how to enable srst feature on H323 gateway

[techguy]

Hi,

I have following commands on my h323 gateway. On CUCM, SRST is enabled and i have given my h323 gateway and SIP port is 5060.

call fallback active

call-manager-fallback

max-conferences 8 gain -6

transfer-system full-consult

ip source-address 10.1.209.1 port 2000

max-ephones 250

max-dn 500

dialplan-pattern 1 905 extension-length 11

cor incoming user1 default

I have following question, will this work for Cisco phones having SCCP protocol?

My sip phones are registered with follwoing commands

voice register global

mode cme

source-address 10.1.209.1 port 5060

max-dn 300

max-pool 165

authenticate register

[Amer rajai Sha'er]

Hello,

Why don't you use telephony service instaed of call-manager-fallback , it is better in your case.

Amer

[techguy]

I am not using CUCME as i told i am using CUCM 7.1 and H323 Gateway

[Amer rajai Sha'er · Accepted solution]

Yes i know , you can use the telephony service as a new function called CME over SRST , it gives you CME functionality over the SRST config , you can try it , it will work normally.

Amer

[techguy]

So, i can put all the config under call-manager-fallback under telephony-service? If so, what will be difference? Is there any standard configuration for Cisco IP phones (SCCP) to work under link failure?

Telephony service is not accepting the below command.

cor incoming user1 default

[Amer rajai Sha'er]

Hello,

It's a corlist , not cor and it is on the dn template, here is a template we use evertime we configure a CME-SRST

telephony-service

srst mode auto-provision all

srst ephone template 1

srst dn template 1

srst dn line-mode octo

max-ephones XX

max-dn XX

ip source-address XX.XX.XX.XX port 2000

time-zone 31

date-format dd-mm-yy

max-conferences 8 gain -6

call-forward pattern .T

moh music-on-hold.au

multicast moh 239.1.1.1 port 16384 route XX.XX.XX.XX

transfer-system full-consult

transfer-pattern .T

secondary-dialtone 9

!

ephone-dn-template 1

corlist incoming CSS-Mobile

[Marwan ALshawi]

Also you need to conisder some SIP related config

For SIP SRST mode you need to

• •-          Allow SIP to SIP calls
  
• •-          Registerer server need to be enabled to support SIP phones registrations default mode SRST you just need to to specify max DNs and phones
  
• •-          Register pool required for multiple phones – to specify for example source IP when register to the SRST
  

Srst example for SIP

Voice service voip

Allow-connection Sip to SIP

Allow-connectionSIP to H323

SIP

   Bind control source-interface [ ur lan interface ]

   Bind media source-interface [ur lan interface

   Registrar server

Voice registrar global

Max-dn [number]

Max-pool [number]

Voice register pool 1

Ip network x.x.x mask y.yy,y

Application sip.app

Dtmf-relay rtp-nte

Codec g711ul

Cor incoming [ ur cor here for SIP ]

Good luck

If helpful Rate

[techguy]

So, now i can conclude following

For SCCP SRST  ( In other words, Cisco Phones)

We use Call-manager-fallback or Telephony-Service ( which is also CME-SRST) command

For SIP SRST (Port 5060)

We use Voice registrar global command

[techguy]

what does this command do " srst mode auto-provision all "

what does this command do " srst ephone template 1 " and we didnt create any template ?

what does this command do " srst dn line-mode octo "

what does this command do " secondary-dialtone 9 "

Thanks for your support.

[Amer rajai Sha'er]

Hello,

srst mode auto-provision all "

Means when phone go to srst put all config in show run (no need to do it manually , when you go to srst , do a sh run and you will find all the config (ephone , ephone dn , etc..) in the show run.

" srst ephone template 1 "

Means when phones go to srst use template 1 for all learned phones , at the nd my post there is a template.

" srst dn line-mode octo "

Means single line has 8 channel, when the phones go to SRST , it will save the config for each dn as octo-line (Busy trigger 8)

" secondary-dialtone 9 "

When user press 9 They here secondary dialtone it's not necessary but sometimes the cutsomers asked for .

Amer

[techguy]

For incoming calls / outgoing calls during SRST mode,

what will be the dial peers??. Is there any standard and template to make it work. Thanks for consistent help.

[Amer rajai Sha'er]

Hello,

No , as i am sure you know , everyDN in the SRST is a dial-peer , so when the call come in and the extension is available , it will be forwarded direclty .

In case of outgoing , as i saw in your post this is a H.323 gateway , so the phones will use the same dial-peers that are already configured on the gateway .

Amer

[techguy]

SO YOU MEAN TO SAY, DO I NEED TO CREATE EVERY DN AND EPHONE ON GATEWAY IN ORDER TO WORK IN SRST MODE ??? lIKEWISE I REGISTERED SIP BY PUTTING FOLLOWING COMMANDS

voice register dn 1

number 1234

name ABCD

!

voice register pool 1

id mac 0004.F221.XXXX

number 1 dn 1

cor incoming user2 default

username ANYUSER password 123456

codec g711ulaw

FOR SCCP PHONES ( WHO ARE USING TELEPHONY-SERVICE ) COMMAND TO WORK UNDER SRST CONDITION, I NEED TO CREATE THOSE EXTENSION ON GATEWAY TOO. LETS SAY IF I PUT COMMAND

srst mode auto-provision all, IT CAN DOWNLOAD THE PHONE INFO FROM MY CUCM.

FOLLLOWING IS THE TELEPHONY-SERVICE CONFIG

telephony-service

srst mode auto-provision all

max-ephones 165

max-dn 500

ip source-address 10.1.211.1 port 2000

time-zone 21

time-format 24

date-format dd-mm-yy

dialplan-pattern 1 90[234567]....... extension-length 10

dialplan-pattern 2 905........ extension-length 11

max-conferences 8 gain -6

call-forward pattern .T

moh music-on-hold

multicast moh 239.1.1.1 port 16384

transfer-system full-consult

transfer-pattern .T

secondary-dialtone 9

[Amer rajai Sha'er]

Hello,

Listen , i am not really sure about SCCP and SIP at the same time as SRST , i think i read it somewhere that in case of SRST it is either SCCP or SIP but i am not sure , i am may be mistaken .

about the sccp:

You need to create the dn and ephone manually only on case you are using telephony service , not call manager fallback , so that you can configure more feature on the SRST , even you can configure extnesion mobility, so that's why procision all is better to be used.

About loading , every ip phone has a device pool , in this device pool there is a srst reference , so that the phone at the registeriation from the CCM , it will download this info into the ip phone , when the call manager become unavailable , the phone will try to register with the SRST gateway , at that time , the gateway will check if this ephone is already configured , if it is it will give the ip phone the config related to , otherwise if the auto prevision is on , the gateway will write the config into the gateway so it can be used .

Hope this is clear.

Amer

[techguy]

thanks alot for detailed info.

I have implemented SRST for SIP phones and following is the config

voice register dn  1

  number 2598

  name test

!

voice register pool  1

  id mac 0004.F224.xxxx

  number 1 dn 1

  cor incoming user2 default

  username test password 123

  codec g711ulaw

Everything is fine as i can dial outside from that extension. But when i dial inside, after message from ivr, then it seems it couldnt find the extension created on the router. I can see the virtual dial-peer created by gateway.

Any clue???