Cisco WIP310 SIP WiFi handset and CUE voicemail issues

Unanswered Question
Nov 4th, 2009

Hi all,

I have a 2821 running CME 7.1 and CUE 3.2 on AIM service module and I am trying to configure it to work with the WIP310 (ex Linksys)SIP WiFi handset.

I have managed to get handset to handset calls and even managed to get a handset to connect directly to the CUE when I dial the Voicemail pilot number.

However at the moment I have 2 issues.

1) When I dial the voicemail pilot number direct I get the polite lady asking me to press '1' to configure a name. However no key presses on the handset seem to recognized. I think it may be a DTMF relay issue but I am not sure.

2)If I call another handset and let it ring it wont forward to voicemail after the configured amount of time. It just gives an unobtainable tone.

I can access the voicemail options when using an attached ATA with analogue phone though.

Any Ideas out there? The config looks as follows:

oice service voip

clid network-provided

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol pass-through g711ulaw

h323

sip

registrar server expires max 1200 min 300

!

voice register global

mode cme

source-address 192.168.2.1 port 5060

max-dn 192

max-pool 58

authenticate register

authenticate realm cme

timezone 22

time-format 24

date-format D/M/Y

dialplan-pattern 1 01481818... extension-length 3

voicemail 100

url directory http://192.168.2.1:80/localdirectory

tftp-path flash:

file text

create profile sync 001507236570587A

network-locale GB

!

voice register dn 1

number 701

call-forward b2bua busy 100

call-forward b2bua noan 100 timeout 20

allow watch

name ChrisM

no-reg

label 818701

!

voice register dn 2

number 702

call-forward b2bua busy 100

call-forward b2bua noan 100 timeout 20

allow watch

name Jon

no-reg

label 818702

!

voice register pool 1

id mac 0026.CB0E.6454

number 1 dn 1

presence call-list

dtmf-relay sip-notify

username 701 password 701

codec g711ulaw

!

voice register pool 2

id mac 0026.CB0E.661C

number 1 dn 2

presence call-list

dtmf-relay sip-notify

username 702 password 702

codec g711ulaw

!

dial-peer voice 2 voip

description **** CUE Voicemail Pilot Number ****

destination-pattern 100

b2bua

session protocol sipv2

session target ipv4:10.1.10.1

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 3 voip

description **** CUE Auto Attendant number ****

destination-pattern 150

b2bua

session protocol sipv2

session target ipv4:10.1.10.1

dtmf-relay sip-notify

codec g711ulaw

no vad

!

ephone-dn 99

number 799

description **** Fax Machine ****

call-forward busy 100

call-forward noan 100 timeout 15

hold-alert 30 originator

!

!

ephone 99

device-security-mode none

mac-address 0026.0B5C.F02C

max-calls-per-button 2

username "fax"

type ata

button 1:99

!

I have this problem too.
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williamsryan Wed, 11/04/2009 - 08:16

Hi all again,

Ok further to my previous post I have resolved part 1).

As I suspected it was something relating to the dtmf-relay. I needed to add the following to the voice register pools:

dtmf-relay rtp-nte sip-notify.

I still require some assistance with the second part though.

In addition to the previous post I am also having trouble with caller IDs not being passed when dialing from handset to handset, or the phone labels not being displayed on the handsets.

I also left this of the above configuration:

!

sip-ua

presence enable

!

any help would be greatly appreciated.

R

williamsryan Thu, 11/05/2009 - 08:49

Hi all,

Right, I have resolved issue 2) as well. The issue was to do with the Dialplan-patterns configured under the voice register global and telephon-service. This was passing the full E164 telephone number to CUE.

The only thing left now is to do with the SIP phones not picking up the Label and Username details upon registering. This means that the caller-ID is the extension number rather than name. Additionally the global Voicemail pilot configured under the voice register global is not being passed to the phones.

One more thing I need to set up is the corporate directory.

Does any one have any ideas of the URL to be passed/configured so that the SIP handsets can access it.

Thanks

R

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