SCN override to PSTN

[Nick M]

Hi All,

I am running 10.0.0.6 build 3 on 2 IPO 500v2 currently, they're both also running the application servers to provide voicemail.

Voicemail verions: IP Office Application Server - Linux PC R10.0.0.6

I am looking to force specific a shortcode / number range 85XX over the PSTN instead of the SCN. I require the SCN because I am currently using distributed voicemail.

I have tried this:

http://www.tek-tips.com/viewthread.cfm?qid=1734803

The issue with the above is that when I set the channels to 0, it breaks the voicemail retrieval (*17) on the remote end of the SCN.

I wish they still offered VPNM, and then I would not need voicemail over the SCN.

 

[IPGuru]

if these are extns/groups on the other system why would you be wanting to send them to PSTN?

(I see from your other post that the systems are in London & Chicago so you definitely would not want to be paying international call charges)


Do things on the cheap & it will cost you dear

 

[Nick M]

The main reason is because of latency between sites and the call quality.

We are bringing a third site online later this week in Hong Kong, and that site is roughly 200ms from Chicago and 300ms from London. Chicago to London is roughly 100ms.

So I would much rather use the PSTN.

 

[IPGuru]

in which case you are going to need to change the way users dial.
they will need to dial the full PSTN number or you will need to use a prefix in the shortcode as the IP Office always check against extn number first.

one option would to add the chicargo users into the directory & setup a speedial shortcode


sc #XXX
tn:N
feature speeddial

then the users need to simply dial # & the directory index

I would still expect you to suffer the same call quality issues when accessing the VM though



Do things on the cheap & it will cost you dear

 

[Pepp77]

300ms shouldn't be an issue, I have a customer with an IP500 in HK that has a 306ms RTT between itself and the Primary SE in Romania and it is used for ACCS calls and in the 3 months it has been in there have been no complaints of call quality as the RTT doesnt really dictate the quality, that is more to do with Loss and Jitter.

| ACSS SME |

 

[Nick M]

What codec do you use at 300ms?

 

[Pepp77]

G711 over a Direct Media Path VPLS connection.

| ACSS SME |