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CME and SPA3000 - Calls don't get to CME

stephan_maurer
Level 1
Level 1

Hi guys,

I'm trying to configure a SPA3000 with a CME. I would like the calls from PSTN to go the SPA and then to CME.

I have an analog phone on the SPA. If I call the analog line from PSTN the analog phone rings. That works.

But how could I route the calls to CME? Should I configure a SIP Trunk between the SPA-CME?

Has anybody done this?

Thanks.

Regards

1 Accepted Solution

Accepted Solutions

Kevin Monteiro
Level 7
Level 7

Hi Stephen,

This would behave like a sip end-point. Add it as a sip-device to the cme.

Sample config:

voice service voip

no ip address trusted authenticate

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

h323

  h225 h245-address on-connect

sip

  registrar server

!

!

voice register global

mode cme

source-address 192.168.4.97 port 5060

max-dn 10

max-pool 10

load 8961 sip8961.9-3-2-10

voicemail 5674

create profile sync 0001087425399153

!

voice register dn  1

number 1001

!

voice register dn  2

number 1010

!

voice register dn  3

number 1003

!

voice register dn  4

number 1004

!

voice register dn  5

number 1086

!

voice register template  2

feature-button 1 DnD

button-layout 1-5 line

button-layout 6-12 feature-button

dialplan 1

session-transport tcp

!

voice register pool  1

id mac ACA0.16FD.C169

type 8961

number 1 dn 1

dtmf-relay rtp-nte sip-notify

username cisco password cisco

codec g711ulaw

!

voice register pool  2

id mac A40C.C395.9450

type ata

number 1 dn 2

template 1

dtmf-relay sip-notify

username Cisco password cisco

codec g711ulaw

!

Let me know if this works.

Regards,

Kevin

View solution in original post

7 Replies 7

Kevin Monteiro
Level 7
Level 7

Hi Stephen,

This would behave like a sip end-point. Add it as a sip-device to the cme.

Sample config:

voice service voip

no ip address trusted authenticate

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

h323

  h225 h245-address on-connect

sip

  registrar server

!

!

voice register global

mode cme

source-address 192.168.4.97 port 5060

max-dn 10

max-pool 10

load 8961 sip8961.9-3-2-10

voicemail 5674

create profile sync 0001087425399153

!

voice register dn  1

number 1001

!

voice register dn  2

number 1010

!

voice register dn  3

number 1003

!

voice register dn  4

number 1004

!

voice register dn  5

number 1086

!

voice register template  2

feature-button 1 DnD

button-layout 1-5 line

button-layout 6-12 feature-button

dialplan 1

session-transport tcp

!

voice register pool  1

id mac ACA0.16FD.C169

type 8961

number 1 dn 1

dtmf-relay rtp-nte sip-notify

username cisco password cisco

codec g711ulaw

!

voice register pool  2

id mac A40C.C395.9450

type ata

number 1 dn 2

template 1

dtmf-relay sip-notify

username Cisco password cisco

codec g711ulaw

!

Let me know if this works.

Regards,

Kevin

Hi Kevin,

thanks for the info.

I tried but doesn't register.

should I configure something on the SPA? on the Line1, Proxy?

Thanks.

Regards

Hi Kevin,

got it, the config works.

The SPA registers but if i call from PSTN the analog phone rings.

How can I get it to call a sccp phone that is already registered on the CME?

Thanks

Hi Stephen,

Could you be a little specfic about the way you want to route the calls from the PSTN?

-Kevin

Hi Kevin,

I would like to forward all calls from the SPA to a sccp phone which is also registered on the CME.

I tried call-forward b2bua all but doesn't work. is this only for SIP?

Should I change this on the SPA?

Thanks.

the call fwd all must be done on the spa. it works now.

problem with audio still, i get noise between the 2 phones...

any idea? codec is the same g711ulaw.

Thanks.

felipecarneiro
Level 1
Level 1

Hello,

I'm trying to configure a SIP trunk between CME and a SPA3000. Calls from the analog phone to VoIP (SIP and SCCP) are working perfectly, but I'm unable to call from the VoIP phones to the PSTN connected on SPA3000.

How have you configured your CME and the SPA300?

Thanks in advance!

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