11-04-2009 04:29 AM - edited 03-18-2019 10:47 AM
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
!
11-04-2009 08:16 AM
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
11-05-2009 08:49 AM
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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