Hello dear community,
I’ve to deploy an * solution with 4 sites separated by a WAN (full operator VPN, not internet, no NAT, only ip routing).
Each sites will have its own PSTN gw. As endpoints, we will have sccp and softphone (sip).
As I’m coming from Cisco’s world, I think to deploy Cisco’s routers and configure some local ressources on it: transcoding, multicast moh, conference.
I planned to have one ipbx on the central sites, controlling each PSTN gw (4) by SIP. I recently discovered that rtp, beetwen sccp and sip, will always hairpinning by the asterisk, that is unacceptable because twice bandwidth will be used on the WAN. So I decided to flash all sccp to sip.
But i’m still unaware on several points:
- For full SIP integration, with n sites, what is the recommended design? Is it the n-sites = n-asterisk-with-IAX recommended? That is, how do you deal with dialplan management? dialplan overlapping? voicemail box moving for users who migrate from one site to another? with system management?
- Many users will travel beetwen our sites, so how their calls would go through the local pstn gw (ie emergency calls) for offnet calls? How will they use the local ressources (ie music on hold, conference, transcoding)?
- stop here for moment