Monday, April 13, 2015

Cisco CUBE: ATT SIP To Cisco Cube Router Configuration Example

One thing I have noticed is that working on a SIP config for an AT&T SIP trunk is not the same as most other providers.  In fact, its been really hard to even find a config out there to look at.  All I could find is the several hundred pages of an AT&T document.  So Im pasting my router config here in the hopes this helps someone looking for an AT&T SIP config for their CUBE.  See below.  I hope its helpful.
One thing to note is that in the dial-peers, you do not have to point to the AT&T media servers.  You only have to point your dial-peers to the SIP proxy servers.

======== Cisco CUBE config for AT&T SIP trunk ========
CUBE#sh run
Building configuration...


Current configuration : 9877 bytes
!
! Last configuration change at 23:16:45 CDT Thu Apr 2 2015 by admin
version 15.2
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
service password-encryption
service sequence-numbers
!
hostname CUBE
!
boot-start-marker
boot-end-marker
!
!
no logging queue-limit
logging buffered 1000000
no logging rate-limit
no logging console
no logging monitor
!
no aaa new-model
clock timezone CST -6 0
clock summer-time CDT recurring
!
ip cef
!
!
ip dhcp excluded-address 10.10.10.1
!
!
!
ip domain name company.com
no ipv6 cef
!
multilink bundle-name authenticated
!
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice service voip
 address-hiding
 mode border-element
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 redirect ip2ip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 sip
  bind control source-interface GigabitEthernet0/0
  error-passthru
  asserted-id pai
  early-offer forced
  midcall-signaling passthru
  privacy-policy passthru
  g729 annexb-all
!
voice class codec 1
 codec preference 1 g729r8 bytes 30
 codec preference 2 g711ulaw
!
voice class sip-profiles 1
 response ANY sip-header Allow-Header modify "UPDATE," ""
 request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
 response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
 request REINVITE sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""
 request INVITE sdp-header Audio-Attribute add "a=ptime:30"
!
!
voice iec syslog
!
!
voice translation-rule 9
 rule 1 /^9/ //
!
!
voice translation-profile outbound
 translate called 9
!
!
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
!
redundancy
!
ip tftp source-interface GigabitEthernet0/0
ip ssh version 2
!
!
interface Loopback0
 no ip address
!
interface Embedded-Service-Engine0/0
 no ip address
 shutdown
!
interface GigabitEthernet0/0
 description Local LAN
 ip address 192.168.1.2 255.255.255.0
 ip virtual-reassembly in
 duplex auto
 speed auto
!
interface GigabitEthernet0/1
 description ATT Network
 ip address 12.13.14.10 255.255.255.248
 duplex full
 speed 100
!
interface GigabitEthernet0/2
 description VLAN1
 no ip address
 duplex auto
 speed auto
!
interface GigabitEthernet0/2.1
 description VLAN1
 encapsulation dot1Q 1 native
 ip address 192.168.1.3 255.255.255.0
!
!
ip forward-protocol nd
!
no ip http server
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
ip route 0.0.0.0 0.0.0.0 192.168.1.1
ip route 12.194.138.0 255.255.255.0 12.13.14.9
!
!
control-plane
!
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm 192.168.1.13 identifier 2 priority 1 version 7.0
sccp ccm 192.168.1.12 identifier 1 priority 2 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 2 priority 1
 associate ccm 1 priority 2
 associate profile 1 register ATNAL-DC-xcode
 associate profile 2 register ATNAL-DC-cfb
!
dspfarm profile 1 transcode
 description DC Xcoder
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 20
 associate application SCCP
!
dspfarm profile 2 conference
 description DC conference bridge
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 20
 associate application SCCP
!
dial-peer voice 1 voip
 description Incoming 10-digit calls from AT&T - Facing CUBE for destination
 session protocol sipv2
 incoming called-number [2-9]..[2-9]......$
 voice-class codec 1
 voice-class sip asymmetric payload full
 voice-class sip asserted-id pai
 voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 fax rate 14400 bytes 48
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback none
 no vad
!
dial-peer voice 2 voip
 description Incoming Peer for Outbound calls to AT&T - Facing CUBE for desti
 session protocol sipv2
 incoming called-number 9T
 voice-class codec 1
 voice-class sip asymmetric payload full
 voice-class sip asserted-id pai
 voice-class sip privacy-policy passthru
 voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 fax rate 14400 bytes 48
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback none
 no vad
!
dial-peer voice 12 voip
 description Destination of 10-digit calls from AT&T - Facing CUCM Publisher
 destination-pattern [2-9]..[2-9]......$
 session protocol sipv2
 session target ipv4:192.168.1.12
 voice-class codec 1
 voice-class sip asymmetric payload full
 voice-class sip asserted-id pai
 voice-class sip privacy-policy passthru
 voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 fax rate 14400 bytes 48
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback none
 no vad
!
dial-peer voice 13 voip
 description Destination of 10-digit calls from AT&T - Facing CUCM Subscriber
 destination-pattern [2-9]..[2-9]......$
 session protocol sipv2
 session target ipv4:192.168.1.13
 voice-class codec 1
 voice-class sip asymmetric payload full
 voice-class sip asserted-id pai
 voice-class sip privacy-policy passthru
 voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 fax rate 14400 bytes 48
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback none
 no vad
!
dial-peer voice 20 voip
 description Outgoing calls to AT&T - Facing AT&T Network for Call Setup
 translation-profile outgoing outbound
 destination-pattern 9T
 session protocol sipv2
 session target ipv4:12.194.138.181         <----------- AT&T SIP Proxy
 voice-class codec 1
 voice-class sip asymmetric payload full
 voice-class sip asserted-id pai
 voice-class sip privacy-policy passthru
 voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 fax rate 14400 bytes 48
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback none
 no vad
!
dial-peer voice 30 voip
 description Outgoing calls to AT&T - Facing AT&T Network for Call Media Stre
 translation-profile outgoing outbound
 destination-pattern 9T
 session protocol sipv2
 session target ipv4:12.194.138.69    <----------- AT&T SIP Proxy
 voice-class codec 1
 voice-class sip asymmetric payload full
 voice-class sip asserted-id pai
 voice-class sip privacy-policy passthru
 voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 fax rate 14400 bytes 48
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback none
 no vad
!
!
sip-ua
!
!
!
gatekeeper
 shutdown
!
!
!
scheduler allocate 20000 1000
ntp server us.pool.ntp.org
!
end

CUBE#term len 24
CUBE#

4 comments:

  1. Wow...great post!! I have not worked with any SIP trunks as yet. After reading your post, I'm not sure I want too!!

    ReplyDelete
    Replies
    1. Thanks Brad. AT&T was not alot of help, but I'm hoping this will be helpful to folks.

      Delete
  2. Shane, you are a life saver with this. Thank you very much! You need a paypal account for tips!

    ReplyDelete

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