configuration - Call failed using Asterisk -



configuration - Call failed using Asterisk -

i trying on 2 pcs, first acts sip server & client (has asterisk & twinkle installed) , other client only(has twinkle installed) . seek create phone call between them using ethernet cable -no internet- established wired connection , gave each of them address , gave 1st 1 asterisk installed ip 192.168.0.1 & 2nd 1 192.168.0.2. fist 1 has username 100 & 2nd 1 101.

i wrote in terminal "sudo asterisk -rvvvvvvv" "sip reload" "dialplan reload" "sip set debug on" & found wired connection find 2 twinkles on 2 pcs registered asterisk when seek create phone call between them , twinkle said " phone call failed 404 not found " think problem in extensions.conf can not figure out wrong, 1 can help me .? here total output on terminal mediafire.com/?6g0uuhkai5vcahk

also tell me if there wrong in file beginner.

thanks in advance here configuration files: sip.conf

[general]

bindport=5060

udpbindaddr=192.168.0.1:5060

allowguest=yes

disallow=all

allow=gsm

delayreject=yes

nochecksums=no

pedantic=no

srvlookup=yes

autodomain=yes

sipdebug = yes

domain=192.168.0.1

nat=no

notifyringing=yes

notifyhold=yes

register => 100:sarasara@192.168.0.1/internal-phones

register => 101:saadsaad@192.168.0.1/internal-phones

peer auth=100:sarasara@192.168.0.1

peer auth=101:saadsaad@192.168.0.1

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[192.168.0.1]

usereqphone = yes

nat=no

fromdomain=192.168.0.1

fromuser=100

secret=sarasara

username=100

context=internal-phones

authname=100

dtmfmode = rfc2833

canreinvite=yes

notifyringing=yes

notifyhold=yes

peer auth=100:sarasara@192.168.0.1

peer auth=101:saadsaad@192.168.0.1

disallow=all

allow=gsm

[100]

type=friend

context=internal-phones

secret=sarasara

nat=no

qualify=no

host=dynamic

dtmfmode = rfc2833

permit=192.168.0.1

[101]

type=friend

context=internal-phones

secret=saadsaad

qualify=no

host=dynamic

nat=no

dtmfmode = rfc2833

permit=192.168.0.1

extensions.conf

[globals]

[general]

exten => 100,1,dial(sip/100,60)

exten => 101,1,dial(sip/101,60)

exten => s,1,hangup

[internal-phones] exten => 100,1,dial(sip/100,60)

exten => 101,1,dial(sip/101,60)

exten => s,1,hangup

enable sip debugging (sip set debug on) view sip response call. (503, 403 or 404 ?). also, create sure soft-phone listening on udp port other 5060, conflict asterisk.

configuration sip asterisk ethernet sip-server

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