Hi,
I have a problem with polarity reverse on answer
I use asterisk 1.4.30 linux kernel version 2.6.27 dahdi version 2.2.1 and
analog card is Sangoma a400 with fxo ports
this is my config
[trunkgroups]
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
answeronpolarityswitch=yes
and then i call from sip to mobile over gsm gw (nokia 32) which have a polarity
reverse i pick up the mobile phone. in sip phone i hear that polarity revers
was but at the asterisk shows
Exiting with DIALSTATUS=CHANUNAVAIL
core show channels verbose
Channel Context Extension Prio State
Application Data CallerID Duration
Accountcode BridgedTo
DAHDI/11-1 from-zaptel 8685XXXX 1 Dialing AppDial
(Outgoing Line) 8685XXXXXX 00:00:29
(None)
SIP/293-00000003 splius 8685XXXX 1 Ring Dial
dahdi/11/8685XXXXX 293 00:00:29
(None)
Show not bridged but conversation was normal poth sides everything hear
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2894 do_setnat: Setting NAT on RTP
to Off
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2899 do_setnat: Setting NAT on VRTP
to Off
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2904 do_setnat: Setting NAT on UDPTL
to Off
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:4797 sip_alloc: Allocating new SIP
dialog for 842ada2a-77b3...@192.168.xx.xx - INVITE (With RTP)
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2894 do_setnat: Setting NAT on RTP
to On
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2899 do_setnat: Setting NAT on VRTP
to On
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2904 do_setnat: Setting NAT on UDPTL
to On
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2271 __sip_ack: Stopping
retransmission on '842ada2a-77b3...@192.168.xx.xx' of Response 101: Match Found
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2894 do_setnat: Setting NAT on RTP
to On
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2899 do_setnat: Setting NAT on VRTP
to On
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2904 do_setnat: Setting NAT on UDPTL
to On
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:15115 handle_request_invite:
Checking SIP call limits for device 293
[Mar 26 14:36:38] DEBUG[12577]: pbx.c:1859 pbx_extension_helper: Launching
'Dial'
-- Executing [8685xx...@splius:1] Dial("SIP/293-00000002",
"dahdi/11/8685XXXXX") in new stack
[Mar 26 14:36:38] DEBUG[12577]: chan_dahdi.c:8450 dahdi_request: Using channel
11
[Mar 26 14:36:38] DEBUG[12577]: rtp.c:1650 ast_rtp_make_compatible: Channel
'DAHDI/11-1' has no RTP, not doing anything
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable DIALEDTIME.
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable ANSWEREDTIME.
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable DIALEDPEERNAME.
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable DIALEDPEERNUMBER.
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable DIALSTATUS.
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable SIPCALLID.
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable SIPUSERAGENT.
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable SIPDOMAIN.
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables:
Not copying variable SIPURI.
[Mar 26 14:36:38] DEBUG[12577]: chan_dahdi.c:2301 dahdi_call: Dialing
'8685XXXXX'
[Mar 26 14:36:38] DEBUG[12577]: chan_dahdi.c:2379 dahdi_call: Deferring
dialing... (res -1)
[Mar 26 14:36:38] DEBUG[11975]: channel.c:1133 channel_find_locked: Avoiding
initial deadlock for channel '0x24a3c60'
-- Called 11/8685XXXXX
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3157 set_format: Set channel
DAHDI/11-1 to read format alaw
[Mar 26 14:36:38] DEBUG[12577]: channel.c:3157 set_format: Set channel
SIP/293-00000002 to read format ulaw
[Mar 26 14:36:39] DEBUG[12577]: chan_dahdi.c:5048 __dahdi_exception: Exception
on 25, channel 11
[Mar 26 14:36:39] DEBUG[12577]: chan_dahdi.c:4142 dahdi_handle_event: Got event
Hook Transition Complete(12) on channel 11 (index 0)
[Mar 26 14:36:39] DEBUG[12577]: chan_dahdi.c:4883 dahdi_handle_event: Sent
deferred digit string: T8685XXXXXw
[Mar 26 14:36:41] DEBUG[12577]: chan_dahdi.c:5048 __dahdi_exception: Exception
on 25, channel 11
[Mar 26 14:36:41] DEBUG[12577]: chan_dahdi.c:4142 dahdi_handle_event: Got event
Dial Complete(9) on channel 11 (index 0)
[Mar 26 14:36:41] DEBUG[12577]: chan_dahdi.c:1784 dahdi_enable_ec: Enabled echo
