We have successfully configured Asterisk to interconnect with an Avaya CM via SES, more details on how this is done here:
Now, there are two edge cases that we are trying to work on to get this fully functional in all use cases. These are:
- Dialing outbound (Asterisk -> SES -> CM -> ISDN/PSTN) when a SIT Tone is reached (ie - disconnected or number changed message) we only hear silence on the Asterisk side and then a ‘Temporarily Unavailable’ message once the message has completed playback from the ISDN network. The reason is that Answer Supervision is never taken, so the call is never answered so Asterisk hears silence (or plays ringing if instructed in the dial cmd). The issue is that we want the caller on Asterisk to hear the message so they know if it is disconnected or changed and may update their database. The thing to keep in mind is that the extension/station dial to the calling party is over the SES to an Avaya station/handset, and this is when it may not be heard. The SIT tone may be heard if dialed with a SIP/Xlite extension right off of Asterisk.
Any ideas on this?
- Dialing inbound (ISDN/PSTN -> CM -> SES -> Asterisk) to a station, we hear silence until the call is picked up. How may we ensure the SES delivers ringing so the caller has a ‘normal’ experience? I know that Answering/Ringing in the dialplan of Asterisk is an option, but not preferred in this scenario.
Any insight from anyone on these points would be greatly appreciated. Thanks!