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Beginner

SCCP to SIP phone call drops after picking up handset

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

1 ACCEPTED SOLUTION

Accepted Solutions
Beginner

Re: SCCP to SIP phone call drops after picking up handset

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.

View solution in original post

4 REPLIES 4
Beginner

Re: SCCP to SIP phone call drops after picking up handset

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

Beginner

Re: SCCP to SIP phone call drops after picking up handset

And what would be my order of preference? It's hardcoded on the outbound dial-peers as well.

 

Highlighted
Beginner

Re: SCCP to SIP phone call drops after picking up handset

you can keep high bandwidth codec at the top like g711 and its flavors followed by low bandwidth codec like g729.
Beginner

Re: SCCP to SIP phone call drops after picking up handset

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.

View solution in original post

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