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IP Office - force call over PSTN rather than SCN 4

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IanRichIcom

Vendor
Oct 9, 2012
25
GB
Hi All,

I have a strange request- customer has 4 IP500 in an SCN, one of the sites Data connections is very poor at the moment and they cannot hear callers over the SCN (Lots of packet loss and Jitter) so they have asked me to route intersite traffic over the PSTN rather than SCN. However they still want the SCN to be up for Receptionists to see BLF and they still want to dial the Extension number- and not a prefix.

I have added a short Code 24N/Dial/xxxxxxxxxN/line group 0 (where xxxx is the std and first digits of ddi as extension numbers match last 4 didits of ddi and line group 0 is the PRI)

No matter what i try the system makes a match to Extensions and dials across the SCN- does anyone know of a way to change this to achieve what i need?#


Any help appreciated



Ian@Icom
 
SCN extentions are seen as extns so take priority over shortcodes.

keeping the SCN alive for the BLF info etc yet routing the calls via the PSTN is not goint to be a robust solution even if achievable (which I doubt).
I have used channel bonding on ISDN dialup networking to prove SCN issues to be the network in the past but keeping it nailed up for any length of time may be expensive.

The network issue is the true problem that needs addressing, what is the issue & what is being done to resolve it

A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear
 
HI IPGuru,

Thanks for the reply

I know the extension match takes preference over the short code but was wondering if there is another way without disabling SCN or adding a digit to dial string. I thought not but wanted some more clever brains to confirm.

The data issue is that his link from one particular site to all the others is dropping packets- and not only on the IPOffice SCN but apparently on the data side as well. We do not look after the DATA and the provider has suggested that the whole network may need to be reconfigured and this will take time. They are working on the issue but the IT Manager wants a short term fix for the sites as there is a lot of intersite traffic. 3 sites can call each other fine but the main site- 70 users- has poor speech quality. They can hear the other sites but the other sites get warbling and breakup on all calls. SSA show packet loss and jitter on these calls.

Ian@Icom
 
yes, this is/was entirely possible - I've done it! Sadly , I can't remember precisely how as it was a long time ago - and may have been retired/depracated in more recent versions (so long ago it was version 3.2)

There is a failover mechanism "built in" on SCN - and it doesn't rely on shortcode match on extension. You need to add soem shortcodes to manipulate the extension numebr intoa full PSTN number (and this is substantially easier if the DDi range is regular)





Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.
 
Just an idea (not tested):
Try to set Voice + Outgoing ch. to 0 on the SCN trunk. (IPO needs a reboot).

This might trigger the failover that Matt mentions.

Kind regards

Gunnar
__________________________________________________________________
Hippos have bad eyesight, but considering their weight, it’s hardly their problem

2cnvimggcac8ua2fg.jpg
 
Hi Matt,

If only it was that simple- the fallback only works if there is no response. in my instance there is always a response as the SCN is up- just poor quality so i cannot invoke the fallback.

Regards

Ian

Ian@Icom
 
>Try to set Voice + Outgoing ch. to 0 on the SCN trunk. (IPO needs a reboot).


Not if you do this there isn't! I'd forgotten that (thanks Gunnaro) SCN is up, but there is no channel available for the speech.

Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.
 
I do have my moments [smile]

Kind regards

Gunnar
__________________________________________________________________
Hippos have bad eyesight, but considering their weight, it’s hardly their problem

2cnvimggcac8ua2fg.jpg
 
Hi All,

I am afraid the above "fix" did not work as it appears that the SCN does not use routes in the traditional way. No matter what i change the Outbound ID to it still uses the SCN for the call. I have set this up in our Office with 2 IP500 linked via SCN with 6xx on IP500 A and 2xx on IP500 B. The ip 500 B SCN outbound ID is set to 20, shortcode ? is set to use ARS Table and in the ARS table i have a shortcode 6N/Dial/01xxxxxxxN (the xxxxxxx is actually the STD code and first 4 digits of ddi number)/ use line ID 0 (which is a BRI trunk i have on IP500B. IP500A is out inhouse system with PRI and last 3 digits of DDI match the Extn numbers.

As soon as i dial 616 for example the system uses the SCN OUTGOING ROUTE 20 and calls across the SCN. If i change the SCN outgoing to 1 or 1 or any other number it still uses this.

Me thinks this is not possible

Regards

Ian

Ian@Icom
 
>No matter what i change the Outbound ID

Not the Outbound ID! You need to change the number of voice channels available to the IP trunk to zero. This will mean that the IP can't pass the call across the IP link because you've limited the resources

Once you've done that a monitor trace might well be of use (and the Outbound group ID - if this is a guess may well need to match the ARS table id)

As I said, I did this for a customer years ago (i.e. before ARS was introduced to the product == pre 4.0). If I recall correctly, it worked ok, but (and its a big one) there is no way of controlling tromboning (i.e site A passes call to site B who then transfers call to site A - now using 3lines etc)

Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.
 
Like this:

g8coxpyskbw28olfg.jpg


Kind regards

Gunnar
__________________________________________________________________
Hippos have bad eyesight, but considering their weight, it’s hardly their problem

2cnvimggcac8ua2fg.jpg
 
Hi Both Matt and Gunnaro,

Have done this and it still does not work- Monitor shows the system trying to get to the Networked PBX user that it thinks exists in the SCN but does not select a line or route.

I am giving up on this and telling the client it is not possible when the SCN is active- unless they dial an additional digit.

Thanks for all your help and replies

Ian@Icom
 
Hey there,

use an undocumented feature, which i get from the avaya support.
create an ARS naming **NET**
put in the shortcodes you need.
set the scn channel to 0

now it worked like pstn overflow. we used this with 5 IPOs and all sites doesn't have enough bandwith on the vpn, so all calls are overflowed. only signaling is used over scn.


sorry for my bad english, i am german ;-) hope you understand what i mean
 
Intrigrant, this reminds me of a customer who only used there 1 channel SCN for signaling :)


BAZINGA!

I'm not insane, my mother had me tested!

 
That's good.


Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.
 
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