SIP Trunk from Etisalat

[hasnain.raza]

Hello All, We're setting up a BE6K and we have a SIP trunk from Etisalat which has an RJ45 connection coming to our router. Can someone give us the configuration that needs to be placed on the router to allow calls to flow to and from Etisalat's SIP trunk? All we've got from Etisalat is some bunch of credentials and IP addresses. We appreciate your support in this regard.

[George Sotiropoulos]

You need to be familiar with Cisco CUBE configuration, you may start with the following:

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/117300-configure-cube-00.html

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-overview.html

Last but not least the post below refers to Etisalat config on CUBE/CME

https://community.cisco.com/t5/ip-telephony-and-phones/sip-trunk-pri-from-etisalat-cme-configuration/td-p/2570491

G

[George Sotiropoulos]

+5 to Java, this session will help

https://www.ciscolive.com/c/dam/r/ciscolive/latam/docs/2018/pdf/BRKCOL-2125.pdf

G

[TONY SMITH]

I've just received info for an Etisalat trunk we're going to be deploying in the next few weeks (hopefully).  It looks straightforward, they're providing a separate physical connection with private addressing so the CUBE will have a separate inside and outside interfaces.  As is often the case they've not specified codec or number formats, so we'll have to work those out from what they present once it goes live.  It's unclear whether they expect registration/authentication so I'll probably try without and take it from there.

If you get started with a basic config then you can post up how you get on and we can help if there turn out to be any funnies.

[hasnain.raza · Accepted solution]

We've successfully completed SIP trunk Configuration. Below is the configuration for your reference.

!
voice rtp send-recv
!
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
h225 connect-passthru
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g722-64
codec preference 3 g711alaw
!
voice class h323 1
h225 timeout tcp establish 3
!
!
!
!
voice iec syslog
!
!
voice translation-rule 1
rule 1 /8847165/ /89/
!
voice translation-rule 2
!
voice translation-rule 3
rule 1 /^9\(.........$\)/ /\1/
rule 2 /^9\(.......$\)/ /04\1/
!
voice translation-rule 4
rule 1 /^9\(.......$\)/ /04\1/
!
voice translation-rule 10
rule 1 /^.*/ /048847165/
!
!
voice translation-profile LOCAL
translate called 3
!
voice translation-profile MISSED
translate calling 2
translate called 1
!
voice translation-profile OUTGOING
translate calling 10
translate called 3
!
!
!
license udi pid ISR4321/K9 sn FDO23120GE5
diagnostic bootup level minimal
spanning-tree extend system-id
!
!
!
username admin secret 5 $1$Jyjv$K9x30e6WRk82toTQ1PlaM1
!
redundancy
mode none
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0/0
ip address 192.168.0.225 255.255.255.0
negotiation auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.0.225
!
interface GigabitEthernet0/0/1
ip address 192.168.1.10 255.255.255.0
negotiation auto
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
shutdown
negotiation auto
!
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
!
!
!
!
snmp-server community public RO
!
!
control-plane
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 1 voip
destination-pattern [1238].
session protocol sipv2
session target ipv4:192.168.0.220
voice-class codec 1
dtmf-relay rtp-nte h245-alphanumeric
!
dial-peer voice 100 voip
translation-profile outgoing OUTGOING
destination-pattern .T
session protocol sipv2
session target sip-server
session transport udp
voice-class codec 1
no voice-class sip localhost
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 200 voip
translation-profile incoming MISSED
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 1
dtmf-relay sip-notify rtp-nte
no vad
!
dial-peer voice 2 voip
destination-pattern ^89
session protocol sipv2
session target ipv4:192.168.0.220
voice-class codec 1
dtmf-relay rtp-nte sip-kpml
no vad
!
!
sip-ua
credentials number 048847165 username 48847165.etisalat password 7 02071341390D1A7308 realm etisalat.com
authentication username 48847165.etisalat password 7 02071341390D1A7308 realm etisalat.com
no remote-party-id
max-forwards 6
retry invite 2
retry bye 1
retry register 10
timers expires 360000
timers connect 100
registrar dns:48847165.etisalat expires 3600
sip-server dns:48847165.etisalat:5060
connection-reuse
!
!
line con 0
transport input none
stopbits 1
line aux 0
stopbits 1
line vty 0 4
password 7 052B142F331F1C2A5D544541
login
!
ntp master 7
ntp server 192.168.0.225 prefer
wsma agent exec
!
wsma agent config
!
wsma agent filesys
!
wsma agent notify
!
!
end

[mohammedkhan0507]

Dear we have problem configuring the etisalat sip trunk on isr 4321 please can you help in by giving me the configuration 

i tried the configuration you posted but i am unsuccessful my trunk details are 

customer ip 192.168.1.19

sbc 192.168.1.6

username 045968000

password :*******

domain 45968000.etisalat

did range 045968000 to 045968699

i am having issue in dial-peers and translation rule 

please help me 

[hasnain.raza]

Hi,
I will share the full configuration tomorrow and if that doesn't solve help you open a TAC case to solve the issue.

