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: -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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