[cisco-voip] MOH and SRST With SIP trunk

Mon Jun 20 09:07:07 EDT 2011

Looks like you need a dial-peer for your extensions that points toward your SRST IP address with a higher preference than the peer going to  CUCM>

Dennis Heim
Network Voice Engineer
CDW  Advanced Technology Services
10610 9th Place
Bellevue, WA 98004

317.569.4255 Single Number Reach
317.569.4201 Fax
dennis.heim at cdw.com<mailto:dennis.heim at cdw.com>
cdw.com/content/solutions/unified-communications/<http://www.cdw.com/content/solutions/unified-communications/>

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Granger, Simon
Sent: Monday, June 20, 2011 7:33 AM
To: 'Matt Slaga (AM)'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MOH and SRST With SIP trunk

In Call manager, we have a SIP trunk configured to the Branch Router, then a voip dial peer as follows:

voice rtp send-recv
!
voice service voip
 address-hiding
 allow-connections h323 to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 sip
  header-passing sip-sip
  error-passthru
  early-offer forced
!
voice class codec 3001
 codec preference 1 g729r8
 codec preference 2 g711alaw
 codec preference 3 g711ulaw
!
voice class codec 3000
 codec preference 1 g711ulaw
 codec preference 2 g729r8
 codec preference 3 g711alaw
!
voice class h323 1
  h225 timeout tcp establish 5
  call preserve
!
voice class sip-profiles 1
 request INVITE sip-header To modify "(<.*:)9(.*@<mailto:.*@>)" "\1\2"
 request INVITE sip-header SIP-Req-URI modify "INVITE sip:9(.*@)" "INVITE sip:\1"
 request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
 request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
 response 200 sdp-header Audio-Attribute modify "sendonly" "sendrecv"
!
!
voice translation-rule 1
 rule 3 /^0/ /9\0/ type unknown unknown
 rule 4 /^[1-9]/ /900\0/ type unknown unknown
!
voice translation-rule 2
 rule 1 /^3/ /85213/
 rule 2 /\(.*\)/ /85213219/
!
voice translation-rule 3
 rule 1 /^026613/ /3/
 rule 2 /^85213/ /3/
 rule 3 /\(.*\)/ /3200/
!
voice translation-rule 4
 rule 1 /^9\(.*\)/ /\1/
!
voice translation-rule 10
 rule 1 /200/ /85213299/
!
!
voice translation-profile PSTN_INCOMIN
 translate calling 2
 translate called 3
!
voice translation-profile PSTN_OUTGOING
 translate calling 1
!
voice translation-profile SIP_INCOMING
 translate calling 1
 translate called 2
!
voice translation-profile SIP_OUTGOING
 translate calling 3
!
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
interface Loopback1
 description IPT Loopback
 ip address 10.205.4.1 255.255.255.255
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip pim sparse-mode
 h323-gateway voip bind srcaddr 10.205.4.1
!
interface Tunnel0
description DMVPN-Tunnel

sccp local Loopback1
sccp ccm 10.255.20.11 identifier 1 version 7.0
sccp ccm 10.248.20.12 identifier 2 version 7.0
sccp ccm 10.200.20.11 identifier 3 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate ccm 3 priority 3
 associate profile 1 register brusrtr1xcode
!
dspfarm profile 1 transcode
 codec ilbc
 codec g722-64
 codec g729r8
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 8
 associate application SCCP
!
dial-peer cor custom
!
!
dial-peer voice 1020 pots
 service stcapp
 port 0/0/0
!
dial-peer voice 1021 pots
 service stcapp
 port 0/0/1
!
dial-peer voice 1022 pots
 service stcapp
 port 0/0/2
!
dial-peer voice 1023 pots
 service stcapp
 port 0/0/3
!
dial-peer voice 70 pots
 destination-pattern 7T
 port 0/1/1
!
dial-peer voice 9 voip
 description Outgoing to COLT
 translation-profile outgoing SIP_OUTGOING
 destination-pattern 9.T
 session protocol sipv2
 session target ipv4:84.14.241.168
 session transport udp
 voice-class codec 3000
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no vad
 supplementary-service pass-through
!
dial-peer voice 4003 voip
 translation-profile outgoing SIP_INCOMING
 preference 1
 destination-pattern 01133600...
 session protocol sipv2
 session target ipv4:10.255.20.11
 session transport udp
 voice-class codec 3000
 voice-class sip bind control source-interface Loopback1
 voice-class sip bind media source-interface Loopback1
 dtmf-relay rtp-nte
 no vad
!
sip-ua
 retry invite 4
 retry response 3
 retry bye 2
 retry cancel 2
 retry register 5
 timers register 250
 sip-server ipv4:84.14.241.168
!
!
!
call-manager-fallback
 max-conferences 2 gain -6
 transfer-system full-consult
 timeouts interdigit 5
 ip source-address 10.205.4.1 port 2000
 max-ephones 8
 max-dn 8
 system message primary Backup Mode - Use full numbers
 dialplan-pattern 1 8521.... extension-length 8
 transfer-pattern .T
 transfer-pattern 8521....
 voicemail 85211900
 pickup 85213200
 alias 1 85213200 to 85213228
 alias 2 85213200 to 85213219
 alias 3 85213200 to 85213223
 alias 4 85213299 to 85213228
 call-forward pattern .T
 call-forward busy 85211900
 call-forward noan 85211900 timeout 12
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 10.205.4.1 10.205.20.1
 time-zone 44
 time-format 24
 date-format dd-mm-yy

From: Matt Slaga (AM) [mailto:matt.slaga at dimensiondata.com]
Sent: 20 June 2011 13:02
To: Granger, Simon; cisco-voip at puck.nether.net
Subject: RE: MOH and SRST With SIP trunk

Is the SIP trunk terminating on the branch router or at the central office?

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Granger, Simon
Sent: Monday, June 20, 2011 7:06 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MOH and SRST With SIP trunk

Hello all again,

Thanks for all the comments regarding my last post.

We have a Sip trunk that can dial in and out, but as soon as the system goes into SRST mode, all the phones re-register, but when you phone the Sip trunk number it does not ring on the phone.

If you do a debug ccsip messages, you can see it trying to make contact with the Call manager servers. Any Thoughts?

Secondly, we use multicast MOH, and if you call IP Phone to IP Phone that works fine, but it does not work at all across the SIP Trunk.

Thanks

Simon Granger
Senior Server Specialist

FM Global
1 Windsor Dials, Windsor, Berkshire, SL4 1RS, UK
T: +44 (0)1753 750154
F: +44 (0)1753 868700
www.fmglobal.com

________________________________
Registered No. 755780 England
Registered Office: FM Insurance Company Limited
1 Windsor Dials, Windsor,
Berkshire, UK, SL4 1RS
Regulated by the Financial Services Authority.
VAT No. G.B.: 792 4276 02

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