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CUBE Router - SIP Trunk Answer Points on SRST and SRST Review

Hi All. 

 

Requesting for your advice how can we modify the dial peer that when the Network between the CUCM and the ISR 4331 breaks, all calls to the number 77755512.. gets answered at the Local Operator Phone 75000. And also if I am missing any thing on the SRST configuration.

 

Currently when calls comes on 7775551200, the call is routed to CUCM at the Main Site, and then in the CUCM there is a Translation Pattern for the 77755512XX block to be translated to 750XX. The Phone 750XX is located at the Remote Site where the ISR is based. The Phones are in the subnet 172.16.50.0/24

 

Thanks a lot in advance.

 

Adrian.

 

We have the current setup.

Telecom Provider------SIP Trunks---G0/0/1(ISR4331 CUBE)G0/0/0---------Network------------CUCM

 

On the 4331 - G0/0/0 - 172.16.10.1, G0/0/1 - 10.0.0.1

CUCM - 172.16.100.100

We do NAT between the G0/0/0 and the G0/0/1 to hide our IP from the Service Provider.

Phones are on the Network ---- 172.16.50.0/24 at the local site.

 

SIP Trunks are working fine and now we want to implement SRST, such that if the Network between the ISR 4331 and the CUCM Breaks, all calls are answered at the ISR 4331 Remote Site Locally.

 

The current SIP Dial Peers for the SIP Trunks are :

 

dial-peer voice 500 voip
translation-profile outgoing TelecomSP
destination-pattern 9T
session protocol sipv2
session target ipv4:10.0.0.2 [ Private SIP Trunk Provider]
session transport udp
voice-class codec 100
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte

 

dial-peer voice 600 voip
destination-pattern 77755512..$
session protocol sipv2
session target ipv4:172.16.100.100 [ CUCM IP Address]
session transport udp
incoming called-number 9T
voice-class codec 100
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte

 

Requesting for your advice how we can achieve this.

 

In my understanding, once the phone registers to the SRST, they should be able to make outgoing calls since Dial-Peer 500 would still be fine.

 

Current configuration is as follows:

voice service voip
ip address trusted list
ipv4 10.0.0.2
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
trace
sip
header-passing
error-passthru
registrar server
asymmetric payload full
midcall-signaling block
early-offer forced
no silent-discard untrusted
no call service stop

 

voice register global
mode esrst
no allow-hash-in-dn
system message "Limited Services"
max-dn 100
max-pool 100
!
voice register pool 1
id network 172.16.50.0 mask 255.255.255.0
dtmf-relay rtp-nte cisco-rtp sip-notify
no vad

 

 

5 Replies 5

I think you would need to configure the alias command in SRST. For details see this document and the screenshots.

Snag_88c732.png

image.png



Response Signature


Thanks a lot Roger

 

I tried to configure but ended with this error.

 

Remote1(config-register-pool)#alias 1 7775551200 to 75000
Failed: CME mode!

Have also tried to change the Voice Register Global to CME Mode and still ended with the same error.

 

We will be most grateful for your advise.

 

Regards

 

Adrian.

Try with this.

voice register global
 default mode

That is what we have on our voice gateways. I just tested to create an alias under the voice register pool and it worked.



Response Signature


Thanks a lot Roger

Have modified as and the alias got accepted. Thanks a lot for it. But when i tested by breaking the link between the CUCM and the Local Site, noticed the following problems.

 

1. The Cisco 7811 and 3905 Phones displayed the " Limited Services" message.

2. When we call from one phone to another, it rings, but when we answer, there is isnt any voice stream, its just a disconnect tone.

3. We could not receive any incoming calls from PSTN.

4. We could not dial out to the PSTN.

 

___________

voice service voip
ip address trusted list
ipv4 10.0.0.2
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
trace
sip
header-passing
error-passthru
registrar server
asymmetric payload full
midcall-signaling block
early-offer forced
no silent-discard untrusted
no call service stop

 

voice register global
mode default
no allow-hash-in-dn
system message "Limited Services"
max-dn 100
max-pool 100
!
voice register pool 1
id network 172.16.50.0 mask 255.255.255.0

alias 1 7775551200 to 75000 preference 0
dtmf-relay rtp-nte cisco-rtp sip-notify
no vad

 

dial-peer voice 500 voip
translation-profile outgoing TelecomSP
destination-pattern 9T
session protocol sipv2
session target ipv4:10.0.0.2 [ Private SIP Trunk Provider]
session transport udp
voice-class codec 100
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte

 

dial-peer voice 600 voip
destination-pattern 77755512..$
session protocol sipv2
session target ipv4:172.16.100.100 [ CUCM IP Address]
session transport udp
incoming called-number 9T
voice-class codec 100
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte

 

 

show voice register all


CONFIG [Version=14.1]
========================
Version 14.1
Mode is srst
Max-pool is 50
Max-dn is 50
VRF NA
Outbound-proxy is enabled and will use global configured value
Security Policy: DEVICE-DEFAULT
Allow-hash-in-dn is disabled
Forced Authorization Code Refer is enabled
System message is "Limited Services"
timeout interdigit 10
timeout transfer recall 0
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
MWI unsolicited notify is disabled
Active registrations : 0

Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 9
Registration success : 4
Registration failed : 5
unRegister requests : 4
unRegister success : 4
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 1
Last register request time : 19:11:50.517 XXXX Fri Feb 4 2022
Last unregister request time : 19:21:43.291 XXXX Fri Feb 4 2022
Register success time : 19:11:17.941 XXXX Fri Feb 4 2022
Unregister success time : 19:21:43.291 XXXX Fri Feb 4 2022

 

VOICE REGISTER DN
=================

VOICE REGISTER POOL
===================
Pool Tag 1
Config:
Network address is 172.16.50.0, Mask is 255.255.255.0
Proxy Ip address is 0.0.0.0
Alias Tag: 1
Number pattern is 75000
Alias is 7775551200
DTMF Relay is enabled, rtp-nte, cisco-rtp, sip-notify
kpml signal is enabled
Lpcor Type is none

Reason for unregistered state:
No registration request since last reboot/unregister

paging-dn: config 0 [multicast] effective 0 [multicast]

VRF:
NA

Dialpeers created:

Dial-peers for Pool 1:

Statistics:
Active registrations : 0

Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 4
Registration success : 4
Registration failed : 0
unRegister requests : 4
unRegister success : 4
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time : 19:11:17.940 XXXX Fri Feb 4 2022
Last unregister request time : 19:21:43.291 XXXX Fri Feb 4 2022
Register success time : 19:11:17.941 XXXX Fri Feb 4 2022
Unregister success time : 19:21:43.292 XXXX Fri Feb 4 2022

 

 

Please is it possible for you to review the configs for SRST for me please.

 

Regards

 

Adrian.

 

 

 

b.winter
VIP
VIP

Hi,

2. It sounds like a codec negotiation problem. Also set a codec or codec list in the voice-register pool.

3. According to your dial-peers, you have no incoming dial-peer, to match incoming calls from the provider? Not even for "normal" mode. In your dial-peer 600, do something like this:

voice class uri 600 sip

 host ipv4:10.0.0.2

!

dial-peer voice 600

 incoming uri via 600

!

Also, I would recommend taking logs.

4. I would recommend taking logs.