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[General] Backfeeding house with VOIP to get dialtone to alarm s »
« Voice Pulse Dropping My Phone Number ... Global NAPS?  
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JoelC707

join:2002-07-09
Tucson, AZ
clubs:

PIAF "Anonymous Caller"

I have two servers running PIAF 1.3 (Asterisk 1.6, before they finalized it). Both are using ViaTalk for trunks. I was having some issues with inbound calls not ringing through and instead getting dumped into the VT voicemail. One suggestion I found said to enable "allow anonymous inbound sip calls". I was a little leery about that but it seems to have fixed the issue. The problem is it has created a new one.

Occasionally we will get a call on either number/server that will ring to just one extension. Normally all calls should ring to a ring group so that all the phones in the office ring and I can answer the call no matter where I am. When that one extension rings the caller id displays "anonymous call". Sometimes I get legit calls like this and other times it's a junk sales call. How can an inbound call ring directly to an extension when I don't have any inbound routes configured to ring a specific extension on either server?

I have three physical phones here and a total of 6 extensions (two servers, two numbers, two extensions per phone, one for each server/number). I know I don't need two servers for two numbers but it's just the way I have it setup. When it does this it doesn't pick a single phone to ring each time. One time it might ring the one on my desk, the next it might ring the one on my uncles desk or my dad's desk. It seems to like the phone on my dad's desk more as it does ring that one more often than the others. It doesn't do it that often either, maybe a few times a month so it's just mildly annoying at this point.

kieranmullen
Premium
join:2005-12-12
Portland, OR
clubs:
·Gizmo5
·Skype
·Vitelity VOIP
·magicjack.com
·Verizon FIOS
·Vonage
·ViaTalk
·VoicePulse

said by JoelC707 See Profile :

How can an inbound call ring directly to an extension when I don't have any inbound routes configured to ring a specific extension on either server?
extenion@server eg 201@71.245.97.21

ring group eg 302@71.245.97.21

inbound route DID@server eg 15555551212@71.245.97.21
--
KieranMullen »360oregon.com

JoelC707

join:2002-07-09
Tucson, AZ
clubs:

OK, that makes sense but how would they know the extension to dial? These are people that otherwise dial our normal 10 digit number. One person in particular who this happens with often doesn't even understand VoIP really. I can see that would be how the telemarketers get to a specific extension but it would seem like the person who calls my normal number should hit the ring group as I've defined.

Regarding the telemarketers direct dialing, wouldn't they try extension 100 first considering that's the most common first extension used? I do indeed start with 100 and go up in 5's (105, 110, etc). The extension most commonly "hit" is 105 actually. Extension 100 has never been hit with this actually now that I think about it. Is this VT's doing in some odd way?

kieranmullen
Premium
join:2005-12-12
Portland, OR
clubs:
·Gizmo5
·Skype
·Vitelity VOIP
·magicjack.com
·Verizon FIOS
·Vonage
·ViaTalk
·VoicePulse

I only said how it is possible. I have no idea if that was even done in this case.

I have read that in most cases that one should not run with annon allowed, but I do. However I also have a route ANY DID with ANY CID the call will be terminated. (I do not know if that would cause any issues in the long run)

I have not recieved a telemarketing call in a long time. I do not believe many companies are telemarketing doing this yet at all.
--
KieranMullen »360oregon.com

JoelC707

join:2002-07-09
Tucson, AZ
clubs:

I would normally do the ANY DID/ANY CID route and just use that as the only inbound. The problem is I have two numbers on one of those servers and I want them to ring to different ring groups to identify which number the person called. Thus I've got to specify a DID. I too have read that anon calling should not be enabled but it seemed to cure my problems. I need to upgrade the servers, as I'm still running what amounts to a beta version Asterisk 1.6 basically. I doubt it would be the cause of my problems but it needs to be done anyway.
-
Forums » VOIP etc » Voice Over IP - VOIP » VOIP Tech Chat[General] Backfeeding house with VOIP to get dialtone to alarm s »
« Voice Pulse Dropping My Phone Number ... Global NAPS?  


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