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bigblokeOffline



Joined: Aug 15, 2004
Posts: 66
Location: Newport South Wales
Status: Offline
Posted: Nov 27, 2006 - 08:22 AM Reply with quote Back to top
Hi guys,

I Would have posted this to the FAQ but not a mod here

From time to time I need a "cheaper-than-orange" conference bridge facility. It would be ok for the other participants to ring me and since multiple users would be ringing me all the
call revenues generated go into the pot - sounds like an all round winner for the community service (ok as well as me!) Wink

I modified my settings to permit callwaiting and 3 way calling but it didnt play (second caller got busy tone)

before I start hacking my dialplans too much - can I just check whether there is some service provision limitation with your peering company please ? (cant see why it should be ...its all revenue after all Wink )

Regards

BB
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ianplainOffline
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Joined: Jul 05, 2004
Posts: 2772
Location: Bath UK
Status: Offline
Posted: Nov 27, 2006 - 10:13 AM Reply with quote Back to top
Hi

What you realy need to use is meetme or conference (Will require patching).
The problem you will have is your upstream bandwidth if you are using ADSL

Ian
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bigblokeOffline



Joined: Aug 15, 2004
Posts: 66
Location: Newport South Wales
Status: Offline
Posted: Nov 27, 2006 - 04:58 PM Reply with quote Back to top
Hi Ian,

I know theres the internal conference faility within * but the question being, how do I get (in fact - can voipuser's inbound service support) multiple subscribers dialing in to my asterisk box using the single E164 number ? I know it can work on BT - when the appropriate class of service is set against the line, (call hold, call waiting, hookflash, and
conference call) but I'm looking to do the hosting / bridging
within * myself

The * box is presently connected only to voipuser as its a
test platform so I can't use my pstn line as one source and the voipuser number as a second source ?

none of the other conference parties are using voip (almost certainly dialling in from mobiles) so I cant get one to use SIP URI dial - in for example, to keep the single incoming line free for a second caller.

Any thoughts ?

Regards

BB
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deanOffline
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Joined: Dec 13, 2003
Posts: 6906
Location: London
Status: Offline
Posted: Nov 27, 2006 - 05:16 PM Reply with quote Back to top
Hiya Rob,

No restrictions on that this end. Check logs - your * box is probably sending back a "busy" message. The PSTN gateway does what it's told - it will only play back a "busy" tone if that's what you told it to do... unless of course that's what our SIP server told it to do (I don't honestly know the answer to that, but I can find out). Is this SIP or IAX termination? 0844 986 **** or something else?

Quote:
How many concurrent inbound calls can 1 No. support Please


There is currently no limit, but if you are anticipating a large number of concurrent calls (say > 16) then it would be really useful if you could let me know in advance Wink

If it's a SIP number (0844 986 ****) PM me the number and I'll run a tail|grep on it on our SER server and ask you to dial it a couple of times so I can see what's happening at our end.


Last edited by dean on Nov 27, 2006 - 05:21 PM; edited 1 time in total
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ianplainOffline
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Joined: Jul 05, 2004
Posts: 2772
Location: Bath UK
Status: Offline
Posted: Nov 27, 2006 - 05:20 PM Reply with quote Back to top
Hi

I dont believe there is a limit in place I know the 0844 number I use has had many callers on the one number. Just try it. Point the number at an IVR or directly at meetme.

Your problem will be as I have said before, thats bandwidth, and with meetme, Horsepower and timing can be an issue.

How many callers are you expecting?

Ian

PS for meetme to work you need either a zaptel card or ztdummy installed
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deanOffline
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Joined: Dec 13, 2003
Posts: 6906
Location: London
Status: Offline
Posted: Nov 27, 2006 - 05:23 PM Reply with quote Back to top
Ian - do you know if the 0844 986/933 ranges also accept multiple callers? I've never tried and I'm not sure how Daniel set it up when he config'd openSER for us.
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bigblokeOffline



Joined: Aug 15, 2004
Posts: 66
Location: Newport South Wales
Status: Offline
Posted: Nov 28, 2006 - 12:47 AM Reply with quote Back to top
Thanks Chaps, Glad to know that its my end !

Something to play with.

My conferences would be usually only 3 or 4 users I imagine.

It is a SIP inbound number 0844986xxxx that the callers will be dialling. Although I am now tending to use IAX soft extensions for "internal" clients as the whole NAT thing is getting me down when working away from home.

It will certainly be my dialplan - its VERY simplistic! Wink

The idea came to me the other day as I have (in all honesty)
hardly ever used this service properly and was trying to think of good ways to swell the community coffers whilst adding service value to me. When I needed to set up a conference at very short notice between Switzerland,York, Somerset and Basingstoke it occured to me! hey! lots of inbound revenue for the outbound pot - it seemed a "win-win"
way of using the service. I "lashed up" what I thought would be a valid solution but as I said previously, it didn't work.

I had an idea relating to using 2 voipuser accounts to generate mainly inbound revenue whilst adding value to me.

Dean, if you dont mind I'll PM you with the details to ensure what I have in mind is considered "acceptable use".

Ian, summarising your observations :

I have 10mb down 512 up as my primary link with 8 mb/256 DSL as my secondary! (dual homed - latter pipe was free) - and there are rumours that my #1 ISP may well be trebling my upstream next year as part of an planned service upgrade so there shouldn't be too much of an issue.

I also have QoS on the firewall host so VOIP packets get precedence , also asterisk is now running on the firewall host itself to reduce the number of NATs by one !

I do have a digium card too with 1 x fxo 3 x fxs . Host PC is an Athlon 3200+ with 756 of RAM running primarily as a linux firewall / Wireless AP / SOHO Astersk server

Regards & thanks again both

BB
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daveOffline



Joined: Dec 17, 2004
Posts: 71

Status: Offline
Posted: Nov 28, 2006 - 12:48 AM Reply with quote Back to top
I appriciate that this post may not be what the original thread is about, but others like me may read it and could find it useful.

I have set up a very easy conference call facility with easypabx.com I don't use my voipuser number as it happens but I see no reason why you could not. I use a sipgate number as a DDI into easypabx, set the extension to the conference call extention and it works a treat.

Hope that helps someone

Dave
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