Accepted Solutions
Sorry, I had the bind statements on the dial peers incorrect. This should work.
voice service voip
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
!
dial-peer voice 9000 voip
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind control source-interface GigabitEthernet0/2
!
dial-peer voice 100 voip
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind control source-interface GigabitEthernet0/2
I've set the global bind to the interface you say is used by the phones and the dial peer bind to the interface used for your SIP circuit.
Yes I have, it shows the wrong time since you haven't configured the time zone in you CME configuration. I don't know what TZ you're in, so you'll need to select the appropriate time-zone number to set under telephony-service.
telephony-service
time-zone <number>
time-format 24
time-zone ?
<1-56> select timezone name used by IP phones (offset in minutes)
1 Dateline Standard Time -720
2 Samoa Standard Time -660
3 Hawaiian Standard Time -600
4 Alaskan Standard/Daylight Time -540
5 Pacific Standard/Daylight Time -480
6 Mountain Standard/Daylight Time -420
7 US Mountain Standard Time -420
8 Central Standard/Daylight Time -360
9 Mexico Standard/Daylight Time -360
10 Canada Central Standard Time -360
11 SA Pacific Standard Time -300
12 Eastern Standard/Daylight Time -300
13 US Eastern Standard Time -300
14 Atlantic Standard/Daylight Time -240
15 SA Western Standard Time -240
16 Newfoundland Standard/Daylight Time -210
17 E. South America Standard/Daylight Time -180
18 SA Eastern Standard Time -180
19 Mid-Atlantic Standard/Daylight Time -120
20 Azores Standard/Daylight Time -60
21 GMT Standard/Daylight Time +0
22 Greenwich Standard Time +0
23 W. Europe Standard/Daylight Time +60
24 GTB Standard/Daylight Time +60
25 Egypt Standard/Daylight Time +120
26 E. Europe Standard/Daylight Time +120
27 Romance Standard/Daylight Time +120
28 Central Europe Standard/Daylight Time +120
29 South Africa Standard Time +120
30 Jerusalem Standard/Daylight Time +120
31 Saudi Arabia Standard Time +180
32 Russian Standard/Daylight Time +180
33 Iran Standard/Daylight Time +210
34 Caucasus Standard/Daylight Time +240
35 Arabian Standard Time +240
36 Afghanistan Standard Time +270
37 West Asia Standard Time +300
38 Ekaterinburg Standard Time +300
39 India Standard Time +330
40 Central Asia Standard Time +360
41 SE Asia Standard Time +420
42 China Standard/Daylight Time +480
43 Taipei Standard Time +480
44 Tokyo Standard Time +540
45 Cen. Australia Standard/Daylight Time +570
46 AUS Central Standard Time +570
47 E. Australia Standard Time +600
48 AUS Eastern Standard/Daylight Time +600
49 West Pacific Standard Time +600
50 Tasmania Standard/Daylight Time +600
51 Central Pacific Standard Time +660
52 Fiji Standard Time +720
53 New Zealand Standard/Daylight Time +720
54 Venezuela Standard Time -270
55 Pacific SA Daylight Time -180
56 Pacific SA Standard Time -240
From the table you shared, it is clearly mentioned that the incoming number is 4 digits. From the Debug CCSIP messages attached to the OP, your ISP is sending calls with the called number 8800. However, your translation rule 11 says to translate the full number to 103, and you have applied this on the incoming dial-peer 100. The translation will never occur, and the call will never land on the phone 103 as you are expecting.
voice translation-rule 11
rule 1 /^6143888800$/ /103/
Either change the rule to match the incoming as rule 1 /^8800$/ /103/ or add the 4 digit rule as the second rule on the same translation.
voice translation-rule 11
rule 1 /^6143888800$/ /103/
rule 2 /^8800$/ /103/
For the outgoing call could you share the debug CCSIP messages as i didn't find the outgoing calls on the attached debug messages.
