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Youssef Zizou

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call transfer doesn't work on SIP Phones

Hi Experts,
I installed CME (ISR 4321 IOS version 15.5), with 40 Phones 7821, 1 IP Phone 8841, another 8851, and 2 IP Conference 8831 using SIP Protocol
all of these SIP Phones work fine, but the call transfer doesn't work, I don't know if some config missing, you find below the sh run,
thank you for your help.
CME-EHM#sh run
Building configuration...

Current configuration : 10874 bytes
!
! No configuration change since last restart
!
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no platform punt-keepalive disable-kernel-core
!
hostname CME-EHM
!
boot-start-marker
boot-end-marker
!
vrf definition Mgmt-intf
 !
 address-family ipv4
 exit-address-family
 !
 address-family ipv6
 exit-address-family
!
card type e1 0 1
!
no aaa new-model
!
subscriber templating
multilink bundle-name authenticated
!
isdn switch-type primary-net5
!
crypto pki trustpoint TP-self-signed-246793832
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-246793832
 revocation-check none
 rsakeypair TP-self-signed-246793832
!
crypto pki certificate chain TP-self-signed-246793832
 certificate self-signed 01
  ////////////////////////****////
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
  call start slow
 sip
  bind control source-interface GigabitEthernet0/0/1.30
  bind media source-interface GigabitEthernet0/0/1.30
  registrar server expires max 3600 min 1800
!
voice class codec 100
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
 codec preference 4 clear-channel
!
voice class dualtone-detect-params 77
 freq-max-deviation 30
!
voice class custom-cptone morocco
 dualtone disconnect
  frequency 435
  cadence 500 500
!
voice class custom-cptone joindre
 dualtone conference
  frequency 600 900
  cadence 300 150 300 100 300 50
!
voice class custom-cptone leave
 dualtone conference
  frequency 400 800
  cadence 400 50 200 50 200 50
!
voice register pool-type  7800
 xml-config maxNumCalls 2
 xml-config busyTrigger 2
 phoneload-support
 num-lines 2
 description Cisco IP Phone 7821
 reference-pooltype 7821
!
voice register pool-type  8841
 xml-config maxNumCalls 30
 xml-config busyTrigger 30
 phoneload-support
 num-lines 2
 description Cisco IP Phone 8841
 reference-pooltype 6961
!
voice register pool-type  8851
 xml-config maxNumCalls 30
 xml-config busyTrigger 30
 phoneload-support
 num-lines 2
 addons 1
 description Cisco IP Phone 8851
 reference-pooltype 6961
!
voice register global
 mode cme
 source-address 10.10.30.254 port 5060
 max-dn 100
 max-pool 42
 load 6961 sip88xx.11-5-1SR1-1
 load 6921 sip88xx.11-5-1SR1-1
 load 7821 sip78xx.11-0-1-11
 timezone 22
 time-format 24
 date-format D/M/Y
 tftp-path flash:
 file text
 create profile sync 000373019812351A
!
!
voice register dn  2
 number 1001
 name Samir
!
voice register dn  4
 number 0015
 name Manager
 label Manager
!
voice register dn  5
 number 0014
 name STANDARD
 label STANDARD
!

