10-14-2020 12:30 PM
I'm used to setting up 2811's using SCCP phones which are pretty simple to configure. We recently went from the 7900 series to the 8800 series leading to a new router model and setting up SRST to use SIP. So far it's gone pretty poorly, I can see SRST on CUCM recognizing the phones, but the phones will never register with the 2911.
I'm using test equipment at my home, but with the exception of having a PRI everything else is pretty identical to one of our remote sites. I've Googled and followed the admin guide for the basic and optional setup. Basic didn't work but I still did the optional as a hail mary.
I'm using a pair of 8861's as my test devices and the h.323 items are to include some of the older 7945's once I get this squared away. Any help is appreciated.
version 15.6
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service sequence-numbers
!
hostname 42_2911
!
boot-start-marker
boot system flash:c2900-universalk9_npe-mz.SPA.156-3.M5.bin
boot-end-marker
!
!
card type t1 0 2
logging buffered 51200
logging console critical
!
no aaa new-model
network-clock-participate wic 2
!
!
!
!
!
!
!
ip name-server 192.168.41.100
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
isdn switch-type primary-ni
!
cts logging verbose
!
voice-card 0
!
voice service voip
no ip address trusted authenticate (tried it with 10.42.10.0 on the trusted list as well)
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 600 min 60
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
!
!
!
voice register global
default mode
no allow-hash-in-dn
max-dn 25
max-pool 8
!
voice register pool 1
translation-profile outgoing 1
id network 10.42.10.0 mask 255.255.255.0
number 1 42..
preference 10
incoming called-number
proxy 172.16.0.10 monitor probe icmp-ping
alias 1 304444.... to 4210
no digit collect kpml
dtmf-relay rtp-nte sip-notify
voice-class codec 1
!
!
!
voice translation-rule 1
rule 1 /42../ /4210/
!
!
voice translation-profile 1
translate called 1
!
!
!
!
!
!
application
service alt default
!
global
service alternate default
!
!
vxml logging-tag
license udi pid CISCO2911/K9 sn FTX1832AHML
hw-module pvdm 0/0
!
!
!
archive
log config
hidekeys
!
!
controller T1 0/2/0
cablelength long 0db
pri-group timeslots 1-12,24
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address 10.42.10.2 255.255.255.0
ip flow ingress
duplex auto
speed auto
h323-gateway voip bind srcaddr 10.42.10.2
!
interface Serial0/2/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-qsig
isdn incoming-voice voice
!
!
ip forward-protocol nd
!
no ip http server
ip http secure-server
!
!
!
!
!
control-plane
!
call fallback active
!
voice-port 0/0/0
!
voice-port 0/0/1
!
voice-port 0/0/2
!
voice-port 0/0/3
!
voice-port 0/2/0:23
!
!
!
!
!
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer cor custom
!
!
dial-peer voice 1 pots
description All Outgoing Calls 9 Prefix
destination-pattern 9T
port 0/2/0:23
!
dial-peer voice 1000 voip
description Main Number
preference 1
destination-pattern 304444....
session target ipv4:192.168.41.100
dtmf-relay h245-signal h245-alphanumeric
codec g711ulaw
!
dial-peer voice 42600 pots
incoming called-number 4260
direct-inward-dial
!
dial-peer voice 4260 pots
preference 1
destination-pattern 4260
no digit-strip
direct-inward-dial
port 0/0/0
!
!
sip-ua
retry notify 6
retry options 1
timers notify 100
registrar ipv4:192.168.41.100 expires 600
!
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.42.10.2 port 2000
max-ephones 25
max-dn 96
system message primary Phones running on backup.
system message secondary Call 123-456-7890
transfer-pattern ....
transfer-pattern .......
transfer-pattern 42..
transfer-pattern 4210
keepalive 60
default-destination 4210
voicemail 910987654321
call-forward busy 4211
call-forward noan 4211 timeout 15
Solved! Go to Solution.
10-15-2020 08:00 AM - edited 10-15-2020 08:08 AM
This would be the SRST part from our in-house configuration template for voice gateways.
