05-22-2012 01:13 PM - edited 03-16-2019 11:17 AM
I have an cme 9 with 3905 sip phones and sip trunk. 100% sip. Everything works fine execpt features like hold, call waiting and transfer.
Something easy like take a call from an internal phone (or offnet call) and transfer it to another sip phone is not working!.
In the moment that I use the pause (hold) button, transfer button the current call is droped and get a dial tone again.
do I need setup somethink like service dsapp ??
Here some fragments of the configuration:
!
voice service voip
no ip address trusted authenticate
address-hiding
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
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
registrar server expires max 1200 min 300
early-offer forced
midcall-signaling passthru
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729br8
codec preference 3 g711alaw
!
!
voice register global
mode cme
source-address 192.168.100.254 port 5060
timeouts interdigit 5
max-dn 50
max-pool 25
load 3905 CP3905.9-2-1-0
authenticate register
authenticate realm cibrealm
timezone 13
time-format 24
date-format D/M/Y
tftp-path flash:
file text
create profile sync 0002315003536705
user-locale ES load CME-locale-es_ES-Spanish-8.8.2.4.tar
!
!
voice register dn 1
number 5861
call-forward b2bua noan 5555 timeout 20
allow watch
pickup-group 1
name 5861
shared-line
no-reg
label 5861
mwi
!
voice register dn 2
number 5862
allow watch
pickup-group 1
name 5862
shared-line
no-reg
label 5862
mwi
!
!
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 1
conference hardware
max-ephones 20
max-dn 50
ip source-address 192.168.100.254 port 2000
calling-number initiator
cnf-file location flash:
max-conferences 1 gain -6
web admin system name admin xxxxxxxxx
dn-webedit
time-webedit
transfer-system full-consult
!
any help will be apreciated and reted!!.
Thanks.
05-22-2012 02:21 PM
Hi Rafael,
Please try the following:
config t
voice service voip
supplementary-service h450.12
exit
telephony-service
transfer pattern .T
end
Regards,
Tere.
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05-23-2012 06:59 AM
Tere,
I will try, but I have some doubs, because all the phones are sip.
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