[asterisk-biz] asterisk-biz Digest, Vol 38, Issue 72

david.verenzuela at celnova.net david.verenzuela at celnova.net
Wed Sep 26 11:40:26 CDT 2007


Hello all

I can give you some advices about your questions:

>> - Each work-at-home person should only need a headset + softphone
>> software on their PC, right?   (Is the software audio quality as good
>> as separate hardware SIP phones now, if run on a modern-speed PC?
>> Recommendation for best?)
>>

Answer: A work-at-home person only need headset + softphone + good  
quality Internet link.

At the moment there are a variety of softphones in the market, some of  
then good some bad, I personal recomend to you "idefisk" you can  
download at www.asteriskguru.com (it's IAX and SIP). Respect to the  
audio quality it depends of the computer and the aplications that you  
have running on certain moment, while more applications you have  
running on your PC the audio quality degrees. If you goint to use  
softphone or IP Phones I recomend to you do it using IAX Protocol  
becauses the comunications it's over internet, on a LAN enviroment I  
recomend SIP IP PHONES.

>> - To have land-line (incoming) phone numbers in a country, should we
>> have an Asterisk server in that country?  A rack-mount server in a
>> telco/colo center, that receives the calls in on a Digium card then
>> SIP-routes them out to the work-at-home SIP phones?   (I'm assuming it
>> shouldn't be all-one-central server, since calls from Australia to
>> Australia would actually be crossing the ocean twice, reducing call
>> quality.)
>>

Yes, your rigth you need to have an asterisk server on each country,  
this it will help you to build an stablish and well comunications  
plataform, the quality of the calls it will be great to the agents. It  
will be an error asuming to just have one server for register all the  
agents around the world, it's just will not work.

>> - Can Asterisk report-back IAX-softphone availability, without needing
>> to pass a call?   So our website can say how many agents are available
>> (in which countries) to take their call now?

I'dont know exactly what you mean when you say report-back. When you  
have all yours server around the wolrd connected to your Main server  
in your country you will be able to link to each server and find out  
the status of all your system Asterisk, that includes Agents, Queues,  
etc..

>>
>> - Is there a better way you'd recommend setting up phones for the
>> "SITUATION" described, above?   Since there is no legacy phone number
>> or contract, it could be 100% VoIP.

My best recomendation to you it's to build a distribuite plataforms of  
Asterisk around the worl, it depends of how many agents you will have  
to determine the dimension of the server, and each of this server  
conected to the Telco by analog pr digital lines. Also yo have to  
calculate the number of simultaneos calls that you will have on  
certain time.

I hope this information help you to build your plataform.

Please contact me for more help

Quoting asterisk-biz-request at lists.digium.com:

