SRST Configuration for SIP Phones

[Jagsuvce G]

Hi,

We have configured SRST for SCCP IP Phones and its working fine. But for SIP IP Phones 7821, am not sure what configs to be done on Gateways.

My configs on gateway are as follow...

application
global
service alternate Default

dial-peer voice 11 voip
destination-pattern 2....
session target ipv4:192.168.1.6
session transport tcp
!
dial-peer voice 12 pots
translation-profile incoming Incoming_Pstn
destination-pattern 0T
direct-inward-dial
port 0/0/1:15
forward-digits all

call-manager-fallback
max-conferences 12 gain -6
transfer-system full-consult
ip source-address 192.168.1.6 port 2000
max-ephones 720
max-dn 900

Pls suggest what all configs needs to be added for SIP Phones to work in SRST Model.

[Rajan]

You can refer the below documents for SIP SRST configuration:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide.pdf

Or modify the configuration shared in this thread according to your needs:

https://supportforums.cisco.com/discussion/11166176/sip-configuration-srst-ver-151-3t

HTH

Rajan

[Deepak Rawat]

Look at this great document by Kevin which covers everything that you require for SIP Phones to work in SRST along with the t/s steps involved:

https://supportforums.cisco.com/document/12746456/how-implement-cisco-unified-sip-srst

Regards

Deepak

[Jagsuvce G]

Hi Deepak,

I am confused on this part...

sip-ua

registrar ipv4:192.168.0.2 expires 3600 >>> 192.168.0.2 is CUCM IP address

It should be SRST Router/Gateway IP Address right ??

[Deepak Rawat]

The syntax is correct since CUCM is B2B UA, it can do the job. More details below:

http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_r1_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1548510

Regards

Deepak

[Jagsuvce G]

Hi Deepak,

Am confused,  can u pls explain a bit on this part ??  We are using all gateways - MGCP, so does this command really required ??

sip-ua

registrar ipv4:192.168.0.2 expires 3600 >>> 192.168.0.2 is CUCM IP address

Pls suggest.

[Deepak Rawat]

This command is specific for SIP gateways, in case you have MGCP you do not need it then.

BTW, this thing was mentioned in the document I shared as well

"

4) Enable sip-ua for external registrar (required by external registrar CUCM)

 sip-ua

registrar ipv4:192.168.0.2 expires 3600 >>> 192.168.0.2 is CUCM IP address

Note:

• The CLI is to enable SIP gateway to register E.164 numbers of analog telephone voice ports (FXS) and IP phones virtual voice ports (EFXS) with an external SIP proxy or registrar server.
• Depending on the gateway type (SIP vs. MGCP) configured on CUCM, the SRST router may or may not need this configuration.

"

Regards

Deepak

[Jagsuvce G]

Thanks Deepak,

One last query, should i have to configure any dial peer "voip" and "POTS"for SIP Phones ??

[Deepak Rawat · Accepted solution]

For Call Routing, you need not to do anything different opposed to what you should be doing currently anyways i.e, the required dial-peers should be there on the gateway already to route the calls out to PSTN.

There is nothing specific for SIP vs SCCP Phones, it all depends on the dial-plan and not the type of phone.

Regards

Deepak

Regards

Deepak