Scott Lawrence ha scritto:
On Mon, 2009-06-01 at 23:12 +0200, Alberto wrote:
I absolutely don't know what is a 'tel' scheme URL is. Basically I
expect a SIP trunk to behave like any other gateway.
Since today I felt it's perfectly possible to call back someone that
called from the ISDN gateway through a for example a GSM gateway to
archive better rates. Other thing I felt is sipXecs dial plan does it
all. With sip trunks I feel I'm loosing this power if "number" with a
"@" will take privileged or no routes.
Again most "hard" gateway provide some form of number manipulation.
Why a sip trunk should not have such a feature?
I perfectly understand this is not legal, in fact I don't expect this
to happen for all Sip trunks. Some sort of number manipulation should
be a configurable option when it is appropriate for the ITSP
configured in the trunk. For my ITSP it's always appropriate. I own a
couple of cheap sip phones ... they're all capable of calling back all
incoming number when registered directly to my ITSP. There must be
something that could be done.
This is a subtle point - if the phone received a call from n...@itsp
(where 'NNNN' is a number), and then the user tries to call that back,
there are two ways the phone _could_ generate that call:
* It could just dial 'NNNN' as a dial string and send it to your
pbx.
If it did things that way, then we can manipulate the dial
string and use your dial plans to optimize routing.
* It could just send the sip address 'n...@itsp' back to the pbx
If the phone does this, then the pbx really can't be sure that
the number is really a PSTN number.
There are some hints available in addresses that we could exploit but
don't yet - unfortunately those hints are not always reliable.
But if I'm absolutely sure NNNN is a number for my ITSP why not simply
add a configurable switch or option in the sip trunk to manipulate
incoming numbers? This is probably a simple approach and will help and
administrator that know his ITSP behavior to bring all outgoing calls in
the dial plan again.
Last missing piece in a Sip Trunk, I'm a bit off topic I know but just
cause I'm talking about trunks... , is a very basic QoS mechanism. A
physical gateway will be limited to the number of channel. When they
are all busy it will cause the next call to fall somewhere else or
fail. Our sip trunk is always doing that? If my ITSP allows a max
fixed number of calls and this number is larger than the capacity of
my WAN we are in trouble. If my ITSP does not have max call limit, I'm
sure they will charge a poor quality call like a good one, will the
trunk eat up all calls without falling back in other gateway?
We do have an issue open for that:
http://track.sipfoundry.org/browse/XX-5871
Thanks Ranga! Anyway you don't just have an open issue ... you already
had it working in some sipxbridge beta. ;-)
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