SRST dial peer VOIP and POTS

[Horacio Gutierrez]

Hello,

I thought that when SRST takes effects it uses the dial peers, I understand that the POTS dial peer goes through PSTN cloud to make a call to the outside call. But how is the VOIP dial peer use when SRST takes effect.  Is VOIP dial peers used when calls go thru the WAN but the only difference the call manager is not controlling the call flow?

Thanks,

[Suresh Hudda]

SRST will take in service when CUCM is not reachable from SRST enabled router, that`s why CUCM doesnt have any call control on that router and VoIP dial peers are used when we are sending calls to IP (destination).

Suresh

[Sreekanth Narayanan]

Hi Horacio,

The SRST gateway usually will have 1 WAN connection to the CUCM, and when this goes down, IP connectivity to the CUCM is no longer there. So unless the WAN provider that you have is a fully-meshed Frame relay or MPLS connection that gives you IP connectivity to the other sites, voip dial-peers are not going to be of any use.

Only when the branch sites all have IP connectivity to each other without having to go through HQ (where UCM is usually located), voip dial-peers will be good. Else, pots dial peers are the ones that are primarily used so as to provide the users at that site continuing telephony service.

Thanks

Sreekanth

[Ayodeji Okanlawon]

Horacio,

The SRS dial-peers will have a better match in SRST than the dial-peers pointing to cucm. When phones register in SRST, pots dial-peers are automatically created for them. These dial-peers will have a more specific match than the usual wild card dial-peers you configure to send calls to cucm..

Example..

A gateway has the following dial-peers configured

dial-peer voice 4000 voip
 destination-pattern 4...
 sess targ ipv4:10.10.210.11 ! UCM Sub
 voice-class codec 1
 voice-class h323 1
!
dial-peer voice 4001 voip
 destination-pattern 4...
 sess targ ipv4:10.10.210.10 ! UCM Pub
 voice-class codec 1
 voice-class h323 1
 preference 1

In SRST the following dial-peers were created on the gateway

SiteC-RTR#sh telephony-service dial-peer
!
dial-peer voice 20001 pots
destination-pattern 4002$
huntstop
progress_ind setup enable 3
port 50/0/1
!
dial-peer voice 20002 pots
destination-pattern 4001$
huntstop
progress_ind setup enable 3
port 50/0/2

As you can the pots dial-peer for the phones 20001 and 20002 have a better match (4001$) and (4002$). The $ means end of digit...Hence when a call comes in for 4001, it will route the call to extension 4001 in SRST instead of the dial-peer 4000 or 4001.

[Yea9632]

Hello ayodeji,
In my case these dial-peers were no created automatically at all
did I missed something ?a command ? instead I created a new dial-peer but it works all the time instead of just when the wan link is down, any idea please ?