CME to CME

Unanswered Question
Feb 11th, 2010
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Hello,

For the past few weeks now I have been tinkering with the idea of connecting two Cisco 1760's running CME.  I have read several posts and cisco documentation relating to the topic and have failed with each config I have tried.  I am hoping that someone could please lend some insight or point out the flaw in my configuration. 


CME1 (1760-v)


voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip


dial-peer voice 110 voip
destination-pattern 4002
session target ipv4:x.x.x.x (CME 2 IP)
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad



CME2 (1760)


voice service voip
  allow-connections h323 to h323
  allow-connections h323 to sip
  allow-connections sip to h323
  allow-connections sip to sip


dial-peer voice 110 voip
destination-pattern 2...
session target ipv4:x.x.x.x (CME 1 IP)
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad



Any help is appreciated.

Thanks

-Jeff

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markbatts Fri, 02/12/2010 - 03:01
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Do you have a copy of the whole configuration?


cheers

mark

paolo bevilacqua Fri, 02/12/2010 - 03:46
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voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip


Not needed.


destination-pattern must cover remtoe extensions.

Aslo you need incoming called-number . on voip DP.

stowellp Fri, 02/12/2010 - 04:25
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Hi Jeff,


A lot of the config is missing, so I am making the assumption that ephones and dns are all configured on the local CMEs and you are not connecting on to a CUCM.



Have you tried running a debug voice dial peer inout on each gateway while you attempt to make a call? This will tell you a) if the outbound dp is being matched on the calling CME and b) whether the call is getting to the called CME


Cheers,

Phil

paolo bevilacqua Fri, 02/12/2010 - 04:32
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Phil,

Sometime is possible to tarot-read incomplete config.

Likely, OP has no incoming called-number to match DP, consequently call goes to DP that is G729, and fails as coded mismatch.

Jeffrey Miller Fri, 02/12/2010 - 15:21
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Hey guys,

Sorry,  I am fairly new to this and may have left out a few details.  I have posted my DP's and config files below, but as it seems Router 2 (1760) has no problem making calls to Router 1 (1760-v), which leads me to believe that I am doing something wrong in the config for Router 1.  Router 1 has 3 Ephones connected to it as well as a SIP trunk and FXO card.  Router 1 works fine independently but can not make calls to Router 2.  Router 1 is also connected to a asterisk/trixbox server, and all calls between Router 1 and Trixbox work fine.  I do however, on Router 2, have one Ephone configured and one SIP softphone.  When I call the softphone from Router 1 i get a fast busy signal, when I call the Ephone i get a slow busy signal (not sure if that is relevant or not)


Router 1 (1760-v)



!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
!
!
voice class codec 1
codec preference 1 g711ulaw
!
!
voice translation-profile 4Digits2E164
translate called 1
!
!
!
voice-port 3/0
echo-cancel coverage 32
no comfort-noise
timeouts call-disconnect 2
timing hookflash-out 500
connection plar opx 3000
description xxx xxx-xxxx
station-id name PSTN Line
caller-id enable
!
voice-port 3/1
!
!
!
!
!
!
dial-peer voice 100 voip
description ** Extensions 3xxx to Trixbox
destination-pattern 3...
monitor probe icmp-ping
session protocol sipv2
session target ipv4:192.168.x.x
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 101 voip
description *** PSTN connection on Trixbox
translation-profile outgoing E164
destination-pattern 9T
monitor probe icmp-ping
session protocol sipv2
session target ipv4:192.168.x.x
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 102 voip
description ** Extensions to trixbox_2 passed through Local Trixbox
destination-pattern 1...
monitor probe icmp-ping
redirect ip2ip
session protocol sipv2
session target ipv4:192.168.x.x
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
!
dial-peer voice 105 pots
description *** PSTN connection
preference 4
destination-pattern 7T
progress_ind setup enable 3
progress_ind progress enable 8
port 3/0
!
dial-peer voice 110 voip

description ** Connection to 1760-v
destination-pattern 4...
session target ipv4:192.168.x.x
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer inbound selection sip-trunk
!
!
!
!
sip-ua
retry register 10
timers connect 100
  sip-server ipv4:192.168.x.x (trixbox IP address)
  host-registrar
!
!
telephony-service
load 7960-7940 P00308000400
load 7921 cmterm_7920.4.0-03-02
load 7970 TERM70.7-0-1-0s
load 7912 CP7912080001SCCP051117A
max-ephones 15
max-dn 15
ip source-address 192.168.x.x port 2000
application telephony-service
calling-number local
dialplan-pattern 1 10.. extension-length 4
max-conferences 4 gain -6
moh music-on-hold.au
dn-webedit
transfer-system full-consult




Router 2 (1760)




!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
!
!
!
voice register global
mode cme
source-address 192.168.x.x port 5060
max-dn 15
max-pool 15
authenticate register
!
voice register dn  1
number 4002
!
voice register pool  1
id mac 0000.0000.0000
number 1 dn 1
username xxxx password xxxx
codec g711ulaw


!
!
!
!
!
dial-peer voice 100 voip
description *** connection to local Trixbox server EXTENSIONS
destination-pattern 3001
monitor probe icmp-ping
session protocol sipv2
session target ipv4:192.128.x.x
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 101 voip
description *** Access to 1760-v PSTN Connection
translation-profile outgoing E164
destination-pattern 7T
monitor probe icmp-ping
session target ipv4:192.168.x.x
dtmf-relay rtp-nte
codec g711ulaw
!
!
dial-peer voice 104 voip
description *** Access to Trixbox PSTN Connection
translation-profile outgoing E164
destination-pattern 9T
monitor probe icmp-ping
session protocol sipv2
session target ipv4:192.168.x.x
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 110 voip
description *** Extensions on 1760-v
destination-pattern 2...
session target ipv4:192.168.x.x
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
!


dial-peer inbound selection sip-trunk
!
!
sip-ua
no remote-party-id
retry invite 2
retry register 10
timers connect 100
sip-server ipv4:192.168.x.x (Trixbox IP address)
  host-registrar
!
!
telephony-service
max-ephones 15
max-dn 15
ip source-address 192.168.x.x port 2000
application telephony-service
calling-number local
time-zone 5
dialplan-pattern 1 10.. extension-length 4
max-conferences 4 gain -6
moh music-on-hold.au
dn-webedit
transfer-system full-consult
!
!
ephone-dn 1 dual-line
number 4001
name office phone
call-forward noan 2001 timeout 5
application telephony-services
!
!
!


I am sorry if there is redundant config parameters in here, I am still learning CME


Any help is appreciated.

Thanks

-Jeff

paolo bevilacqua Fri, 02/12/2010 - 15:27
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As indicated before. You need incoming called-number.

Remember, incomping DP is important as outgoing is.

Jeffrey Miller Fri, 02/12/2010 - 19:22
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P.Bevilacqua


Thank you for your fast reply, and please forgive me for my ignorance, but I am not familiar with Incoming Dial-Peers.  I have done a little research since your post and I'm not sure If I am doing this correctly.  Could you please verify my config snippet below.



Router 1 (2xxx)


dial-peer voice 111 voip

incoming called-number 4...

codec g711ulaw



Router 2 (4xxx)


dial-peer voice 111 voip

incoming called-number 2...

codec g711ulaw


I added these entries and still no luck.  I also should add that these routers are on the local network and are not remote.


Thanks you for your help

-Jeff

paolo bevilacqua Sat, 02/13/2010 - 03:29
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Swap around 2xxx and 4xxx.

You can also put the command under the existing voip DP:

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