Cisco CME/IOS GW Config for BV

npope · #1 (**permalink**) August 12th, 2005, 03:54 AM

I finally got my IOS CallManager Express working with Broadvoice.

I've got it registering with sip.broadvoice.com and I've got the message-waiting lights working. There are few tweaks needed to get it working

I don't see many posts asking about IOS-based products? Would anyone be interested in a tutorial on how to get CCME and/or an IOS-based GW talking to BV?

If there is interest, I'll post a quick setup guide.

-Nick

npope · #2 (**permalink**) August 12th, 2005, 12:43 PM

I've already recieved an email asking for the config, so I guess that officially makes it interesting:

Here are the relavent parts of the config I used to get CCME working with BroadVoice:

Obviously the items in the config that begin with YOUR should be replaced with the IP addresses/names of the appropriate items for your setup.

I used a sample BV number: 1234567890 and password YOURSIPPASSWORD in the config below.

The sticky points were:
- Using sip.broadvoice.com for my dial peer targets, not proxy.broadvoice.com
- Using "dtmf-relay rtp-nte" under the dial peer to get DTMF working
- Using "no remote-party-id" under sip-ua*

Getting MWI (Message Waiting Indicator) to work involved:
- Adding the "mwi-server" with the "unsolicited" keyword to my sip-ua
- Adding "mwi sip" to the ephone-dn's that you want the MWI light to work

Things to note:
- I used the "voice class codec" in my dial-peers to allow the use of multiple codecs
- If you just want G.711U, then just add "codec g711u" to your dial-peer and skip all the "voice class codec" stuff

*Later versions of IOS add the "Remote Party ID" SIP header. They put phonenum@gatewayipaddress in the field. BV (and any SIP compliant provider) replaces the From information with info from this field. Well, BV doesn't like From fields to end in anything but sip.broadvoice.com (well, maybe a couple of others, but certainly not the IP address of you router!). So by inserting this command, it suppresses the creation of this field and the BV proxy will use the From field. This item is what gave me the biggest headache. IOS debug commands are your friend

By the way, the "sip-ua" and dial-peer parts of this config should work for any generic IOS device (later than 12.3(8)T or so).

Lastly, there is some extra fluff in the dial peers. I turned off VAD and set IPP to 5 for my RTP packets. These are not necessary but I've found them to be useful additions.

I hope this helps someone!
-Nick

<PRE>
ip domain name YOUR.DOMAIN.NAME
ip name-server YOUR.DNS.SERVER
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
ip route 0.0.0.0 0.0.0.0 YOUR.DEFAULT.ROUTE
!
!
!
dial-peer voice 1 voip
destination-pattern 1..........
voice-class codec 1
session protocol sipv2
session target dns:sip.broadvoice.com
dtmf-relay rtp-nte
ip qos dscp cs5 media
no vad
!
!
dial-peer voice 86 voip
destination-pattern *86
voice-class codec 1
session protocol sipv2
session target dns:sip.broadvoice.com
dtmf-relay rtp-nte
ip qos dscp cs5 media
no vad
!
sip-ua
authentication username 1234567890 password YOURSIPPASSWORD
no remote-party-id
mwi-server dns:sip.broadvoice.com expires 3600 port 5060 transport udp unsolicited
registrar dns:sip.broadvoice.com expires 3600
!
!
telephony-service
voicemail *86
!
!
!
ephone-dn 1 dual-line
number 1234567890
mwi sip
!
!
!
!
ephone 1
mac-address XXXX.YYYY.ZZZZ
button 1:1
!
!
ephone 2
mac-address XXXX.YYYY.ZZZZ
button 1:1
!
</PRE>

bkmeyer · #3 (**permalink**) October 23rd, 2005, 03:57 PM

Is it possible to restrict which dial-peers and ephnone-dn's attempt registration? During CCSIP debug i see all the ephone-dns and dialpeers trying to register with the SIP Server?

npope · #4 (**permalink**) October 24th, 2005, 11:32 AM

Under ephone-dn you should be able to use the "no-reg" modifier to the "number" command to prevent CME from registering that ephone-dn with the SIP registrar.

I'm not sure if it works for dial peers or not...

-Nick

bosseye · #5 (**permalink**) November 12th, 2005, 04:33 PM

Nick,
Thank you for posting - now I know there's someone else out there using CCME and a SIP gateway service! My problem is how to you get NAT and the ACLs set up appropriately - I wonder could you be kind enough to share this part of your config.

BTW - I have 3 phones on SCCP, with a dial-peer set to sipgate.co.uk (same thing as Broadvoice), and I noticed thta CME does indeed try to register all the phones.
TIA/David

tdar · #6 (**permalink**) January 21st, 2006, 08:41 PM

To add to the thread, here's the config that ViaTalk gives for their customers:

voice translation-rule 1
rule 1 /^2.../ /13035551212/
!
voice translation-rule 2
rule 1 /^91(..........$)/ /11/
rule 2 /^9(..........$)/ /11/
rule 3 /^9011($)/ /0111/
rule 4 /^9(.11)/ /1/
!
voice translation-profile SIP
translate calling 1
translate called 2
!
!
!
dial-peer voice 10 voip
translation-profile outgoing SIP
destination-pattern 9[49]11
session protocol sipv2
session target dns:555.303.1.switch.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 11 voip
translation-profile outgoing SIP
destination-pattern 9303.......
session protocol sipv2
session target dns:555.303.1.switch.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 12 voip
translation-profile outgoing SIP
destination-pattern 9720.......
session protocol sipv2
session target dns:555.303.1.switch.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 13 voip
translation-profile outgoing SIP
destination-pattern 91[2-9].........
session protocol sipv2
session target dns:555.303.1.switch.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 14 voip
translation-profile outgoing SIP
destination-pattern 9011T
session protocol sipv2
session target dns:555.303.1.switch.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw

