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SRST Version 8.0 on VG3945 IOS Version 15.1(1)T2 does it support 99XX sip phones on srst

Dear Experts, I am wondering does SRST Version 8.0 on VG3945 IOS Version 15.1(1)T2 does it support 99XX sip phones on srst as per the sh call-manager-fallback the phone types 99XX are not included in the below list. 3945#sh call-manager-fallback CONFIG (Version=8.0) ===================== Version 8.0 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_suppo... ip source-address 10.10.5.50 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 1500 max-dn 2500 max-conferences 8 gain -6 dspfarm units 0 dspfarm transcode sessions 0 huntstop no huntstop channel huntstop channel 0 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 dd-mm-yy timezone 31 Saudi Arabia Standard Time secondary-dialtone 9 call-forward noan 37851 timeout 60 call-forward pattern .T transfer-pattern .T keepalive 30 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 MODE system message secondary SRST MODE 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 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. configuration for SIP Phones: ! voice service voip ! registrar server ! voice register global max-dn 100 max-pool 42 ! voice register dn 1 number 23000 ! voice register dn 2 number 23001 ! voice register dn 3 number 23002 ! voice register pool 1 id network 10.10.6.20 mask 255.255.255.0 dtmf-relay h245-alphanumeric codec g711alaw ! voice register pool 2 id network 10.10.6.20 mask 255.255.255.0 dtmf-relay h245-alphanumeric codec g711alaw ! voice register pool 3 id network 10.10.6.20 mask 255.255.255.0 dtmf-relay h245-alphanumeric codec g711alaw does this configuration correct if anything else is need please add to it... Thanks in advance.
2 REPLIES

Hi John.As per this doc ,

Hi John.

As per this doc , version 8.0 supports 99xx phones as regular SIP phones http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/requirements/guide/srs81spc.html

In the config reported, you can add

voice register global

mode srst

source-addres x.x.x.x port 5060

 

 

HTH

 

 

Regards

 

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

Dear , Please check the below

Dear ,

 

Please check the below configuration it should be working with SIP phones

 

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
--------
 sip
  registrar server expires max 600 min 60
-------
voice register global
 system message SRST Active
 max-dn 50
 max-pool 19
!
voice register pool  1
 id network xx.xx.xx.xx mask xx.xx.xx.xx
 dtmf-relay rtp-nte cisco-rtp sip-notify
 codec g711ulaw
 no vad
----------
sip-ua 
 registrar ipv4:yy.yy.yy.yy expires 600

 

 

 

On UCM SRST reference configuration for gateway

 

SIP Network/IP Address : yy.yy.yy.yy

SIP Port :  5060

 

 

for xx.xx.xx.xx insert the network ID

for yy.yy.yy.yy insert Gateway IP

 

Best Regards,

 

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