02-11-2010 09:41 PM - last edited on 03-25-2019 07:56 PM by ciscomoderator
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
02-12-2010 03:01 AM
Do you have a copy of the whole configuration?
cheers
mark
02-12-2010 03:46 AM
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.
02-12-2010 04:25 AM
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
02-12-2010 04:32 AM
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.
02-12-2010 03:21 PM
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
02-12-2010 03:27 PM
As indicated before. You need incoming called-number.
Remember, incomping DP is important as outgoing is.
02-12-2010 07:22 PM
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
02-13-2010 03:29 AM
Swap around 2xxx and 4xxx.
You can also put the command under the existing voip DP:
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