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> Today's Topics:
>
>    1. Re: consultant questions : glad to pay for email	reply
>       (Sarfaraz Chougule)
>    2. Asterisk Venezuela certified (david.verenzuela at celnova.net)
>    3. Re: consultant questions : glad to pay for	email	reply
>       (Dave Walker)
>    4. Asterisk Venezuela - Certified Experience
>       (david.verenzuela at celnova.net)
>    5. Re: consultant questions : glad to pay for email	reply (Enky)
>    6. Inbound IP Trunks - Lata 132 (Mike Roberts)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 26 Sep 2007 07:00:26 -0700
> From: "Sarfaraz Chougule" <sarfaraz.chougule at gmail.com>
> Subject: Re: [asterisk-biz] consultant questions : glad to pay for
> 	email	reply
> To: "Commercial and Business-Oriented Asterisk Discussion"
> 	<asterisk-biz at lists.digium.com>
> Message-ID:
> 	<1c639ee50709260700q7cebfe74vf1fcb8b268f266ef at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello Miles,
>
> Please see few options for your questions below.
>
> - Each work-at-home person should only need a headset + softphone
> software on their PC, right?   (Is the software audio quality as good
> as separate hardware SIP phones now, if run on a modern-speed PC?
> Recommendation for best?)
> *My Option - Any standard configuration PC should work. Good and free
> Softphones are available to download, they provide very good quality.)*
> - To have land-line (incoming) phone numbers in a country, should we
> have an Asterisk server in that country?  A rack-mount server in a
> telco/colo center, that receives the calls in on a Digium card then
> SIP-routes them out to the work-at-home SIP phones?   (I'm assuming it
> shouldn't be all-one-central server, since calls from Australia to
> Australia would actually be crossing the ocean twice, reducing call
> quality.)
> *My Option - In this case I would setup my Asterisk server in office to make
> outbound call (here you might need to purchase intenational calling minutes
> from any VoIP provider) to terminate calls to country's landline number (say
> your agent's home phone number)*
> - Can Asterisk report-back IAX-softphone availability, without needing
> to pass a call?   So our website can say how many agents are available
> (in which countries) to take their call now?
> *My Option - Asterisk portal (if you have installed) can display phones
> registered (online).
> *- Is there a better way you'd recommend setting up phones for the
> "SITUATION" described, above?   Since there is no legacy phone number
> or contract, it could be 100% VoIP.
> *My Option - the options above are pretty much easy to setup.*
>
>
>
>
> On 9/26/07, Miles Keaton <mileskeaton at gmail.com> wrote:
>>
>> I've got some Asterisk-consultant questions for any experts here.
>>
>> I was going to try to find a consultant first then ask them these
>> questions, but decided to just post it to the list, and I'll be glad
>> to PayPal $50 each to the first few people who give through replies to
>> the 4 questions below.
>>
>>
>> SITUATION:
>>
>> - We already have a 100% Asterisk setup in our USA office, for the
>> past year, working well.
>>
>> - We're going to be setting up international offices, with people
>> working from home in their own country.
>>
>> - They need to be available by a regular incoming phone number in
>> their country, but the number has to be ours.   (So, in case the
>> person flakes, we can route calls to a different person.)
>>
>> - Call-roundabout (seeking?) setup, where one central number can ring
>> the next-available-agent (many agents, each working from home).
>>
>> - Work-at-home agents should be able to make outgoing international
>> calls through our system.
>>
>> - All calls (in and out) should be recorded and logged.
>>
>>
>> QUESTIONS:
>>
>> - Each work-at-home person should only need a headset + softphone
>> software on their PC, right?   (Is the software audio quality as good
>> as separate hardware SIP phones now, if run on a modern-speed PC?
>> Recommendation for best?)
>>
>> - To have land-line (incoming) phone numbers in a country, should we
>> have an Asterisk server in that country?  A rack-mount server in a
>> telco/colo center, that receives the calls in on a Digium card then
>> SIP-routes them out to the work-at-home SIP phones?   (I'm assuming it
>> shouldn't be all-one-central server, since calls from Australia to
>> Australia would actually be crossing the ocean twice, reducing call
>> quality.)
>>
>> - Can Asterisk report-back IAX-softphone availability, without needing
>> to pass a call?   So our website can say how many agents are available
>> (in which countries) to take their call now?
>>
>> - Is there a better way you'd recommend setting up phones for the
>> "SITUATION" described, above?   Since there is no legacy phone number
>> or contract, it could be 100% VoIP.
>>
>>
>> Thanks!
>>
>> _______________________________________________
>>
>> Sign up now for AstriCon 2007!  September 25-28th.
>> http://www.astricon.net/
>>
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-biz mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-biz
>>
>
>
>
> --
>   With Best Regards,
> **************************
>    Sarfaraz Chougule
>
> **************************
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>
> Message: 2
> Date: Wed, 26 Sep 2007 08:23:30 -0600
> From: david.verenzuela at celnova.net
> Subject: [asterisk-biz] Asterisk Venezuela certified
> To: asterisk-biz at lists.digium.com, info at eagertech.com
> Message-ID: <20070926082330.rmykyx65hcw4k0os at www.celnova.net>
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>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 26 Sep 2007 15:34:44 +0100
> From: Dave Walker <DaveWalker at ubuntu.com>
> Subject: Re: [asterisk-biz] consultant questions : glad to pay for
> 	email	reply
> To: Commercial and Business-Oriented Asterisk Discussion
> 	<asterisk-biz at lists.digium.com>
> Message-ID: <1190817284.5550.60.camel at dave-laptop>
> Content-Type: text/plain; charset="us-ascii"
>
>
> On Wed, 2007-09-26 at 13:31 +0100, Miles Keaton wrote:
> <SNIP>
>>
>> QUESTIONS:
>>
>> - Each work-at-home person should only need a headset + softphone
>> software on their PC, right?   (Is the software audio quality as good
>> as separate hardware SIP phones now, if run on a modern-speed PC?
>> Recommendation for best?)
>
> A softphone will only be good as the headset.   A good quality (ie,
> plantronics) headset will be equally as good as a conventional SIP
> handset.  The only things you need to worry about, is the potential need
> for port forwarding.
>
>> - To have land-line (incoming) phone numbers in a country, should we
>> have an Asterisk server in that country?  A rack-mount server in a
>> telco/colo center, that receives the calls in on a Digium card then