cancellation on channel 11
[Mar 26 14:36:41] DEBUG[12577]: chan_sip.c:7241 transmit_response_with_sdp:
Setting framing from config on incoming call
[Mar 26 14:36:41] DEBUG[12577]: rtp.c:2901 ast_rtp_write: Ooh, format changed
from unknown to alaw
[Mar 26 14:36:41] DEBUG[12577]: rtp.c:2918 ast_rtp_write: Created smoother:
format: 8 ms: 20 len: 160
[Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:5048 __dahdi_exception: Exception
on 25, channel 11
[Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:4142 dahdi_handle_event: Got event
On hook(1) on channel 11 (index 0)
[Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:1816 dahdi_disable_ec: disabled
echo cancellation on channel 11
[Mar 26 14:36:57] DEBUG[12577]: channel.c:1585 ast_hangup: Hanging up channel
'DAHDI/11-1'
[Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:2816 dahdi_hangup:
dahdi_hangup(DAHDI/11-1)
[Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:2851 dahdi_hangup: Hangup:
channel: 11 index = 0, normal = 25, callwait = -1, thirdcall = -1
[Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:1816 dahdi_disable_ec: disabled
echo cancellation on channel 11
[Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:3290 dahdi_setoption: Set option
TDD MODE, value: OFF(0) on DAHDI/11-1
[Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:1752 update_conf: Updated
conferencing on 11, with 0 conference users
-- Hungup 'DAHDI/11-1'
== Everyone is busy/congested at this time (1:0/0/1)
[Mar 26 14:36:57] DEBUG[12577]: rtp.c:1576 ast_rtp_early_bridge: Channel
'<unspecified>' has no RTP, not doing anything
[Mar 26 14:36:57] DEBUG[12577]: app_dial.c:1901 dial_exec_full: Exiting with
DIALSTATUS=CHANUNAVAIL.
== Auto fallthrough, channel 'SIP/293-00000002' status is 'CHANUNAVAIL'
[Mar 26 14:36:57] DEBUG[12577]: channel.c:1482 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/293-00000002'
[Mar 26 14:36:57] DEBUG[12577]: channel.c:1482 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/293-00000002'
[Mar 26 14:36:57] DEBUG[12577]: channel.c:1585 ast_hangup: Hanging up channel
'SIP/293-00000002'
[Mar 26 14:36:57] DEBUG[12577]: chan_sip.c:3709 sip_hangup: Hangup call
SIP/293-00000002, SIP callid 842ada2a-77b3...@192.168.xx.xx)
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:2271 __sip_ack: Stopping
retransmission on '842ada2a-77b3...@192.168.33.12' of Response 102: Match Found
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11881 sip_dump_history:
---------- SIP HISTORY for '842ada2a-77b3...@192.168.xx.xx'
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11885 sip_dump_history: * SIP Call
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 001. Rx
INVITE / 101 INVITE / sip:8685xx...@192.168.xx.xx
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 002.
AuthChal Auth challenge sent for - nc 0
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 003.
TxRespRel SIP/2.0 / 101 INVITE - 407 Proxy Authentication Required
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 004.
SchedDestroy 32000 ms
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 005. Rx
ACK / 101 ACK / sip:8685xx...@192.168.xx.xx
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 006. Rx
INVITE / 102 INVITE / sip:8685xx...@192.168.xx.xx
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 007.
CancelDestroy
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 008.
Invite New call: 842ada2a-77b3...@192.168.xx.xx
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 009.
AuthOK Auth challenge succesful for 293
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 010.
NewChan Channel SIP/293-00000002 - from 842ada2a-77b3...@192.168.xx.xx
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 011.
TxResp SIP/2.0 / 102 INVITE - 100 Trying
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 012.
TxResp SIP/2.0 / 102 INVITE - 183 Session Progress
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 013.
TxRespRel SIP/2.0 / 102 INVITE - 503 Service Unavailable
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 014.
Hangup Cause Unknown
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 015. Rx
ACK / 102 ACK / sip:8685xxx...@192.168.xx.xx
[Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11891 sip_dump_history:
--
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