Using Asterisk as voicemail for CCME4 - need Cisco config

[bosseye]

Hi Everyone,
Have CCME4 on a 2811 working just fine using SCCP, but need voicemail. Have Asterisk, which my colleague can configure, but cannot find any published configs for the Cisco end of things (have seen lots for asterisk though).

Does anyone have such a config they could post - I think what I need is a SIP trunk to Asterisk.
br

 

[whykap]

What IOS are u running on the 2811? As far as I know to do a SIP trunk you need advanced-enterprise IOS and in addition the Gatekeeper IPIP GW option on the router.

As far as your question goes I do not have much input. This would probably be better posted on an Asterisk forum. If it truly an SIP trunk you need (I would think you are correct) there are plenty of examples on cco. Again you would need to have the right IOS image to do so.

 

[bosseye]

Hi,
My 2811 image is the c2800nm-adventerprisek9-mz.124-11.T.bin which is pretty full featured and current. Any chance you could post me a link to a sample config, hopefully with specific regard to Asterix with vmail (as I seem unsuccessful in finding these?).
Appreciate your help.
br/

 

[whykap]

Here is link on how to configure a SIP to SIP trunk on a 2811 router. Not specific to Asterix but it should get you started:

http://www.cisco.com/en/US/products...figuration_guide_chapter09186a00806b0a14.html

Again, I believe you will need to upgrade your IOS from
c2800nm-adventerprisek9-mz to
c2800nm-adventerprisek9_ivs-mz

It requires 64MB of Flash and 256MB of DRAM minimum to run this IOS which you probably already have.

By the way I could not find anything specific to asterix either. try an asterix specific forum as I mentioned earlier and see if you come up with something there.

As always share if you come up with something fruitful.

 

[bosseye]

That was a helpful link - thanks. Actually it was so helpful it sort of diverted me from SIP trunks to a pre-requisite something else I've been hankering after for a while - mixed SCCP and SIP client support.

Simple - just take one working SCCP CME and add the following lines:

voice register global
mode cme
source-address 10.0.0.1 port 5060
max-dn 12
max-pool 12

voice register dn 1
number 109
name SIP Softphone

voice register pool 1
id mac 0013.0252.B80A
number 1 dn 1
username "109" password 109
codec g711ulaw

Have tested this with non-Cisco clients (such as X-lite/EyeBeam) and it works just fine.

In the meantime, I'm off to sort out SIP Trunks! If you turn up anything else helpful in the meantime, please post.
br

 

[bosseye]

Am at the final hurdle with CME4-Asterisk and have hit a snag re: call-forwarding on no-answer:

- for an e-phone-dn that relates to a true POTS line (ie. FXO) - all works fine.

- for an e-phone-dn that relates to a SIP account, after the timeout period (ie. call-forward noan 123 timeout 20) the incoming call (from the SIP VSP to CME) is dropped.

Any hints?

Relevant config elements are:

dial-peer voice 111111 pots
destination-pattern 111111
port 0/3/1
authentication username 111111 password <snip>

ephone-dn 24
number 111111 no-reg both
description Deskphone
call-forward noan 123 timeout 20

where 123 is a SIP Asterisk Mailbox:

voice register global
mode cme
source-address 10.0.0.1 port 5060
max-dn 24
max-pool 12

voice register dn 4
number 123
name Asterisk

voice register pool 1
id mac 0011.2233.4455
number 1 dn 4
max registrations 36
username xxx password yyy
codec g711ulaw