I have set up asterisk 1.8.13.1 on a Raspberry Pi. I also have a Sipura SPA3000. I am able to make calls from an X-Lite soft phone (extension 3001) to a phone connected to the SPA3000 (extension 3000).
I am now trying to use the X-Lite to call out to the world through the PSTN line connected to the FXO interface on the SPA3000. sip.conf is:
[code][SIPURA]
type=friend
port=5061
host=192.168.1.213
context=from-trunk-sip-SIPURA
[3001]
deny=0.0.0.0/0.0.0.0
secret=xxxxx
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
dial=SIP/3001
permit=0.0.0.0/0.0.0.0
callerid=device <3001>
;Extension 3000 is essentially the same as 3001[/code]
extensions.conf is:
[code][globals]
SIPURA=SIPURA
[from-internal]
exten => 3000,1,Dial(SIP/3000)
exten => 3001,1,Dial(SIP/3001)
exten => _NXXXXXX,1,Answer()
same => n,Dial(SIP/${SIPURA}/${EXTEN})[/code]
When I dialed 9876543 from the X-Lite (192.168.1.5), I hear a dial tone which sounds like the dial tone from the telco. However, it is not followed by the dialing of the digits. The asterisk output is as follows:
== Using SIP RTP CoS mark 5
-- Executing [9876543@from-internal:1] Answer("SIP/3001-00000002", "") in new stack
-- Executing [9876543@from-internal:2] Dial("SIP/3001-00000002", "SIP/SIPURA/9876543") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/SIPURA/9876543
-- SIP/SIPURA-00000003 is ringing
-- SIP/SIPURA-00000003 answered SIP/3001-00000002
-- Locally bridging SIP/3001-00000002 and SIP/SIPURA-00000003
[Sep 30 17:56:58] NOTICE[17449]: res_rtp_asterisk.c:2339 ast_rtp_read: Unknown RTP codec 126 received from '192.168.1.5:62980'
== Spawn extension (from-internal, 9876543, 2) exited non-zero on 'SIP/3001-00000002'
What can be the problem? Was the number dialed before the SIPURA answered? When I pressed the keys on the X-Lite while hearing the dial tone, the dial tone wasn’t cut like what would happen when you dial into an analog line.
Thanks in advance.