SRST Basic Configuration

Here is the basic configuration in order to enable SRST:

ip source-address port 2000 strict-match
max-ephones 20
max-dn 40

Don’t forget also to put the SRST Reference under the Call Manager Device Pool

SRST : Quick Definition

Survivable Remote Site Telephony (SRST) is a feature which ensures that IP Phones can continue to function even if they are unable to communicate with Call Manager. During a failure, Cisco IP Phones register with the local SRST router which provides Call Processing and Control.

With the Connection Monitor Duration, Cisco IP Phones do not fail back immediately to ensure that the Call Manager in question is back online and is stable.

Here is also another presentation of Cisco SRST ( Cisco video)

E&M : Quick Definition

E&M stands for a lot of Acronym like

  • Ear an Mouth
  • Earth and Magneto
  • Receive and Transmit

E&m uses an RJ48 interface and is often used to provide tie-line connectivity to PBX. there are 5 types of E&M Interfaces ( I, II , III , IV, V) which are all supported by Cisco except the Type IV.

On this line , we can find the following signaling :

  • Wink-start
  • delay -start
  • immediate-start
  • tone-start

Need to more explanations about the different modes !

FXO : Quick Definition

FXO is an RJ11 interface which is providing a connection to a PSTN Central Office (CO) or to a PBX interface.

A FXO is considered as a telephony line. We can then find 2 types of signaling :

  • Loop-start signaling : this is a type of supervisory signaling that allows indication of on-hook and off-hook states. But this signaling is vulnerable to a condition knows as glare that involves 2 endpoints seizing a trunk a the same time.
  • Ground-start signaling : Same as loop-start signaling but this time we suppress the glare effect on these lines.

Ringback issues

When you are working with H323 configurations, you can experience teh following ringtones issues , let’s see these issues and how to fix it quickly:

  • No Ringback tone on an IP Phone when calling a destination in the PSTN : This comes from a mistmatch between the Call Manager and the PSTN. It can be fixed with the following command progress_ind alert enable 8 under the POTS dial-peer.
  • No Ringback tone on a phone in the PSTN when calling an IP Phone: The problem is more similar with the previous one but it differs because it is coming from the other side as this time , the setup message received from the PSTN did not include a progress indicator (PI=0). Indeed, the gateway assumes the network is an ISDN end-to-end network so normally the tone must be handled by the PSTN but it is not an end-to-end call , this is why you have to update again the progress-indicator with the command progress_ind setup enable 3 which is setting the tone in-band.
  • No Ringback tone from the PSTN Phone which is forwarded from an IP Phone to another IP Phone : in this case, the issue is caused by the fact that the voice gateway is tearing down logical channel when you are transferring the call , so Audio is not sent. To fix this , you need then to change a parameter under the Call Manager setting ToSendH225UserInfoMsg and put it to True
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