cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
668
Views
0
Helpful
2
Replies

Features with sip phones

Rafael Jimenez
Level 4
Level 4

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.

2 Replies 2

testeven
Cisco Employee
Cisco Employee

Hi Rafael,

Please try the following:

config t

      voice service voip

            supplementary-service h450.12

            exit

      telephony-service

            transfer pattern .T  

            end

Regards,

Tere.

If you find this post helpful, please rate!

Regards, Tere. If you find this post helpful, please rate! :)

Tere,

I will try, but I have some doubs, because all the phones are sip.