Integrate SIP Trunk

[simaobarbosa20815]

I have been recently  requested to integrate sip trunk on existing  Voice service  with  line connected to FXO. The company  decided to replace  the PSTN service to use SIP. The SIP provider  IP address is different  network of the existing  network in the site. 

How do i integrate the sip service  to communicate with the existing network?

The existing  devices  are  voice gateway  2901 ---- connected to  CUCM.

Please HELP 

[Jaime Valencia]

That will depend on what your ISP needs and how they connect to you, so, start by asking them to provide the details for you to connect to them and if they have any config templates they can provide (if applicable)

If you don't have CUBE licenses, you will need to buy them, but that ISR is EOL so you would need to replace it with a newer one, not sure if that ISR was properly sized for your needs.

You'd also need to configure all the necessary dial peers and depending on what the telco wants, use normalization, then adjust your call routing in CUCM as necessary.

If you have never done any of the above, you want to reach out to a reputable consultant or a local Cisco Partner WITH voice specializations for assistance with this.

[Ritesh Desai]

HI @simaobarbosa20815 

Check out this URL for planning of SIP trunks. It also lists what all information is required from SIP TRUNK TELECOM provider for seamless configuration experience. If you have knowledge on configuration part then it would be easy to implement BUT if you aren't aware of protocols and how it works then you should take consultation from Cisco partners.

Although, Cisco partners would request the pre-requisite information from telecom provider.

URL: https://community.cisco.com/t5/collaboration-voice-and-video/sip-trunk-prerequisites/tac-p/4190455#M9887

[CSCO12245191]

Hi @Ritesh Desai , thank you.  I have never implemented the SIP service. i have got the pre-requisite as per you attached

The IP address of the SIP Provider is 10.255.x.x, and the CUCM is 10.131.18x.1X and the voice gateway is 10.139.18x.25x. How the SIP provider IP will communicate with the cucm and the voice gateway knowing that there are in different IP range? 

[Ritesh Desai]

Hey Hi CSCO12245191,



You need to convert the Voice Gateway to CUBE functionality to fully support SIP service. You need to configure by adding some commands on Voice Gateway. AFAIK, PRI and SIP Trunk cannot work on same Voice Router.



Telcom SIP Trunk can directly connect to CUBE or on NW SW (L2 or L3). NW SW port and CUBE must be in same VLAN also the VLAN for SIP TRUNK from service provider is configured.



You have asked very big question.



Once the call arrived on CUBE via WAN interface, then CUBE will match the dial-peer i.e. CUCM and route to CUCM via LAN interface.



I will recommend to have SIP TRUNKS connected to NW SW for fault tolerance or for building redundancy.

[CSCO12245191]

@Ritesh Desai  much appreciated for your help.

would you please send me step by step configure

[George Sotiropoulos]

Hello @CSCO12245191 ,

Below you will find the CUBE Configuration guide:

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

However, as @Jaime Valencia described above, if you haven't worked on this before, it is strongly suggested that you should contact a Cisco Partner Specialized in UC to assist you accordingly.

George

[Roger Kallberg]

That's very likely not something you'll get from anyone here as it would imply that we'd have inside information on your setup and we don't. You'll need to do the work to understand how this is done or seek help from a Cisco partner with the required know how to set this up. See it as a great way to learn some new technology. We have all been new to this at one time or another, what differentiate a bad engineer from a good one is the ability and will to learn new or expand the knowledge into deeper understanding of topics.

Best of luck on this.

[BalajiSivaraj49175]

Can you use the following link or configure configure sip services

1. enable

2. configure terminal

3. voice service voip

4. sip

5. associateregistered-numbernumber

6. exit

1.    enable

2. configure terminal

3.    sip-ua

4.   registrar registrar-server-address: ip-address auth-realm

5.    exit

Configuration of SIP Trunking for PSTN Access (SIP-to-SIP) Configuration Guide, Cisco IOS XE Release 3S (Cisco ASR 1000) - Configuring SIP Trunk Registration [Support] - Cisco

[Roger Kallberg]

What is the purpose of line 5 in the first part? Never seen or use that on any SIP trunk setups.

[Ritesh Desai]

Hi @roger,

Same question from me, even i m thinking i deployed 5 SIP trunks till now and haven’t seen this command or came through...

[BalajiSivaraj49175]

Device(conf-serv-sip)# associate registered-number 1234 - Associates the preloaded route and outbound proxy
details with the registered number. 

voi-sip-trunk-reg.pdf (cisco.com)

[BalajiSivaraj49175]

Enabling the Authorization Header
Use the following configuration to enable or disable authorization header support in REGISTER requests and
associate the realm with the register. The configured private-id of the user is used for populating authorization
header.
SUMMARY STEPS
1. enable
2. configure terminal
3. sip-ua
4. registrar registrar-server-address: ip-address auth-realm
5. exit

[BalajiSivaraj49175]

SIP can be completely verified using the following documentation from gateway level 

Cisco IOS Voice Command Reference - S commands - show sip service through show trunk hdlc [Cisco Unified Border Element] - Cisco

[BalajiSivaraj49175]

SIP message can verified using the following commands

show sip service
show sip-ua calls
show sip-ua connections
show sip-ua map
show sip-ua min-se
show sip-ua mwi
show sip-ua register status
show sip-ua retry
show sip-ua service
show sip-ua srtp
show sip-ua statistics
show sip-ua status
show sip-ua status refer-ood
show sip-ua timers