Regards,
Hasnain
0501842400

[hasnain.raza]

Hello,

Please find the tried and tested configuration below:

!

!

voice rtp send-recv

!

voice service voip

ip address trusted list

ipv4 0.0.0.0 0.0.0.0

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

h225 connect-passthru

sip

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g722-64

codec preference 3 g711alaw

!

voice class h323 1

h225 timeout tcp establish 3

!

!

!

!

voice iec syslog

!

!

voice translation-rule 1

rule 1 /2262244/ /89/

!

voice translation-rule 2

!

voice translation-rule 3

rule 1 /^9\(.........$\)/ /\1/

rule 2 /^9\(.......$\)/ /04\1/

!

voice translation-rule 4

rule 1 /^9\(.......$\)/ /04\1/

!

voice translation-rule 10

rule 1 /^.*/ /042262244/

!

!

voice translation-profile LOCAL

translate called 3

!

voice translation-profile MISSED

translate calling 2

translate called 1

!

voice translation-profile OUTGOING

translate calling 10

translate called 3

!

!

!

license udi pid ISR4321/K9 sn FDO23120GE5

diagnostic bootup level minimal

spanning-tree extend system-id

!

!

!

username admin secret 5 $1$Jyjv$K9x30e6WRk82toTQ1PlaM1

!

redundancy

mode none

!

!

!

!

!

!

!

!

interface GigabitEthernet0/0/0

ip address 192.168.0.225 255.255.255.0

negotiation auto

h323-gateway voip interface

h323-gateway voip bind srcaddr 192.168.0.225

!

interface GigabitEthernet0/0/1

ip address 192.168.1.10 255.255.255.0

negotiation auto

!

interface GigabitEthernet0

vrf forwarding Mgmt-intf

no ip address

shutdown

negotiation auto

!

ip forward-protocol nd

ip http server

ip http authentication local

ip http secure-server

!

!

!

!

snmp-server community public RO

!

!

control-plane

!

!

mgcp behavior rsip-range tgcp-only

mgcp behavior comedia-role none

mgcp behavior comedia-check-media-src disable

mgcp behavior comedia-sdp-force disable

!

mgcp profile default

!

!

!

!

dial-peer voice 1 voip

destination-pattern [1238].

session protocol sipv2

session target ipv4:192.168.0.220

voice-class codec 1

dtmf-relay rtp-nte h245-alphanumeric

!

dial-peer voice 100 voip

translation-profile outgoing OUTGOING

destination-pattern .T

session protocol sipv2

session target sip-server

session transport udp

voice-class codec 1

no voice-class sip localhost

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

!

dial-peer voice 200 voip

translation-profile incoming MISSED

session protocol sipv2

session target sip-server

incoming called-number .T

voice-class codec 1

dtmf-relay sip-notify rtp-nte

no vad

!

dial-peer voice 2 voip

destination-pattern ^89

session protocol sipv2

session target ipv4:192.168.0.220

voice-class codec 1

dtmf-relay rtp-nte sip-kpml

no vad

!

!

sip-ua

credentials number 042262244 username 42262244.etisalat password 7
02071341390D1A7308 realm etisalat.com

authentication username 42262244.etisalat password 7 02071341390D1A7308
realm etisalat.com

no remote-party-id

max-forwards 6

retry invite 2

retry bye 1

retry register 10

timers expires 360000

timers connect 100

registrar dns:42262244.etisalat expires 3600

sip-server dns:42262244.etisalat:5060

connection-reuse

!

!

Let me know if you need any further assistance.

Regards,

Raza

+971-50-1842400

[George Sotiropoulos]

Last but not least, there is a post on the community where you can find a configuration example (or part of it) of a CUBE for this provider

https://community.cisco.com/t5/ip-telephony-and-phones/sip-trunk-registration-problem/td-p/3016336

G

[hasnain.raza]

Dear All,

To enable CUBE on our router do we just need the below license:

CUBE-T-STD list price of $95?

[TONY SMITH]

That part is the licence for one CUBE trunk session, so order the quantity matching the number of channels you're getting from Etisalat.   Your gateway also needs the UC Feature Set licence, or one of the Cisco One packages including UC features.   Are you ordering a new gateway or adding capabilities to an existing one?

[hasnain.raza]

Hello,

We've successfully completed the SIP trunk configuration and did not need any CUBE licenses on ISR 4321.

[Jaime Valencia]

There are also a lot of Cisco Live sessions on CUBE which explain in detail how CUBE works and how it's configured.

[TarekElChabty]

Hi, can you please identify the solution components to connect with Etisalat SIP trunking. 

[TONY SMITH]

What product are you connecting, for example if it's Cisco CUCM you will need a CUBE which comprises a Cisco router with the UC Feature Set and CUBE licensing.  This can be combined with other router functions such as WAN or Internet access, but Etisalat normally provide a dedicated connection for SIP so you will need a free Ethernet interface for that.

CUBE talks to Etisalat, CUCM and the phones talk to the CUBE.

More detailed questions would be better asked in a new thread, rather than adding to an old one that's already marked as solved.