03-19-2019 06:06 AM - edited 03-19-2019 07:13 AM
I am attempting to upgrade VOIP phones from 7942/7962s all running SCCP on my CME routers. I have installed one SIP 8811 voip phone at our main site and have configured voice service on main CME router. The SIP phone can call all local SCCP phones and all local SCCP phones can call the new SIP phone. The new SIP phone can place a call to the remote site CME SCCP phone, however when the remote SCCP (multiple phones) try to call the new SIP it rings but when you pick up the handset the call is quickly disconnected.
The path looks like this: SCCP phone > CME > WAN (OCONUS 220ms delay) > CME > SIP and then back the same way.
I’m not voice expert by no means, and acquired this this system already built, but did see in the trace logs a “Disconnect Cause 65”
I have played with DTMF relay, codec hard coding, no coding, etc to nothing seems to work.
Below are the snipits of affected router configs. Routers are running IOS 15.6.3(M4) on all.
7962/7942 are running SCCP version 9-3-1-57 and in some 8-5 and 8811 is running sip88xx11-5-1-18
Remote Site 2 CME router Configs with all SCCP phones:
gatekeeper
shutdown
!
!
telephony-service
max-ephones 19
max-dn 23
ip source-address 20.0.2.3 port 2000 secondary 20.0.2.4
service phone webAccess 0
timeouts interdigit 5
cnf-file location flash:
load 7942 SCCP42.9-3-1SR1-1S
dialplan-pattern 1 703378.... extension-length 7 extension-pattern 222....
max-conferences 12 gain -6
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
after-hours block pattern 1 9 7-24
after-hours block pattern 2 555 7-24
after-hours block pattern 3 777 7-24
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
dial-peer voice 333 voip
destination-pattern 333....
session target ipv4:30.0.2.1
dtmf-relay h245-alphanumeric h245-signal
codec g711ulaw
ephone-dn 1
number 2224173
description Room 1/Jack #5
ephone-dn 21
number 2220101
description S02 interior phones
ephone 1
no multicast-moh
device-security-mode none
description Room 1/Jack #5 (x-9351)
mac-address 0023.EB53.FB94
after-hours exempt
button 1:1 2:21
Main site 3 CME router sp3nrtr01 with SIP and SCP phones:
telephony-service
max-ephones 4
max-dn 4
ip source-address 30.0.2.3 port 2000 secondary 30.0.2.4
service phone webAccess 0
cnf-file location flash:
load 7942 SCCP42.9-3-1SR1-1S
dialplan-pattern 1 703378.... extension-length 7 extension-pattern 333....
max-conferences 12 gain -6
transfer-system full-consult
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 600 min 60
dial-peer voice 222 voip
destination-pattern 222....
session target ipv4:20.0.2.1
dtmf-relay h245-alphanumeric h245-signal
codec g711ulaw
ephone-dn 4
number 3330100
description S03 All phones
shared-line sip
voice register global
mode cme
source-address 30.0.2.3 port 5060
max-dn 100
max-pool 100
authenticate realm al
timezone 21
tftp-path flash:
create profile sync 6343063011385032
ntp-server 30.0.1.1 mode directedbroadcast
auto-register
!
voice register dn 31
number 3331000
name SP3NPHN01
label S3 Conf RM
!
voice register dn 34
number 3330100
allow watch
shared-line max-calls 16
label S3 All Phones
!
voice register template 1
!
voice register pool 1
id mac 006C.BCBD.66E0
type 8811
number 1 dn 31
number 2 dn 34
dtmf-relay sip-notify
username cisco2 password cisco2
codec g711ulaw
no vad
Solved! Go to Solution.
03-20-2019 09:32 AM
Problem ended up being with dial-peer on Main Site router back to remote router.
Called Number=3331000(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=0
Whereby, the remote CME router has a dialplan pattern that changes the internal prefix to match the external POTs area code and prefix. The Main Site router did not no how to route the call, as the incoming/out going dial-peer was set to the local prefix.
I added the incoming called-number . to the dial peer facing the remoter router and all works.
What I'm not sure of is why this worked for SCCP related calls and not SIP calls.
03-19-2019 07:46 AM
Hi,
The way you have explained it, it looks like a codec issue.
can you Try configuring voice-class codec instead of hard coding g711 on
voice register pool 1.
Regards,
Adarsh Chauhan
Please mark helpful or Correct
03-19-2019 11:08 AM
And what would be my order of preference? It's hardcoded on the outbound dial-peers as well.
03-19-2019 10:04 PM
03-20-2019 09:32 AM
Problem ended up being with dial-peer on Main Site router back to remote router.
Called Number=3331000(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=0
Whereby, the remote CME router has a dialplan pattern that changes the internal prefix to match the external POTs area code and prefix. The Main Site router did not no how to route the call, as the incoming/out going dial-peer was set to the local prefix.
I added the incoming called-number . to the dial peer facing the remoter router and all works.
What I'm not sure of is why this worked for SCCP related calls and not SIP calls.
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