn q931
09-15-2015 08:04 AM
Just adding to Mohammed's point:
How are you dialing out from the FreePBX? I dont see any VoIP dial-peer to catch "all" for configured for the PSTN on the router, that could be the reason.
Secondly, please share the output from "debug voice dialpeer inout" while you are trying to dial out from the FreePBX. It will explain what is going on in terms of dial-peer.
HTH
10-30-2015 03:27 AM
Hi Wilson
Thaks for comments
Can you plese elaborate and text the commands
10-30-2015 05:04 AM
What Wilson meant to say is there may not be seen any VoIP dial peer matching for the calls from FreePBX to gateway, hence share the output of follwowing commands from gateway;
debug ccsip messages
debug voice ccapin inout
debug isdn q931
- Vivek
10-30-2015 10:39 AM
Dear Vivek,
I can't provide with the traces now bcz am not at my workplace now however i will manage it ASAP.
secondly I am confused with comments "What Wilson meant to say is there may not be seen any VoIP dial peer matching for the calls from FreePBX to gateway
If I am not wrong i think i have created the dialpeer for calls from FreePBX to gateway named as dial-peer 10
dial-peer voice 10 pots //PRi pots outgoing
destination-pattern 9T //my extensions inside use 9+ number to dial outside
port 0/1/0:15 //use this port/pri for outgoing
forward-digits all
Plz PLz comment if this not the outgoing dial peer and plz provide me with exact code to call from pbx to gateway and then to pstn via PRI
10-30-2015 12:22 PM
Yes, you're right. This is outbound dial-peer to make outgoing calls through PSTN but you also need inbound dial-peer for calls from freepbx to gateway, something like below;
dial-peer voice 200 voip
incoming called-number .
session protocol sipv2
codec g711ulaw
Adding above dial peer should fix your issue. If not, I will be requiring the debugs output which I mentioned earlier.
Also let me know that you've configured forward-digits all command under dial-peer voice 10 pots, does service provider wants prefix (9)? Please crossverify and ignore my comment, if 9 is a part of original called number.
- Vivek
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