voice register template  1
 button-layout 1 line
 button-layout 2-9 speed-dial
!
voice register template  2
 softkeys hold  Resume Newcall iDivert
 softkeys idle  Newcall Redial DND Pickup Gpickup Cfwdall
 softkeys ringIn  Answer DND
 softkeys connected  Endcall Hold Trnsfer Park Confrn ConfList RmLstC
 softkeys remote-in-use  Newcall
 conference drop-mode creator
 conference add-mode creator
 conference admin
 transfer max-length 16
!
!
voice register pool  2
 busy-trigger-per-button 2
 id mac 70D3.7946.53F5
 type 7821
 number 1 dn 1
 dtmf-relay rtp-nte
 voice-class codec 100
 username 1235 password 123
 no vad
!
voice register pool  4
 busy-trigger-per-button 2
 id mac 2C31.246E.51EA
 type 8841
 number 1 dn 4
 voice-class codec 100
 no vad
 speed-dial 1 0014 label "kwgmjt"
!
voice register pool  5
 busy-trigger-per-button 2
 id mac 00C1.B1E5.B9D8
 type 8851 addon 1 CKEM
 number 1 dn 5
 template 1
 voice-class codec 100
 no vad
 speed-dial 1 0011 label "abdo"
 speed-dial 2 0012 label "karim"
 speed-dial 3 0013 label "samir"
 speed-dial 4 0014 label "you"
 speed-dial 5 0015 label "kador"
 speed-dial 6 0016 label "imrane"
 speed-dial 7 0017 label "rayan"
 speed-dial 8 0018 label "joude"
 speed-dial 9 0019 label "ibtihal"
 speed-dial 10 0020 label "hamid"
 transfer max-length 16
!
!
voice-card 0/1
 dsp services dspfarm
 no watchdog
!
voice-card 0/2
 no watchdog
!
voice-card 0/4
 no watchdog
!
license udi pid ISR4321/K9 sn FDO21172LNP
!
spanning-tree extend system-id
!
username admin privilege 15 password 0 Elite@2017
!
redundancy
 mode none
!
controller E1 0/1/0
 clock source network
 pri-group timeslots 1-31
!
!
vlan internal allocation policy ascending
!
interface GigabitEthernet0/0/0
 no ip address
 shutdown
 negotiation auto
!
!
interface GigabitEthernet0/0/1
 no ip address
 negotiation auto
!
interface GigabitEthernet0/0/1.30
 encapsulation dot1Q 30
 ip address 10.10.30.254 255.255.255.0
!
interface GigabitEthernet0/0/1.50
 encapsulation dot1Q 50
 ip address 10.10.50.254 255.255.255.0
!
interface Service-Engine0/1/0
!
interface Serial0/1/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn overlap-receiving
 isdn incoming-voice voice
 no cdp enable
!
interface Service-Engine0/2/0
!
interface Service-Engine0/4/0
!
interface GigabitEthernet0
 vrf forwarding Mgmt-intf
 no ip address
 shutdown
 negotiation auto
!
interface Vlan1
 no ip address
 shutdown
!
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:
ip tftp source-interface GigabitEthernet0
ip route 0.0.0.0 0.0.0.0 10.10.30.1
ip route 10.10.30.0 255.255.255.0 GigabitEthernet0/0/0.30
ip route 10.10.50.0 255.255.255.0 GigabitEthernet0/0/0.50
!
tftp-server flash:fbi88xx.BE-01-010.sbn
tftp-server flash:kern88xx.11-5-1SR1-1.sbn
tftp-server flash:rootfs88xx.11-5-1SR1-1.sbn
tftp-server flash:sb288xx.BE-01-019.sbn
tftp-server flash:sip88xx.11-5-1SR1-1.loads
tftp-server flash:vc488xx.11-5-1SR1-1.sbn
tftp-server flash:Busy_Hope.rwb
tftp-server flash:Callback.rwb
tftp-server flash:InRoom_Leaves.rwb
tftp-server flash:InRoom_Light.rwb
tftp-server flash:InRoom_Riff.rwb
tftp-server flash:kern8831.9-3-3-5.sebn
tftp-server flash:Mobile_Dreams.rwb
tftp-server flash:Mobile_Secret.rwb
tftp-server flash:OpenPlan_Cloud.rwb
tftp-server flash:OpenPlan_Discreet.rwb
tftp-server flash:OpenPlan_Echoes.rwb
tftp-server flash:OpenPlan_Ringer.rwb
tftp-server flash:Ringlist-wb.xml
tftp-server flash:rootfs8831.9-3-3-5.sebn
tftp-server flash:sboot8831.9-3-3-5.sebn
tftp-server flash:sip8831.9-3-3-5.loads
tftp-server flash:kern2.78xx.11-0-1-11.sbn
tftp-server flash:kern78xx.11-0-1-11.sbn
tftp-server flash:rootfs2.78xx.11-0-1-11.sbn
tftp-server flash:rootfs78xx.11-0-1-11.sbn
tftp-server flash:sboot2.78xx.11-0-1-11.sbn
tftp-server flash:sboot78xx.11-0-1-11.sbn
tftp-server flash:sip78xx.11-0-1-11.loads
!
control-plane
!
!
voice-port 0/1/0:15
 cptone AE
 bearer-cap Speech
!
!
mgcp profile default
!
telephony-service
 max-conferences 8 gain -6
 transfer-digit-collect orig-call
 transfer-system full-consult
 transfer-pattern .T
 transfer-pattern 00..
!
!
dial-peer voice 1 pots
 description ## Les Appels vers GSM / FIX ##
 destination-pattern 06........
 progress_ind setup enable 3
 progress_ind alert enable 8
 no digit-strip
 direct-inward-dial
 port 0/1/0:15
 forward-digits all
!
dial-peer voice 2 pots
 description ## Les Appels vers les numeros de service ##
 destination-pattern [1-9]...
 progress_ind setup enable 3
 progress_ind alert enable 8
 no digit-strip
 direct-inward-dial
 port 0/1/0:15
 forward-digits all
!
dial-peer voice 3 pots
 description ## VERS 15 Pompiers ##
 destination-pattern 015
 progress_ind setup enable 3
 progress_ind alert enable 8
 no digit-strip
 direct-inward-dial
 port 0/1/0:15
 forward-digits 2
!
dial-peer voice 5 pots
 description ## Les appels vers L International ##
 destination-pattern 00T
 progress_ind setup enable 3
 progress_ind alert enable 8
 no digit-strip
 direct-inward-dial
 port 0/1/0:15
 forward-digits all
!
dial-peer voice 10 voip
 description ## Les appels entrant depuis l'exterieur ##
 destination-pattern 00..
 progress_ind setup enable 3
 session target ipv4:10.10.30.254
 voice-class codec 100
 dtmf-relay h245-alphanumeric
!
sip-ua
!
line con 0
 login local
 stopbits 1
line aux 0
 stopbits 1
line vty 0 4
 privilege level 15
 login local
 transport input telnet ssh
line vty 5 15
 privilege level 15
 login local
 transport input telnet ssh
!
ntp master 6
!
end

Youssef.
Avatar of arnold
arnold
Flag of United States of America image

You need to clarify transfer to where? Between and among internal extensions?
Check whether your incoming process of call handling is such that the forwarding limit is reached on the next forward/transfer.
Might your available sip connections are exceeded.
I.e. To transfer, the connection/call you want to transfer is placed on hold, while a new outgoing connection is initiated to the destination. When you hit transfer a second time, the inbound call is setup via your ... And linked to the outgoing call. At this time you gave two sip connections consume.

Can your phone/setup handle more than one call at a time. I.e. Can you place a call on hold, and initiate a new outgoing call?

If you are forwarding/transferring to an external destination, you have to make sure to configure the system to allow that.
Deals with allowing an external/incoming call from being sent back out.
This question needs an answer!
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