Cisco Phone 8811 call-manager-fallback not register.

[gimp]

Hi,

I've five Cisco Phone's 8811 model and an isr 4300 with the "isr4300-universalk9.16.09.03.SPA.bin" Software image.

I have configured the call-manager-fallback for the esrst, but when I try to connect the phones They did not register.

This is part of the configuration:

interface GigabitEthernet0/0/0.2
description VLAN VOICE
encapsulation dot1Q 2
ip address 192.168.30.129 255.255.255.128
no ip proxy-arp
h323-gateway voip bind srcaddr 192.168.30.129

!

voice service voip
ip address trusted list
ipv4 192.168.14.0 255.255.254.0
ipv4 192.168.10.0 255.255.254.0
ipv4 192.168.12.0 255.255.254.0
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
sip
bind control source-interface GigabitEthernet0/0/0.2
bind media source-interface GigabitEthernet0/0/0.2
rel1xx disable
min-se 3600
header-passing
registrar server
!
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g729r8
codec preference 3 g711ulaw

!voice register global
mode esrst
no allow-hash-in-dn
timeouts interdigit 6
system message BACKUP MODE
max-dn 70
max-pool 35
timezone 44
dialplan-pattern 1 05290848.. extension-length 3 extension-pattern 8..
overlap-signal
!
voice register pool 1
id network 192.168.30.128 mask 255.255.255.128
dtmf-relay rtp-nte sip-notify
voice-class codec 10
overlap-signal

!call-manager-fallback
transport-tcp-tls
secondary-dialtone 0
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 192.168.30.129 port 2000
max-ephones 10
max-dn 20 dual-line
system message primary SRST Backup-System
system message secondary No VPN to Italy
transfer-pattern 0T
keepalive 20
default-destination 100
moh-file-buffer 3000
moh enable-g711 "flash:/WAWW.wav"
time-zone 44

How Can I resolve it ? Is there an incompatibility between the phone model and the router ?

With the "sh call-manager-fallback" I see this:

isr4300-VoiceGateway_Giappone# sh call-manager-fallback
CONFIG (Version=12.3)
=====================
Version 12.3
Max phoneload sccp version 17
Max dspfarm sccp version 18
For on-line documentation please see:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html

ip source-address 192.168.30.129 port 2000
ip qos dscp:
ef (the MS 6 bits, 46, in ToS, 0xB8) for media
cs3 (the MS 6 bits, 24, in ToS, 0x60) for signal
af41 (the MS 6 bits, 34, in ToS, 0x88) for video
default (the MS 6 bits, 0, in ToS, 0x0) for serviceservice directed-pickup
max-ephones 10
max-dn 20 dual-line
max-conferences 8 gain -6
dspfarm units 0
dspfarm transcode sessions 0
huntstop
no huntstop channel
huntstop channel 0
default-destination 100
cnf-file location: system:
cnf-file option: PER-PHONE-TYPE
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line-mode single
moh flash:/WAWW.wav
time-format 12
date-format mm-dd-yy
timezone 44 Tokyo Standard Time
secondary-dialtone 0
transfer-pattern 0T
night-service time is deactivated
keepalive 20 auxiliary 30
timeout interdigit 10
timeout busy 10
timeout ringing 180
timeout transfer-recall 0
timeout ringin-callerid 8
timeout night-service-bell 12
caller-id name-only: enable
system message primary SRST Backup-System
system message secondary No VPN to Italy
Limit number of DNs per phone:
12SP: 76
7902: 76
7905: 76
7906: 76
7910: 76
7911: 76
7912: 76
7920: 76
7921: 76
7925: 76
7931: 76
7935: 76
7936: 76
7937: 76
7940: 76
7941: 76
7941GE: 76
7942: 76
7945: 76
7960: 76
7961: 76
7961GE: 76
7962: 76
7965: 76
7970: 76
7971: 76
7975: 76
7985: 76
anl: 76
ata: 76
bri: 76
CIPC: 76
vgc-phone: 76
IP-STE: 76
6921: 76
6941: 76
6961: 76
6901: 76
6911: 76
7926: 76
6945: 76
8941: 76
8945: 76
Log (table parameters):
max-size: 150
retain-timer: 15
transfer-system full-consult
transfer-digit-collect new-call
local directory service: enabled.
Extension-assigner tag-type ephone-tag.

isr4300-VoiceGateway_Giappone#

Thnaks

Michele

[Mike_Brezicky]

For SIP SRST / ESRST, do you have the SIP IP set in CUCM. Its an extra field in the SRST config, so the IP needs added twice.

Verifying SIP Registrar Configuration
To help you troubleshoot a SIP registrar and voice register pool, perform the following steps.

debug voice register errors
debug voice register events
show sip-ua status registrar

Also, try using show voice register all instead of show call-manager-failback.

[Maren Mahoney]

The 8811 phones are SIP phones, so you wouldn't see anything under telephony-service. Instead, the phone would register to voice register global, which is SIP SRST.

I'm concerned about your statement that the 8811 phone won't "register". Do you mean that it won't fail over? To test SRST, the 8811 phone would need to be registered to CUCM and will have to have learned about its SRST Reference via its Device Pool. Then you can "break" the connection between the 8811 and the CUCM servers and watch it fail over.

With the voice register pool you have configured, it will only accept registrations from phones in the IP address range 192.168.30.128/25 so be sure the 8811 is in that range when it goes through the failover process.

When you are ready to test failover, you can run debug voice register events on the router and watch the failover/registration to the SRST process.

Let us know if you have questions, and let us know how this goes.

Maren

[Vaijanath Sonvane]

Hi,
As @Maren Mahoney has mentioned, consider those items and also try below commands. I had similar issues and this configuration resolved it:

voice service voip
sip
registrar server expires max 600 min 60
!
sip-ua 
registrar ipv4:192.168.30.129 expires 600
!