[A.A.A.A] = LAN Interface IP Address (Voice Vlan) 2.3. SRST configuration The below is functions in SRST configuration that are shared between SCCP and SCCP. • MOH configuration, file(s) and multicast. • Alias configuration. • Default destination. • Date and time format. • Conference settings. • Transfer settings. • Secondary dial tone. voice translation-rule 50 rule 1 /^4662010$/ /0004646XXXXXX/ ;# X = masked number rule 2 /^4665555$/ /0004646YYYYYY/ ;# Y = masked number rule 3 /^632\(....\)$/ /+6329ZZ\1/ ;# Z = masked number rule 4 /^\+632\(.*\)/ /0\1/ rule 5 /^\+63\(.*\)/ /00\1/ rule 6 /^\+\(.*\)/ /000\1/ ! voice translation-profile SRST-IN translate called 50 !
interface Loopback1000
description **** Used for POTS MOH source ****
ip address 192.168.255.10 255.255.255.255
! 2.3.1. SCCP SRST configuration ccm-manager music-on-hold call-manager-fallback translation-profile incoming SRST-IN secondary-dialtone 0 max-conferences 8 gain -6 transfer-system full-consult timeouts interdigit 5 ip source-address [A.A.A.A] port 2000 max-ephones 128 ;set max-ephones to the maximum that router model can support max-dn 256 dual ;max-dn should be set to double amount of max-ephones system message primary SRST Failover default-destination +6329ZZZZZZ ;# Z = masked number moh SampleMOH.wav multicast moh 239.10.138.16 port 16384 route [A.A.A.A] 192.168.255.10 time-zone 41 time-format 24 date-format dd-mm-yy Note: If the site has FXS ports that are controlled via SCCP this is needed for them to operate in SRST as they will not use the normal SRST reference setup in CUCM. sccp ccm [A.A.A.A] identifier 1 version 7.0 ! sccp ccm group 10 associate ccm 1 priority 4 2.3.2. SIP SRST configuration voice service voip allow-connections sip to sip sip registrar server ! voice register global default mode system message SRST Failover max-dn 256 ;max-dn should be set to double amount of max-pool max-pool 128 ;set max-pool to the maximum that router model can support timezone 41 phone-mode phone-only ! voice register pool 1 id network x.x.x.x mask y.y.y.y ;x.x.x.x,/y.y.y.y refer to network range assigned to entire site, not just the voice vlan translation-profile incoming SRST-IN dtmf-relay rtp-nte voice-class codec 1 no vad ! sip-ua registrar ipv4:[A.A.A.A] expires 3600
You also need to verify that the SRSR configuration in CM has an IP for both fields for IPv4.
One other thing to check, make sure that the phone can get updated configuration from CM. Once simple way to do that is to either turn on or off the web interface on the phone in the device configuration in CM and then reset the phone. If this do not change you likely have an issue with the ITL on the phone and it will not trust the changes made in CM. If this is the case you need to do a reset of security settings on the phone.
10-14-2020 01:15 PM
Maybe a silly question, but have you on the SRST reference in CM configured an IP in both SCCP and SIP fields?
10-15-2020 06:09 AM - edited 10-15-2020 06:10 AM
It's registered in CUCM and assigned in the pool properly. Attaching some more info below. Should I have the mode set to CME or ESRST in Voice Register Global? Going to look at some debug logs and see if anything sticks out
42_2911_H323#show sip-ua status registrar
Line destination expires(sec) contact
transport call-id
peer
============================================================
4260 192.168.41.100 8 0.0.0.0
UDP 144EE532-E1111EB-8002C4B1-36737BB2
42_2911_H323#
42_2911_H323#Show voice register pool 1
Pool Tag 1
Config:
Network address is 10.42.10.0, Mask is 255.255.255.0
Number list 1 : Pattern is 42..
Proxy Ip address is 192.168.41.100
Default preference is 10
Incoming called number is
Alias Tag: 1
Number pattern is 4210
Alias is 304444....