>> SIP-routes them out to the work-at-home SIP phones?   (I'm assuming it
>> shouldn't be all-one-central server, since calls from Australia to
>> Australia would actually be crossing the ocean twice, reducing call
>> quality.)
>
> If the call is being routed via SIP anyway - would it not seem logical
> to look for a SIP provider in that country and route the entire call via
> SIP.  One place you could consider getting a DDI is didx.net.
>
> If you keep the entire solution SIP, then you should look at
> round-trip-times between the * server and the outbound supplier.  If
> this is short, then generally - once the call is on their network then
> the geographical location shouldn't matter.  However, this is dependant
> on a good SIP supplier :)
>
>> - Can Asterisk report-back IAX-softphone availability, without needing
>> to pass a call?   So our website can say how many agents are available
>> (in which countries) to take their call now?
>
> Yes, this can work with both SIP and IAX clients.  Will require some
> ingenuity, and calling asterisk commands.  If the website receives many
> hits, it might be better to output to txt file or a db of the status.
> Then the PHP (or equiv) parses the file and shows who is available.
> There are some examples of this method floating around on the net.
>
>> - Is there a better way you'd recommend setting up phones for the
>> "SITUATION" described, above?   Since there is no legacy phone number
>> or contract, it could be 100% VoIP.
>>
>
> It sounds like you are going about this the right way, ask to demo
> providers services - If they won't allow it, then go elsewhere :)
>
> Kind Regards,
> Dave Walker
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> ------------------------------
>
> Message: 4
> Date: Wed, 26 Sep 2007 08:38:40 -0600
> From: david.verenzuela at celnova.net
> Subject: [asterisk-biz] Asterisk Venezuela - Certified Experience
> To: asterisk-biz at lists.digium.com, info at eagertech.com
> Message-ID: <20070926083840.xobdmcp4gocsg4os at www.celnova.net>
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>
> Hello
>
> My name it's David Verenzuela I work with Asterisk since 2005, I have
> a company named CELNOVA in VENEZUELA. I'm developing an IP PBX
> solution over Asterisk. At the moment I'm on a training Bootcamp to
> get the Dcap certification.
>
> I'm computing license from the central university of Venezuela (UCV)
> and also have a certification on Linux Administration.
>
> I'm offering my services of consulting on Asterisk in Venezuela
>
> Let me know by my email: david.verenzuela at celnova.net
>
> Lic. David Verenzuela
> Celnova Consultores C.A.
> Phone: 582122561950
> Celphone: 5804125647699
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Wed, 26 Sep 2007 17:52:35 +0300
> From: "Enky" <asterisk at bgopen.net>
> Subject: Re: [asterisk-biz] consultant questions : glad to pay for
> 	email	reply
> To: "Commercial and Business-Oriented Asterisk Discussion"
> 	<asterisk-biz at lists.digium.com>
> Message-ID: <017801c8004c$e0421200$020210ac at master>
> Content-Type: text/plain; format=flowed; charset="windows-1251";
> 	reply-type=original
>
> Miles,
>
> I am not sure someone will really help you for 50 USD :)
>
> 1. Anyway, be sure, the hardware VoIP phone (SIP or better IAX2) is much
> better than any softphone. At least you can count on commercial codecs, like
> G.729 with the hardware phone. The simplest way to check this is to
> personally try and compare :) I recommend to use low-cost IAX2 hardware
> phone (the price is around 50 USD) because there may be some NAT issues with
> SIP clients in some countries.
>
> 2. If you are looking for most flexible and cheap solution - make a try to
> find DID providers in these countries you are interested in. The price for
> DID is about 5 USD/month each, which is surely cheaper than hosting own
> equipment and paying telco's monthly fees in every country. This will solve
> the internet connectivity too, so you may count on better quality.
>
> 3. Easiest way to track the connected clients is if you use Asterisk
> Realtime and just query the database (say MySQL) when you need.
>
> 4. Best is to use own dialing plan and own extensions. Then you can map the
> real DIDs just like you wish.
>
>
> ----- Original Message -----
>> < Miles Keaton>
>>> QUESTIONS:
>>>
>>> - Each work-at-home person should only need a headset + softphone
>>> software on their PC, right?   (Is the software audio quality as good
>>> as separate hardware SIP phones now, if run on a modern-speed PC?
>>> Recommendation for best?)
>>>
>>> - To have land-line (incoming) phone numbers in a country, should we
>>> have an Asterisk server in that country?  A rack-mount server in a
>>> telco/colo center, that receives the calls in on a Digium card then
>>> SIP-routes them out to the work-at-home SIP phones?   (I'm assuming it
>>> shouldn't be all-one-central server, since calls from Australia to
>>> Australia would actually be crossing the ocean twice, reducing call
>>> quality.)
>>>
>>> - Can Asterisk report-back IAX-softphone availability, without needing
>>> to pass a call?   So our website can say how many agents are available
>>> (in which countries) to take their call now?
>>>
>>> - Is there a better way you'd recommend setting up phones for the
>>> "SITUATION" described, above?   Since there is no legacy phone number
>>> or contract, it could be 100% VoIP.
>>>
>>>
>>> Thanks!
>>>
>>> _______________________________________________
>>>
>>> Sign up now for AstriCon 2007!  September 25-28th.
>>> http://www.astricon.net/
>>>
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
>>> asterisk-biz mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-biz
>>>
>>>
>>
>>
>> _______________________________________________
>>
>> Sign up now for AstriCon 2007!  September 25-28th.
>> http://www.astricon.net/
>>
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-biz mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-biz
>>
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 26 Sep 2007 11:30:40 -0400
> From: "Mike Roberts" <mroberts1818 at gmail.com>
> Subject: [asterisk-biz] Inbound IP Trunks - Lata 132
> To: asterisk-biz at lists.digium.com
> Message-ID:
> 	<46aaf2230709260830m40509105g180f2c8e116840a0 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Xchange Telecom a CLEC based out of NYC is now offering an IP only Inbound T
> (24 channels) for DIDs in lata 132 only for $168 per month, one time setup
> $500 waived with 2 yr commitment - contact Ozzie for more info - 6467227228
>
> -Mike
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> _______________________________________________
>
> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/
>
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