Asterisk with SPA 3000

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Asterisk with SPA 3000

Postby arunish » Sun Mar 15, 2015 1:34 pm

Hello Sir/Madam,

I am trying to sort out this issue from more than 1 month but all in vain. Please help someone.
I am able to configure Asterisk with SPA 3000 for PSTN to VOIP Calls.

I have some issues while configuring VOIP to PSTN calls. If I use dial plan (S0<:9936152459@gw0>), it sends every calls to 9936152459. It is fine and doing as per the plan but when I use the dial plan (xx.), it should dial the dialed number sent via asterisk but it is not dialing out just getting engaged tone.

Hook State: On Line Voltage: -51 (V)
Hook State: Off Line Voltage: -5 (V)

Line-In-Use Voltage: 30

Let me know if more info required.

My configurations is :-

sip.conf

[general]
port=5060
bindaddr=0.0.0.0




;disallow=all
allow=ulaw

[sipurapstn]
type=friend
secret=visual
host=dynamic
context=from-pstn
insecure=invite
allow=ulaw
port-5061
dtmfmode=inband

[1000]
type=friend
host=dynamic
secret=1000
context=from-pstn


[1001]
type=friend
host=dynamic
secret=1001
context=from-pstn

[1002]
type=friend
host=dynamic
secret=1002
context=from-pstn

[1003]
type=friend
host=dynamic
secret=1003
context=from-pstn


extensions.conf

[from-pstn]
;exten =>1000,1,Dial(SIP/1000)
;exten =>1001,1,Dial(SIP/1001)
exten => 999,1,Answer()
exten => 999,n,Playback(welcome)
exten => 999,n(menu_begning),Background(press-0&press-1&press-2&press-3)
exten => 999,n,WaitExten(5)
exten => 999,n,Hangup()


exten => 0,1,Dial(SIP/1000)
exten => 1,1,Dial(SIP/1001)
exten => 2,1,Dial(SIP/1002)
exten => 3,1,Dial(SIP/1003)

exten => 1000,1,Dial(SIP/1000)
exten => 1001,1,Dial(SIP/1001)
exten => 1002,1,Dial(SIP/1002)
exten => 1003,1,Dial(SIP/1003)
exten => _x.,1,Dial(SIP/${EXTN}@sipurapstn)

CLI logs..

:14:31.723 icess0x9a3c16c Candidate 0 added: comp_id=1, type=host, foundation=Hc0a801fe, addr=192.168.1.254:18366, base=192.168.1.254:18366, prio=0x7effffff (2130706431)
00:14:31.723 icess0x9a3c16c Destroying ICE session 0x9a3c16c
00:14:31.723 stuse0x9a2f428 STUN session 0x9a3b7c4 destroy request, ref_cnt=4
00:14:31.723 stuse0x9afbf10 STUN session 0x9afc32c destroy request, ref_cnt=3
00:14:31.723 ice_session.c ICE session 0x9a3c16c destroyed
00:14:31.723 stun_session.c STUN session 0x9a3b7c4 destroyed
00:14:31.723 stun_session.c STUN session 0x9afc32c destroyed
== Using SIP RTP CoS mark 5
-- Executing [9936152459@from-pstn:1] Dial("SIP/1000-0000000a", "SIP/@sipurapstn") in new stack
00:14:31.756 icess0xb749e2a !ICE session created, comp_cnt=2, role is Unknown agent
00:14:31.761 icess0xb749e2a Candidate 0 added: comp_id=1, type=host, foundation=Hc0a801fe, addr=192.168.1.254:19398, base=192.168.1.254:19398, prio=0x7effffff (2130706431)
00:14:31.761 icess0xb749e2a Destroying ICE session 0xb749e2a4
00:14:31.761 stuse0xb5fa981 STUN session 0xb74ce224 destroy request, ref_cnt=4
00:14:31.761 stuse0xb74ce9e STUN session 0xb74cee04 destroy request, ref_cnt=3
00:14:31.761 ice_session.c ICE session 0xb749e2a4 destroyed
00:14:31.761 stun_session.c STUN session 0xb74ce224 destroyed
00:14:31.761 stun_session.c STUN session 0xb74cee04 destroyed
== Using SIP RTP CoS mark 5
-- Called SIP/@sipurapstn
-- SIP/sipurapstn-0000000b is ringing
-- SIP/sipurapstn-0000000b answered SIP/1000-0000000a
-- Channel SIP/1000-0000000a joined 'simple_bridge' basic-bridge <bdc41d90-96cf-405e-a5ff-7d5e6552e75e>
-- Channel SIP/sipurapstn-0000000b joined 'simple_bridge' basic-bridge <bdc41d90-96cf-405e-a5ff-7d5e6552e75e>
> Bridge bdc41d90-96cf-405e-a5ff-7d5e6552e75e: switching from simple_bridge technology to native_rtp
> 0x9a2cad8 -- Probation passed - setting RTP source address to 192.168.1.8:4010
> 0xb5ecc6a0 -- Probation passed - setting RTP source address to 192.168.1.100:16408
-- Channel SIP/sipurapstn-0000000b left 'native_rtp' basic-bridge <bdc41d90-96cf-405e-a5ff-7d5e6552e75e>
-- Channel SIP/1000-0000000a left 'native_rtp' basic-bridge <bdc41d90-96cf-405e-a5ff-7d5e6552e75e>
== Spawn extension (from-pstn, 9936152459, 1) exited non-zero on 'SIP/1000-0000000a'
arunish
Newsterisk
 
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