Hello, Is it possible to implement Asterisk as a Gateway? For example like Cisco 5300 or 5400 with 4 T1. I was planning to buy Digium 4 port T1 card and make Asterisk as a Gateway, which will do the call routing. Any ideas? Thanks, Neal
On 12/28/05, Nitesh Divecha <nitesh@vipernetworks.com> wrote:> Hello, > > Is it possible to implement Asterisk as a Gateway? For example like > Cisco 5300 or 5400 with 4 T1. > > I was planning to buy Digium 4 port T1 card and make Asterisk as a > Gateway, which will do the call routing. > > Any ideas?Sure. It's very possible. However, if you're looking to do 4 DS1's worth of g.729 compression make sure you have at least a p4 class machine to handle the encoding. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/
Yes, asterisk makes a better voip to pstn gateway then Cisco. Asterisk has more advanced call routing and restrictions then Cisco gear. Nitesh Divecha wrote:> Hello, > > Is it possible to implement Asterisk as a Gateway? For example like > Cisco 5300 or 5400 with 4 T1. > > I was planning to buy Digium 4 port T1 card and make Asterisk as a > Gateway, which will do the call routing. > > Any ideas? > > Thanks, > Neal > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Thanks James, Are there any examples of dial plans? Like how to make the default context? I just need a kick start on the config part, as I am really struggling on routing the calls. Thanks, Neal On Dec 28, 2005, at 3:08 PM, James Sizemore wrote:> Yes, asterisk makes a better voip to pstn gateway then Cisco. > Asterisk has more advanced call routing and restrictions then > Cisco gear. > > > Nitesh Divecha wrote: > >> Hello, >> >> Is it possible to implement Asterisk as a Gateway? For example >> like Cisco 5300 or 5400 with 4 T1. >> >> I was planning to buy Digium 4 port T1 card and make Asterisk as >> a Gateway, which will do the call routing. >> >> Any ideas? >> >> Thanks, >> Neal >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >Nitesh Divecha VoIP/Network Engineer Viper Networks 10373 Roselle St. Ste:170 San Diego, CA. 92121 Phone: 858-452-8737 Fax: 858-452-8638 Cell: 1-909-964-5181 vPhone: 544-416-0067 Email: nitesh@vipernetworks.com Web: www.vipernetworks.com "Your Internet Phone Company" A publicly traded Company, OTC: VPER
Nitesh Divecha wrote:
> Are there any examples of dial plans? Like how to make the default
> context?
>
> I just need a kick start on the config part, as I am really struggling
> on routing the calls.
>
Here is a very very simple example using a PRI. You will need more error
routing in a real dial plan:
extensions.conf:
[general]
static=yes
writeprotect=no
country=us
[local]
include => default
[globals]
TRUNK=Zap/g1
LDTRUNK=Zap/g2
[trunk]
;Long distance pstn
exten => _1NXXNXXXXXX,1,Dial(${LDTRUNK}/${EXTEN})
exten => _1NXXNXXXXXX,2,Hangup
;pstn
exten => _X.,1,Dial(${TRUNK}/${EXTEN})
exten => _X.,2,Hangup
[default-out]
;This is where you sent trusted calls from sip.conf out to pstn
include => trunk
[default]
;you send incoming pstn calls here as well as untrusted voip calls.
;here you would route call to local numbers you own via enum or static.
exten => 6153247060,1,Wait(2) ; you need to wait
; long enough to get
; CNAM off line
;send incoming call to your register server.
exten => 5555554444,2,Dial(SIP/5555554444@inside-voip.com)
sip.conf:
[general]
bindport = 5060
bindaddr = 0.0.0.0
context = default ; non trusted call from sip side go here
srvlookup = yes
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
allow=g729
[trusted]
type=friend
context=default-out ; trusted call can go out pstn
host=192.168.0.1
canreinvite=no
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
dchan=96
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=default ;pstn incoming call go here
switchtype=national
signalling=pri_cpe
toneduration=500
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=-1.0
txgain=-1.0
callerid=asreceived
;
group=1
channel=>1-23
channel=>73-95
;
group=2
channel=>25-47
channel=>49-71
Thanks James, That should help to start my project.... Thanks a million... I will keep on updating.. And thanks to all for the inputs.... Thanks, Neal On Dec 29, 2005, at 6:39 AM, James Sizemore wrote:> Nitesh Divecha wrote: > > Are there any examples of dial plans? Like how to make the default > > context? > > > > I just need a kick start on the config part, as I am really > struggling > > on routing the calls. > > > > > Here is a very very simple example using a PRI. You will need more > error routing in a real dial plan: > > extensions.conf: > [general] > static=yes > writeprotect=no > country=us > > [local] > include => default > > [globals] > TRUNK=Zap/g1 > LDTRUNK=Zap/g2 > > [trunk] > ;Long distance pstn > exten => _1NXXNXXXXXX,1,Dial(${LDTRUNK}/${EXTEN}) > exten => _1NXXNXXXXXX,2,Hangup > > ;pstn > exten => _X.,1,Dial(${TRUNK}/${EXTEN}) > exten => _X.,2,Hangup > > [default-out] > ;This is where you sent trusted calls from sip.conf out to pstn > include => trunk > > [default] > ;you send incoming pstn calls here as well as untrusted voip calls. > ;here you would route call to local numbers you own via enum or > static. > exten => 6153247060,1,Wait(2) ; you need to wait > ; long enough to get > ; CNAM off line > ;send incoming call to your register server. > exten => 5555554444,2,Dial(SIP/5555554444@inside-voip.com) > > > > sip.conf: > > [general] > bindport = 5060 > bindaddr = 0.0.0.0 > context = default ; non trusted call from sip side go here > srvlookup = yes > dtmfmode=info > disallow=all > allow=ulaw > allow=alaw > allow=g729 > > [trusted] > type=friend > context=default-out ; trusted call can go out pstn > host=192.168.0.1 > canreinvite=no > > > > zaptel.conf: > span=1,1,0,esf,b8zs > bchan=1-23 > dchan=24 > span=2,1,0,esf,b8zs > bchan=25-47 > dchan=48 > span=3,1,0,esf,b8zs > bchan=49-71 > dchan=72 > span=4,1,0,esf,b8zs > bchan=73-95 > dchan=96 > loadzone = us > defaultzone=us > > > zapata.conf: > [channels] > context=default ;pstn incoming call go here > switchtype=national > signalling=pri_cpe > toneduration=500 > usecallerid=yes > hidecallerid=no > callwaitingcallerid=yes > echocancel=yes > echocancelwhenbridged=yes > echotraining=800 > rxgain=-1.0 > txgain=-1.0 > callerid=asreceived > ; > group=1 > channel=>1-23 > channel=>73-95 > ; > group=2 > channel=>25-47 > channel=>49-71 > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >Nitesh Divecha VoIP/Network Engineer Viper Networks 10373 Roselle St. Ste:170 San Diego, CA. 92121 Phone: 858-452-8737 Fax: 858-452-8638 Cell: 1-909-964-5181 vPhone: 544-416-0067 Email: nitesh@vipernetworks.com Web: www.vipernetworks.com "Your Internet Phone Company" A publicly traded Company, OTC: VPER
The line that reads: exten => 6153247060,1,Wait(2) should have been: exten => 5555554444,1,Wait(2) Nitesh Divecha wrote:> Thanks James, > > That should help to start my project.... Thanks a million... > > I will keep on updating.. > > And thanks to all for the inputs.... > > Thanks, > Neal > > > On Dec 29, 2005, at 6:39 AM, James Sizemore wrote: > >> Nitesh Divecha wrote: >> > Are there any examples of dial plans? Like how to make the default >> > context? >> > >> > I just need a kick start on the config part, as I am really >> struggling >> > on routing the calls. >> > >> >> >> Here is a very very simple example using a PRI. You will need more >> error routing in a real dial plan: >> >> extensions.conf: >> [general] >> static=yes >> writeprotect=no >> country=us >> >> [local] >> include => default >> >> [globals] >> TRUNK=Zap/g1 >> LDTRUNK=Zap/g2 >> >> [trunk] >> ;Long distance pstn >> exten => _1NXXNXXXXXX,1,Dial(${LDTRUNK}/${EXTEN}) >> exten => _1NXXNXXXXXX,2,Hangup >> >> ;pstn >> exten => _X.,1,Dial(${TRUNK}/${EXTEN}) >> exten => _X.,2,Hangup >> >> [default-out] >> ;This is where you sent trusted calls from sip.conf out to pstn >> include => trunk >> >> [default] >> ;you send incoming pstn calls here as well as untrusted voip calls. >> ;here you would route call to local numbers you own via enum or static. >> exten => 6153247060,1,Wait(2) ; you need to wait >> ; long enough to get >> ; CNAM off line >> ;send incoming call to your register server. >> exten => 5555554444,2,Dial(SIP/5555554444@inside-voip.com) >> >> >> >> sip.conf: >> >> [general] >> bindport = 5060 >> bindaddr = 0.0.0.0 >> context = default ; non trusted call from sip side go here >> srvlookup = yes >> dtmfmode=info >> disallow=all >> allow=ulaw >> allow=alaw >> allow=g729 >> >> [trusted] >> type=friend >> context=default-out ; trusted call can go out pstn >> host=192.168.0.1 >> canreinvite=no >> >> >> >> zaptel.conf: >> span=1,1,0,esf,b8zs >> bchan=1-23 >> dchan=24 >> span=2,1,0,esf,b8zs >> bchan=25-47 >> dchan=48 >> span=3,1,0,esf,b8zs >> bchan=49-71 >> dchan=72 >> span=4,1,0,esf,b8zs >> bchan=73-95 >> dchan=96 >> loadzone = us >> defaultzone=us >> >> >> zapata.conf: >> [channels] >> context=default ;pstn incoming call go here >> switchtype=national >> signalling=pri_cpe >> toneduration=500 >> usecallerid=yes >> hidecallerid=no >> callwaitingcallerid=yes >> echocancel=yes >> echocancelwhenbridged=yes >> echotraining=800 >> rxgain=-1.0 >> txgain=-1.0 >> callerid=asreceived >> ; >> group=1 >> channel=>1-23 >> channel=>73-95 >> ; >> group=2 >> channel=>25-47 >> channel=>49-71 >> >> >> >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > Nitesh Divecha > VoIP/Network Engineer > Viper Networks > 10373 Roselle St. Ste:170 > San Diego, CA. 92121 > > Phone: 858-452-8737 > Fax: 858-452-8638 > Cell: 1-909-964-5181 > vPhone: 544-416-0067 > > Email: nitesh@vipernetworks.com > Web: www.vipernetworks.com > > "Your Internet Phone Company" > A publicly traded Company, OTC: VPER > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users