DTMF Relay is enabled, rtp-nte, sip-notify
kpml signal is disabled
Lpcor Type is none
Translation-profile outgoing 1
Reason for unregistered state:
No registration request since last reboot/unregister
paging-dn: config 0 [multicast] effective 0 [multicast]
Dialpeers created:
Statistics:
Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
Registration success : 0
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time :
Last unregister request time :
Register success time :
Unregister success time :
42_2911_H323#show voice register dial-peers
42_2911_H323#
42_2911_H323# show voice register statistics
Global statistics
Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
Registration success : 0
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time :
Last unregister request time :
Register success time :
Unregister success time :
Register pool 1 statistics
Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
Registration success : 0
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time :
Last unregister request time :
Register success time :
Unregister success time :
Reason for unregistered state:
No registration request since last reboot/unregister
10-15-2020 06:46 AM
I posted once before, no idea why it failed to save. Should I have the voice register global in ESRST or CME? Also I checked to see if anything registered and got this. I'm not sure if I'm doing it correctly, but I'm trying see if there are any attempts to register or see if there is anything in debugging it and I'm getting nothing. I'm using term mon and nothing on screen or sh logging.
42_2911_H323#show sip-ua status registrar
Line destination expires(sec) contact
transport call-id
peer
============================================================
4260 192.168.41.100 70 0.0.0.0
UDP 144EE532-E1111EB-8002C4B1-36737BB2
42_2911_H323#
42_2911_H323#Show voice register pool 1
Pool Tag 1
Config:
Network address is 10.42.10.0, Mask is 255.255.255.0
Number list 1 : Pattern is 42..
Proxy Ip address is 192.168.41.100
Default preference is 10
Incoming called number is
Alias Tag: 1
Number pattern is 4210
Alias is 304444....
DTMF Relay is enabled, rtp-nte, sip-notify
kpml signal is disabled
Lpcor Type is none
Translation-profile outgoing 1
Reason for unregistered state:
No registration request since last reboot/unregister
paging-dn: config 0 [multicast] effective 0 [multicast]
Dialpeers created:
Statistics:
Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
Registration success : 0
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time :
Last unregister request time :
Register success time :
Unregister success time :
42_2911_H323#
42_2911_H323#
42_2911_H323#show voice register dial-peers
42_2911_H323#
42_2911_H323# show voice register statistics
Global statistics
Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
Registration success : 0
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time :
Last unregister request time :
Register success time :
Unregister success time :
Register pool 1 statistics
Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
Registration success : 0
Registration failed : 0
unRegister requests : 0
unRegister success : 0
unRegister failed : 0
Auto-Register requests : 0
Attempts to register
after last unregister : 0
Last register request time :
Last unregister request time :
Register success time :
Unregister success time :
Reason for unregistered state:
No registration request since last reboot/unregister
10-15-2020 08:00 AM - edited 10-15-2020 08:08 AM
This would be the SRST part from our in-house configuration template for voice gateways.
[A.A.A.A] = LAN Interface IP Address (Voice Vlan) 2.3. SRST configuration The below is functions in SRST configuration that are shared between SCCP and SCCP. • MOH configuration, file(s) and multicast. • Alias configuration. • Default destination. • Date and time format. • Conference settings. • Transfer settings. • Secondary dial tone. voice translation-rule 50 rule 1 /^4662010$/ /0004646XXXXXX/ ;# X = masked number rule 2 /^4665555$/ /0004646YYYYYY/ ;# Y = masked number rule 3 /^632\(....\)$/ /+6329ZZ\1/ ;# Z = masked number rule 4 /^\+632\(.*\)/ /0\1/ rule 5 /^\+63\(.*\)/ /00\1/ rule 6 /^\+\(.*\)/ /000\1/ ! voice translation-profile SRST-IN translate called 50 !