I would suggest these changes to your configuration.
service password-encryption
!
voice service voip
ip address trusted list
ipv4 10.187.128.3 255.255.255.255
no allow-connections h323 to h323
no allow-connections h323 to sip
no allow-connections sip to h323
mode border-element license capacity <number of sessions>
!
no supplementary-service h450.12
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
no registrar server expires max 600 min 60
no localhost dns:10.187.128.3
!
no voice class sip-profiles 1
!
voice translation-rule 11
rule 1 /8800$/ /103/
rule 2 /....$/ /103/
!
voice class uri PSTN sip
host ipv4:10.187.128.3
!
dial-peer voice 9000 voip
description ** INBOUND CALLS from telecoms **
no session target sip-server
no incoming called-number .
incoming uri via PSTN
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind control source-interface GigabitEthernet0/2
!
dial-peer voice 100 voip
description **** OUTBOUND CALLS to telecoms *****
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind control source-interface GigabitEthernet0/2
!
sip-ua
no retry register 10
no registrar dns:10.187.128.3 expires 3600
no host-registrar
Also I would recommend you to have a look at these documents as they should include all the information needed to understand how to setup a Cube and how call routing in IOS operates.
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5
Explain Cisco IOS and IOS XE Call Routing
To continue troubleshooting on this please turn on these debugs in parallel and post the output in a text file.
- debug ccsip messages
- debug voip ccapi inout
- debug voip translation
You did from what I can see part of configuration changes I suggested. Please follow the advice given completely and if you have any questions or doubts please feel free to come back with them so we can discuss them.
Apart from not following what I suggested you removed this command from sip-ua.
sip-server dns:10.187.128.3
Without that your outbound dial peer doesn’t know where to send the call as your using session target sip-server.
No, there are more than that you skipped. On those commands they have no correlation to FXS ports as they uses POTS dial peers, not VOIP and certainly not H.323. These commands makes your Cube support inter working between SIP and H.323, it hasn’t anything to do with analog ports.
These are the changes I suggested that I see that you didn't do.
service password-encryption
!
voice service voip
ip address trusted list
ipv4 10.187.128.3 255.255.255.255
no allow-connections h323 to h323
no allow-connections h323 to sip
no allow-connections sip to h323
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
!
voice translation-rule 11
rule 1 /8800$/ /103/
!
dial-peer voice 9000 voip
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind control source-interface GigabitEthernet0/2
!
dial-peer voice 100 voip
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind control source-interface GigabitEthernet0/2
You should also add this back.
sip-ua
sip-server dns:10.187.128.3
Apart from this I would as well recommend you do do this as your CME router would hardly act as a DNS server, so no point in "telling" that to the clients that gets an IP via DHCP
ip dhcp pool VOICE
dns-server 8.8.8.8
On your outbound calls, there is no invite sent to your service provider and the call disconnects with cause code 3. This means that there is no route to the destination, see this.
|
Destination address resolution failure |
Typical scenarios include:
|
3 |
CC_CAUSE_NO_ROUTE Indicates that the called party cannot be reached because the network that the call has been routed through does not serve the desired destination. |
Are your dial peer 100 in an up/active state?
On the outbound issue. Try changing the session target on dial peer 100 to session target ipv4:10.187.128.3.
I missed earlier that you define the SIP server under sip-ua as sip-server dns:10.187.128.3. That isn’t correct as you can't use an IP with dns:. So likely you dial peer 100 doesn’t know where to send the call.
If you where to use SIP server this is how I think it should be configured under sip-ua to use an IP.
sip-ua
sip-server ipv4:10.187.128.3
Sorry, I had the bind statements on the dial peers incorrect. This should work.
voice service voip
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
!
dial-peer voice 9000 voip
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind control source-interface GigabitEthernet0/2
!
dial-peer voice 100 voip
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind control source-interface GigabitEthernet0/2
I've set the global bind to the interface you say is used by the phones and the dial peer bind to the interface used for your SIP circuit.
If interface Gi0/1 is for IP phones why do your have NAT setup on that interface? Your phones should never need to access internet.
