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sip trunk

dear all 

i have a cisco router 2811  i need  to config sip trunk and  call through  fastethernet  port 

 

is that Possible  or  must have e1/t1 card  

 

if possible  please help 

 

 

thank you 

2 ACCEPTED SOLUTIONS

Accepted Solutions

sa fas as i know:Yes this is

sa fas as i know:

Yes this is possible, You should be able to reach your SIP provider through that interface.

You also have to bind the sip control interface under voice service voip. and allow sip to sip connection.

 

regards

Chintan

VIP Super Bronze

Raed,  Here are some thoughts

Raed,

 

 

Here are some thoughts for 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:<CUCM_Address>
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:<SIP_Trunk_IP_Address>: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
Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
4 REPLIES

sa fas as i know:Yes this is

sa fas as i know:

Yes this is possible, You should be able to reach your SIP provider through that interface.

You also have to bind the sip control interface under voice service voip. and allow sip to sip connection.

 

regards

Chintan

New Member

thank you  do you can send to

thank you  

do you can send to me a configuration example 

regards

VIP Super Bronze

Raed,  Here are some thoughts

Raed,

 

 

Here are some thoughts for 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:<CUCM_Address>
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:<SIP_Trunk_IP_Address>: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
Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Cisco Employee

Take a look at this.http:/

Take a look at this.

http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html

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