interface Loopback1000
description **** Used for POTS MOH source ****
ip address 192.168.255.10 255.255.255.255
! 2.3.1. SCCP SRST configuration ccm-manager music-on-hold call-manager-fallback translation-profile incoming SRST-IN secondary-dialtone 0 max-conferences 8 gain -6 transfer-system full-consult timeouts interdigit 5 ip source-address [A.A.A.A] port 2000 max-ephones 128 ;set max-ephones to the maximum that router model can support max-dn 256 dual ;max-dn should be set to double amount of max-ephones system message primary SRST Failover default-destination +6329ZZZZZZ ;# Z = masked number moh SampleMOH.wav multicast moh 239.10.138.16 port 16384 route [A.A.A.A] 192.168.255.10 time-zone 41 time-format 24 date-format dd-mm-yy Note: If the site has FXS ports that are controlled via SCCP this is needed for them to operate in SRST as they will not use the normal SRST reference setup in CUCM. sccp ccm [A.A.A.A] identifier 1 version 7.0 ! sccp ccm group 10 associate ccm 1 priority 4 2.3.2. SIP SRST configuration voice service voip allow-connections sip to sip sip registrar server ! voice register global default mode system message SRST Failover max-dn 256 ;max-dn should be set to double amount of max-pool max-pool 128 ;set max-pool to the maximum that router model can support timezone 41 phone-mode phone-only ! voice register pool 1 id network x.x.x.x mask y.y.y.y ;x.x.x.x,/y.y.y.y refer to network range assigned to entire site, not just the voice vlan translation-profile incoming SRST-IN dtmf-relay rtp-nte voice-class codec 1 no vad ! sip-ua registrar ipv4:[A.A.A.A] expires 3600
You also need to verify that the SRSR configuration in CM has an IP for both fields for IPv4.
One other thing to check, make sure that the phone can get updated configuration from CM. Once simple way to do that is to either turn on or off the web interface on the phone in the device configuration in CM and then reset the phone. If this do not change you likely have an issue with the ITL on the phone and it will not trust the changes made in CM. If this is the case you need to do a reset of security settings on the phone.
10-15-2020 08:54 AM
Went a head and factory reset both of them. The configs looked good but it's quick enough it's worth doing. One thing I noticed in sh call-manager-fallback was there are no SIP models listed and Auto-Provision is OFF. SRST in CUCM is set up properly and assigned to the pool. Is there a debug method to see what the phones are attempting to do when they are trying to failover?
CONFIG (Version=11.6)
=====================
Version 11.6
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 10.42.10.2 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 25
max-dn 96
max-conferences 4 gain -6
dspfarm units 0
dspfarm transcode sessions 0
huntstop
no huntstop channel
huntstop channel 0
default-destination 4210
voicemail 918003481623
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
time-format 12
date-format mm-dd-yy
timezone 0 Greenwich Standard Time
call-forward busy 4211
call-forward noan 4211 timeout 15
transfer-pattern ....
transfer-pattern .......
transfer-pattern 42..
transfer-pattern 4210
night-service time is deactivated
keepalive 60 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 Phones running on backup.
system message secondary Call 304-344-1623
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.
10-15-2020 10:02 AM
sh call-manager-fallback Is for SCCP SRST. You need to check this for SIP SRST. Have a look at this document for details on commands used to check SRST. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide/srst_monitoring_and_maintaining.html
10-15-2020 10:08 AM
Also check this blog for some additional information. https://colla.blog/?p=573
10-15-2020 10:46 AM
Well, I'm kicking myself right about now. I brought home equipment I've not used in some time and the VLAN interface did not have the IP address assigned. I must have forgotten to do a cop run start before I powered it off. Guess I got tunnel vision because I had never set it up like this before.
I can call from phone to phone, both H323 and SIP, we run a simple setup where incoming calls should go to 4910 then 4911 setup in the Call-fallback-manager and outgoing calls starting with 9 are sent directly to the PRI. Is my logic correct? I'd try it out, but I don't have a PRI to test it with. This is how it was configured on the 2811's as h.323 so I'm hoping it's the same and I don't need to add any translations
dial-peer voice 1 pots
description All Outgoing Calls 9 Prefix
destination-pattern 9T
port 0/2/0:23
call-manager-fallback
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.42.10.2 port 2000
max-ephones 25
max-dn 96
system message primary Phones running on backup.
transfer-pattern ....
transfer-pattern .......
transfer-pattern 42..
transfer-pattern 4210
keepalive 60
default-destination 4210
call-forward busy 4211
call-forward noan 4211 timeout 15
10-22-2020 08:02 AM
Makes more sense now, I didn't realize the call-manager-fallback was SCCP only. Unfortunately I've never had any formal training on any of this. I've only managed to pick up what I can as I've gone a long so this has been helpful. I'm guessing that's the reason SIP requires the translation-profile for incoming calls then.
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