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New Member

SIP Trunk Configuration

Hi

We are migrating from Analogue to IP Telephony. I have recieved the following guidlines to configure the SIP Trunk:

*For signaling: use IP :  x.x.211.70   ( SIP ) on PORT 5060

*Regarding Numbering Format, use the following:

•             For outgoing Calls :

                The originating Number (A#), should be 96611510XXXX format.

                 The Destination Number should be 0NXXXXXX (N area code) or 00XXXXXXXXX (for international)

•             For incoming Calls:

                The Destination Number (B#), should 011510XXXX Format.

                The originating Number (A#), will be 0NXXXXXXX or 00XXXXXXXXXXX Format

*Use Audio Codec's G711-aLaw ; G711-uLaw & G729

*Use T.38 For FAX

*set DTMF to RFC2833

*Make sure to reply with 200Ok for our OPTIONS messages ( ping messages for the SIP)

* configure the following SIP Timers:    “Min-SE=1800 “’  & “Expires=300”

For connectivity consider the following:

SIP CE: 10.65.13.110 (it might be needed to translate this IP to the PBX local IP).

SIP GW: 10.65.13.109

Subnet mask: /30

SIP VLAN: 1191

Notes:

Kindly make sure to have GO SIP GW (x.x.211.70) routed to SIP GW (10.65.13.109) as next-hop.

Kindly make sure to have SIP CE IP addresses are in VLAN 1191.

Can please anyone explain what have to done?

Regards

3 REPLIES
Hall of Fame Super Gold

SIP Trunk Configuration

Ummmmm ... This looks like a configuration sheet for Asterisk. 

If this is the case, then this is the wrong place to do so.  Your first stop should be VoIP-Info website.  Go here.

New Member

SIP Trunk Configuration

Hi Leo

This is the information we have recieved from our service provider. We have to get this configured onto Cisco 2921 or CUCM 9.1.

Regards

VIP Super Bronze

SIP Trunk Configuration

Ahmed,

Wao..Where do I start...This information is required for configuration on your CUBE..which will be your 2921 router...

Ahmed, here are some pointers I wrote a while ago..

In addition to these points, you will need to configure your cube to be able to route traffic to your ITSP using all the information given to you

1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.

2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM

voice service voip

early-offer forced

3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.

4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.

voice service voip

allow-connections sip to sip

sip

early-offer forced

header-passing

error-passthru

5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP

6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.

voice service voip

ip address trusted list

  ipv4 203.0.113.100 255.255.255.255

  ipv4 192.0.2.0 255.255.255.0

This is imprtant because sometimes your ITSP will send you a single ip address for signalling and will then send media on a different IP adress. So get all the IP address your ITSP is using and add them to the trust list as shown above

7. Configure your inbound and outbound dial-peer approriately

Inbound Dial-Peer for calls from CUCM to CUBE (CUCM sending 9 +all digits dialled to CUBE)

dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte

Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)


dial-peer voice 200 voip
description *** Outbound LAN side dial-peer ***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:
codec g711ulaw
dtmf-relay rtp-nte

Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing


Inbound Dial-Peer for calls from SP to CUBE


dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***------------------(catch-all for all inbound PSTN calls)
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte


Outbound Dial-Peer for calls from CUBE to SP


dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4::XXXX (where XXXX is the port number your provider is using if different from 5060)
codec g711ulaw
dtmf-relay rtp-nte

8. SIP Normalization:

You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to  match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.

9. Media Resources

Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte

e.g

dial-peer voice 1 voip

session protocol sipv2

dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)

If in your environment you will need to do xcoding or CFB then ensure you have PVDMS

.10.FAX

If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing  what they support. I have seen legal cases because of fax failures over sip trunks

Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls

Finally

11. Have a detailed and carefully planned TEST Plan. Test the FF:

  • Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)
  • Outbound calls to information and emergency services
  • Caller ID and Calling Name Presentation
  • Supplementary services like Call Hold, Resume, Call Forward & Transfer
  • DTMF Tests
  • Fax calls – T.38, modem pass-through--whichever one you decide to use

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