Dear All,
I just setup the Asterisk with E100P which it's no problem in Dial In but I
have problem when outdial. The connection method is like this :
E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <->
Asterisk <--> SIP
\-----> Analog PHone
Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect,
Trying, Dialing and then hangup. I've found the log as the following :
*CLI> Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:2332 sip_alloc:
Allocating new SIP call for 95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:5200 check_user_full: Setting
NAT on RTP to 0
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:817 __sip_ack: Stopping
retransmission on '95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101'
of
Response 46613: Found
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:5200 check_user_full: Setting
NAT on RTP to 0
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:6991 handle_request: Check for
res for 2000
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:1633 update_user_counter: Call
from user '2000' is 1 out of 0
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:4423 build_route: build_route:
Contact hop: <sip:2000@192.168.1.101:5060>
Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1255 pbx_extension_helper:
Launching 'ChanIsAvail'
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:6491 zt_request: Using channel
17
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:733 ast_hangup: Hanging up
channel 'Zap/17-1'
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1902 zt_hangup: zt_hangup
(Zap/17-1)
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2417 zt_setoption: Set option
AUDIO MODE, value: ON(1) on Zap/17-1
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1930 zt_hangup: Hangup:
channel: 17 index = 0, normal = 38, callwait = -1, thirdcall = -1
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2329 zt_setoption: Set option
TDD MODE, value: OFF(0) on Zap/17-1
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1151 update_conf: Updated
conferencing on 17, with 0 conference users
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2411 zt_setoption: Set option
AUDIO MODE, value: OFF(0) on Zap/17-1
-- Hungup 'Zap/17-1'
Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1255 pbx_extension_helper:
Launching 'Cut'
Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1255 pbx_extension_helper:
Launching 'Dial'
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:6491 zt_request: Using channel
17
-- Called 17/0085221120000
Urgent handler
Urgent handler
Urgent handler
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set
channel Zap/17-1 to read format ALAW
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set
channel SIP/2000-e12c to write format ALAW
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set
channel Zap/17-1 to write format ALAW
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set
channel SIP/2000-e12c to read form at ALAW
Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:1156 ast_rtp_write: Ooh, format
changed from UNKN to ALAW
Aug 26 15:54:17 DEBUG[-1248367696]: chan_zap.c:1179 zt_enable_ec: No
echocancellation requested
-- Zap/17-1 is ringing
Urgent handler
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1395 ast_indicate: Driver for
channel 'SIP/2000-e12c' does not support indication 3, emulating it
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1510 ast_prod: Prodding
channel 'SIP/2000-e12c'
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set
channel SIP/2000-e12c to write format SLINR
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set
channel SIP/2000-e12c to write format ALAW
Aug 26 15:54:17 DEBUG[-1248367696]: chan_zap.c:1179 zt_enable_ec: No
echocancellation requested
-- Zap/17-1 answered SIP/2000-e12c
Urgent handler
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set
channel SIP/2000-e12c to read format ALAW
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set
channel Zap/17-1 to write format ALAW
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set
channel SIP/2000-e12c to write format ALAW
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set
channel Zap/17-1 to read format ALAW
Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1824 sip_answer: sip_answer
(SIP/2000-e12c)
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:817 __sip_ack: Stopping
retransmission on '95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101'
of
Response 46614: Found
Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:378 ast_rtcp_read: Got RTCP report
of 84 bytes
Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:378 ast_rtcp_read: Got RTCP report
of 118 bytes
-- Channel 0/17, span 1 got hangup
Urgent handler
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:2559 ast_channel_bridge: Bridge
stops because we're zombie or need a soft hangup: c0=SIP/2000-e12c,
c1=Zap/17-
1, flags: No,No,No,Yes
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:2679 ast_channel_bridge: Bridge
stops bridging channels SIP/2000-e12c and Zap/17-1
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:733 ast_hangup: Hanging up
channel 'Zap/17-1'
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1902 zt_hangup: zt_hangup
(Zap/17-1)
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2417 zt_setoption: Set option
AUDIO MODE, value: ON(1) on Zap/17-1
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1930 zt_hangup: Hangup:
channel: 17 index = 0, normal = 38, callwait = -1, thirdcall = -1
Urgent handler
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2066 zt_hangup: Not yet
hungup... Calling hangup once with icause, and clearing call
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2329 zt_setoption: Set option
TDD MODE, value: OFF(0) on Zap/17-1
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1151 update_conf: Updated
conferencing on 17, with 0 conference users
Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2411 zt_setoption: Set option
AUDIO MODE, value: OFF(0) on Zap/17-1
Urgent handler
-- Hungup 'Zap/17-1'
Urgent handler
Aug 26 15:54:17 DEBUG[-1260983376]: app_dial.c:974 dial_exec: Exiting with
DIALSTATUS=ANSWER.
Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1827 ast_pbx_run: Spawn extension
(from-sip,0085221120000,3) exitednon-zero on 'SIP/2000-e12c'
Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:733 ast_hangup: Hanging up
channel 'SIP/2000-e12c'
Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1717 sip_hangup: sip_hangup
(SIP/2000-e12c)
Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1732 sip_hangup:
update_user_counter(2000) - decrement inUse counter
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:817 __sip_ack: Stopping
retransmission on '95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101'
of
Request 102: Found
And I'm not sure what's happening which the call actually didn't
dial out..I
hope someone out there can help me in...
Thanks!
R. Wong
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