[asterisk-users] More clarification on outbound sip channels.

[Jim Dickenson]

-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/



On May 10, 2010, at 7:35 AM, Eddie Mikell wrote:

> Jim, and all:
> 
> Thanks for the response.
> 
> If I can repeat what you are saying:  you don't have to define the multiple lines in sip.conf?

Your SIP provider will limit the number of concurrent outbound calls you can make. If you try to dial more than allowed you will get a SIP message with some error indicating all outbound channels in use.

> 
> For comparison, with my current esi setup, we have 10 outgoing lines.  If one line is busy, then the service just rolls to the next number.  This is set up with the phone service.
> 
> That doesn't have to done with outgoing sip lines?  Does the dialstatus have to be checked when a user dials out?

SIP calls set setup by talking to your SIP provider. They take care of limiting concurrency. Both inbound and outbound. You can have logic in your dialplan using functions GROUP and GROUP_COUNT to keep track of how many channels you are using. Doing this allows you to play a sound file saying all lines are busy try your call later.

If the dial command fails then ${DIALSTATUS} will have values like CHANUNAVAIL CONGESTION NOANSWER BUSY ANSWER CANCEL DONTCALL TORTURE INVALIDARGS

> 
> I understand the incoming lines - we will have a block of DID numbers, and I can check those and send to appropriate extensions.
> 
> Thanks all for helping to clarify.  I have gotten a couple of users who haven't been able to call out, and wasn't sure if I wasn't "rolling over" the sip lines properly.
> 
> Best,
> 
> Eddie Mikell
> 
> 
> 
> From: Jim Dickenson<dickenson at cfmc.com>
> Subject: Re: [asterisk-users] Multiple SIP lines.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 	<asterisk-users at lists.digium.com>
> Message-ID:<EDA8102C-B255-46E0-940D-1EF217566DDF at cfmc.com>
> Content-Type: text/plain; charset=us-ascii
> 
> I think it is typical to have some limited number of outbound channels to your SIP provider. You send all calls, up to your limit, to the same place. The phone numbers your provider gave you are used to route inbound calls to your asterisk box. You will typically have some limited number of inbound channels. All people could call the same number, again controlled by the number of channels your provider allows. A reason to have multiple inbound (DID) numbers is so you can route each number to a specific dialplan extension. You might route one number to the CEO of the company and the other to a voice tree that allows the caller to specify the person's extension they want to talk with.
> -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/ On 
> May 7, 2010, at 11:17 AM, Eddie Mikell wrote:
> 
>>> All:
>>> 
>>> Still experimenting with the asterisk server for the company.
>>> 
>>> My local phone company has given me two sip numbers to experiment with,
>>> say 444-456-1234&  444-456-5678
>>> 
>>> Calling in and out works, and I've spread a couple of the phones out
>>> with some co-workers.
>>> 
>>> My question is this:  Do I have to define multiple sip lines in either
>>> the sip.conf or the extensions.conf?
>>> 
>>> Now when I dial out, I just use
>>> 
>>> exten =>  _9.,1,DIAL(SIP/${EXTEN:1}@xx.tracfone.net).
>>> 
>>> How does it know which sip channel to use?
>>> 
>>> Hope that is clear.
>>> 
>>> Thanks for all the help.
>>> 
>>> Eddie Mikell
>> 
> 
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