My connection to the net is a business FiOS with fixed IP and i am
not about to give that up for QoS. One test on a site i found said
my connection could handle 82 SIP calls. Don't see me needing
anywhere near that.
Well 82 SIP calls would depend on the codec you used. What is your
upload speed on FiOS business class?
5Mb/sec up and 20Mb down as sold to me. I have tested slightly higher
at times at:
http://www.dslreports.com/speedtest
Which leads me to ask the question. Should i turn on QoS in my router
for SIP and RTP? I presume it can't hurt, will probably do nothing if
not using the whole pipe, and should i use the whole pipe at some
point, it would help... right?
TalkLite looks good. I think i may just go with them to start with.
They are set up to handle boxes like CallWeaver. Unlimited channels
both directions. You just pay for the minutes. Seems like the way
to go.
I have had good luck with them. However I have not ported a number
to them yet, so I'm not sure how quickly they can get it done.
Definitely contact them beforehand to work out a deal if available.
Already have. They say up to 30 days, since telcos legally have that
long to respond, but often 10 to 15 days is what they see. And i keep
the right to port the number out again if i need to move. I have
ordered a number to play with while i get up to speed with this all.
I don't want any downtime on my ported numbers so i want to make sure
i have my network and callweaver well oiled before the numbers arrive.
Are there any tools to help with LCR? Any external ACGIs to do it
in real time or preprocessors to munge call price lists and come up
with exten(sion) definitions?
I haven't used any yet, You should be able to adapt the ones for *
that are on Voip-info wiki to use with callweaver.
Ahhhh... super. Will look there.
One small configuration question. I have searched high and low to
find answer to this and no luck so far. What i want is to have
incoming calls from different DID numbers go directly to a
particular extension. What i don't know is how to get the incoming
phone number that was dialed in PSTN land. Does the incoming DID
number that was dialed appear as the extension on an incoming SIP
connection? If so, and i have an entry in sip.conf:
[talklite]
username=username
type=friend
qualify=no
secret=secret
host= ulaw.talklite.net
canreinvite=yes
disallow=all
allow=ulaw
context=incoming-did
and in extensions.conf
[extensions]
exten => 101,1,Dial(SIP/somephone,20,tr)
[incoming-did]
exten => 9995551212,Goto(extensions,101,1)
will "somephone" get dialed by someone calling 9995551212 on a land
line?
correct it will. The dialed number is in the DNID variable I
believe. Its delivered in the SIP Invite message.
The dialed number is in the DNID variable -- just found that in the *
handbook. Does the dialed number get into the EXTEN variable too, or
do i have to move it there and change the [incoming-did] context to
have:
exten => s,Set(EXTEN=${DNID})
exten => _9995551212,Goto(extensions,101,1)
exten => _9995551213,Goto(extensions,102,1)
or just do:
exten => s,1,Gotoif(9995551212==${DNID}?extensions,101,1:2)
exten => s,2,Gotoif(9995551213==${DNID}?extensions,102,1:3)
exten => s,3,Play(congestion) or something...
Peace,
Dan
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