dial-peer voice 15 voip
translation-profile outgoing SIP
destination-pattern *123
session protocol sipv2
session target dns:555.303.1.switch.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw

sip-ua
authentication username 13035551212 password YourPassword
no remote-party-id
registrar dns:555.303.1.switch.vtnoc.net expires 3600
sip-server dns:555.303.1.switch.vtnoc.net

telephony-service
load 7960-7940 P00303020214
max-ephones 10
max-dn 20
ip source-address 10.1.1.1 port 2000
time-zone 6
create cnf-files version-stamp 7960 Jul 27 2005 23:39:25
voicemail *123
mwi sip-server 67.15.74.73 unsolicited
max-conferences 4
call-forward pattern .T
moh flash:music-on-hold.au
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
after-hours block pattern 1 9411 7-24
after-hours block pattern 2 91900 7-24
after-hours block pattern 3 91010 7-24
after-hours block pattern 4 9303976 7-24
after-hours block pattern 5 9720976 7-24
after-hours block pattern 6 91809 7-24
after-hours block pattern 7 91758 7-24
after-hours block pattern 8 91664 7-24
after-hours block pattern 9 91284 7-24
after-hours block pattern 10 91876 7-24
!
ephone-dn 1 dual-line
number 2001 no-reg primary
name Phone1
!
ephone-dn 2 dual-line
number 2002 no-reg primary
name Phone2
!
ephone-dn 10 dual-line
number 13035551212
label Incoming Calls
!
ephone 1
mac-address 0012.D9BB.26DE
button 1:1 2:10
!
ephone 2
mac-address 0030.94C3.E550
button 1:2 2:10
!

heyligerb · #7 (**permalink**) March 15th, 2006, 07:05 PM

Well I got it working, thanks for that. However, I do have one question. Do you think it is possible to get the router to register to multiple BV accounts? I want to use this as a PBX for my office, but can't figure out a way to get multiple SIP-UA sessions. Is that possible in the IOS?

Tks,

Brian

heyligerb · #8 (**permalink**) March 15th, 2006, 08:07 PM

Nevermind, I got it working. Here ya go:

version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CCME
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
!
resource policy
!
!
!
ip cef
no ip dhcp use vrf connected
!
ip dhcp pool ITS
network 1.1.1.0 255.255.255.0
option 150 ip 1.1.1.1
default-router 1.1.1.1
dns-server 65.32.1.65 65.32.1.70
!
ip name-server 65.32.1.65
ip name-server 65.32.1.70
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
interface FastEthernet0/0
ip address dhcp
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 1.1.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
!
ip route 0.0.0.0 0.0.0.0 10.30.0.1
!
!
tftp-server flash:P003-07-4-00.bin
tftp-server flash:P003-07-4-00.sbn
!
control-plane
!
!
!
voice-port 0/3/0
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
dial-peer voice 1 voip
destination-pattern 1..........
voice-class codec 1
session protocol sipv2
session target dns:sip.broadvoice.com
dtmf-relay rtp-nte
ip qos dscp cs5 media
no vad
!
dial-peer voice 86 voip
destination-pattern *86
voice-class codec 1
session protocol sipv2
session target dns:sip.broadvoice.com
dtmf-relay rtp-nte
ip qos dscp cs5 media
no vad
!
dial-peer voice 87 pots
destination-pattern 1234561239
port 0/3/0
authentication username XXXXXXXXXXXX password XXXXXXXXXXXXXXXX
!
dial-peer voice 88 pots
destination-pattern 2234561239
port 0/3/1
authentication username XXXXXXXXXXXX password XXXXXXXXXXXXXXXX
!
sip-ua
no remote-party-id
mwi-server dns:sip.broadvoice.com expires 3600 port 5060 transport udp unsolicited
registrar dns:sip.broadvoice.com expires 3600
!
!
!
!
line con 0
line aux 0
line vty 0 4
login
!
scheduler allocate 20000 1000
ntp clock-period 17179893
ntp update-calendar
ntp server 198.123.30.132
!
webvpn context Default_context
ssl authenticate verify all
!
no inservice
!
!

engelbrechtl · #9 (**permalink**) April 10th, 2006, 04:17 PM

Thanks alot for your post above this has helped me alot.
But i have one problem.
I am using sipgate.co.uk and want to send calls to two diffrent accounts.
Basically have two dial peers one which authenticates using one sipgate.co.uk account and the other one with a diffrent sipgate.co.uk account.

The sip-ua command only allows you to use one authentication username and password, is there another way arround this?

Thanks
Louis

heyligerb · #10 (**permalink**) April 13th, 2006, 01:51 PM

Yes you just authenticate on the dial-peer, see here:
dial-peer voice 87 pots
destination-pattern 1234561239
port 0/3/0
authentication username XXXXXXXXXXXX password XXXXXXXXXXXXXXXX
!
dial-peer voice 88 pots
destination-pattern 2234561239
port 0/3/1
authentication username XXXXXXXXXXXX password XXXXXXXXXXXXXXXX