routing of calls

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commstechadmin
Posts: 21
Joined: Sat Feb 11, 2006 3:45 pm
Location: Toronto

routing of calls

Post by commstechadmin »

1. Brekeke Product Name and version:
Brekeke PBX, version 2.1.6.6,

2. Java version: 1.6.0_07

3. OS type and the version: Windows 2000 server

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:

Problem:
We want that when our caller made a call after a certain no of rings let say 7 rings. If nobody answer and the recipient dont have voicemail.

The call will be routed again to our PBX IVR and play a specific sound file.


Question:

1.)Can we implement it just using a "session timer" and implement
"next route on failure feature" and route it back to our PBX and
play a certain sound file?


or

2.) create a customized plugin.


If no.1 is the best choice i just need to upgrade to PRO ver.
Any



Our current ARS:
MAtching pattern:
sip:011(.+)@


Deploy Pattern:
sip:$1@xx.xx.xx.xx


Parameters:
Session timer:0
Codec: 18
Send Rtcp: ON
Rtp relay: ON

Next route on Failure: (cant set need to upgrade to Pro edition)


Any suggestion?

Thanks..
pjchacon
Posts: 30
Joined: Sun Mar 27, 2005 11:54 am
Location: Hoboken, NJ

Post by pjchacon »

Hi commstech,

I might have a solution for your issue but I need to better understand your call flow.

When a client calls your office what happens?

A. They are greeted by an Auto Attendant.
B. They are greeted by a Receptionist.
C. Each employee has their own DID (Direct Inward Dial) number.
D. You have a hunt group setup that rings at everyone's phone.
E. Other.

Thanks,
Pablo
FWD# 513461
(1) Cisco 7970, (2) Cisco 7960, (1) Cisco SPA941, (3) BudgeTone-100, ZyXEL P2000W WiFi phone, X-Lite, SJPhone, Cisco ATA 186, HT-488, OnDO PBX & SIP Server, Vonage
commstechadmin
Posts: 21
Joined: Sat Feb 11, 2006 3:45 pm
Location: Toronto

Post by commstechadmin »

Hi Pablo,

We have a DID provider. Our client call our 1800 access no. Call goes to our DID provider which route the call to our server.

It will be answer by our IVRCAT for ONDO (SVK Software). Client enter his pin which our RADIUS server check.

Were using RADIUSCAT by (SVK Software), then call is routed to our ONDO PBX then using the ARS it will route to our A-Z ITSP provider.

This setup is for our calling card client.

Our problem is some of our client call let say a certain person in Australia whos cellphone dont have a voicemail and is too busy to answer the call. The client will hear ringing.

They complain to us that we are just providing them a "fake ring" that the actual phone is not ringing.

Providing fake ring is very common practice to some calling card ITSP provider.

We want that after 7 ring it will route again to our PBX and play an IVR.

Were just using Brekeke PBX, Version 2.1.6.6 , Basic.

Thanks,
Joseph
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

using the ARS it will route to our A-Z ITSP provider.



A direction you may want to think about is having your ARS route the call to a user where in the user screen you forward the call to that user and to the A-Z destination. If there was a unlimited line phone connected to that user, you could make a call that rang on both the phone and the A-Z provider revert to an IVR if neither answered in 7 rings using the user screen no answer field. Some type of softphone with unlimited extensions would allow the extension to ring and never answer so if the callee didn't answer it would always go to ivr. Don't know off hand what softphone or device would allow many calls at the same time and never answer
commstechadmin
Posts: 21
Joined: Sat Feb 11, 2006 3:45 pm
Location: Toronto

Post by commstechadmin »

Hi Nick,

It works, but theres a little degragation on the voice quality. Which is understandable because of the added process.


Thanks,
Joseph
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

You might want to upgrade to the beta 2.2 since I know Brekeke did some work on the voice processing. Another issue that could cause voice problem(i'm assuming ivr in your case) is memory or processor. When the person answers the phone there is only one process and when the ivr kicks in there is only one process besides PBX is rated for 60 simutaneous media processes. You should be able to clear it up.
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