[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: *** Our native formats are (ulaw)
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: *** Joint capabilities are (ulaw|alaw)
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: *** Our capabilities are (ulaw|alaw|speex|speex16|g722)
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: This channel will not be able to handle video.
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: SIP/telefon-0000000d: New call is still down.... Trying...
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.10.10.22:5060
[Dec 30 16:36:11] DEBUG[23802] devicestate.c: No provider found, checking channel drivers for SIP - telefon
[Dec 30 16:36:11] DEBUG[23802] chan_sip.c: Checking device state for peer telefon
[Dec 30 16:36:11] DEBUG[23802] devicestate.c: Changing state for SIP/telefon - state 1 (Not in use)
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] pbx_variables.c: Result of 'CALLERIDNUM' is NULL
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] pbx.c: Launching 'Verbose'
[Dec 30 16:36:11] VERBOSE[14595][C-0000005f] app_verbose.c: Receiving SMS from
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] pbx.c: Launching 'Answer'
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] chan_sip.c: SIP answering channel: SIP/telefon-0000000d
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Dec 30 16:36:11] DEBUG[23802] devicestate.c: No provider found, checking channel drivers for SIP - telefon
[Dec 30 16:36:11] DEBUG[23802] chan_sip.c: Checking device state for peer telefon
[Dec 30 16:36:11] DEBUG[23802] devicestate.c: Changing state for SIP/telefon - state 1 (Not in use)
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] chan_sip.c: ** Our prefcodec: (nothing)
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] chan_sip.c: -- Done with adding codecs to SDP
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] chan_sip.c: Setting framing on incoming call: 20
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw)
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.22:5060
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] res_rtp_asterisk.c: 0x7fe7ec0049c0 -- Probation learning mode pass with source address 10.10.10.22:5006
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] pbx.c: Launching 'Monitor'
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] pbx.c: Launching 'SMS'
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] channel.c: Channel SIP/telefon-0000000d setting write format path: slin -> ulaw
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] channel.c: Channel SIP/telefon-0000000d setting read format path: ulaw -> slin
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Dec 30 16:36:11] DEBUG[23816] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #98))
[Dec 30 16:36:11] DEBUG[23816] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.22:5060
[Dec 30 16:36:11] DEBUG[23816] chan_sip.c: = Looking for  Call ID: 4264793084@10_10_10_22 (Checking From) --From tag 3386058873 --To-tag as2584912b
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: Stopping retransmission on '4264793084@10_10_10_22' of Response 3: Match Found
[Dec 30 16:36:11] DEBUG[14595][C-0000005f] res_rtp_asterisk.c: Ooh, format changed from none to ulaw
[Dec 30 16:36:11] DEBUG[23816] chan_sip.c: = Looking for  Call ID: 4264793084@10_10_10_22 (Checking From) --From tag 3386058873 --To-tag as2584912b
[Dec 30 16:36:11] DEBUG[23816][C-0000005f] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Dec 30 16:36:12] NOTICE[14595][C-0000005f] app_sms.c: Received to /var/spool/asterisk/sms/morx/sms-gigaset.2016-12-30T16:36:12-0
[Dec 30 16:36:12] WARNING[14595][C-0000005f] app_sms.c: Unknown message type 02
[Dec 30 16:36:12] DEBUG[23816] chan_sip.c: Allocating new SIP dialog for 6ddc96924a6e465c660a95850fcca939@[::1]:5060 - OPTIONS (No RTP)
[Dec 30 16:36:12] DEBUG[23816] acl.c: For destination '40.85.102.118', our source address is '83.208.99.107'.
[Dec 30 16:36:12] DEBUG[23816] chan_sip.c: Setting AST_TRANSPORT_UDP with address 83.208.99.107:5060
[Dec 30 16:36:12] DEBUG[23816] chan_sip.c: SIP call-id changed from '6ddc96924a6e465c660a95850fcca939@[::1]:5060' to '1ec3037471ac917e6c28f9586e1493c8@83.208.99.107:5060'
[Dec 30 16:36:12] DEBUG[23816] chan_sip.c: Initializing initreq for method OPTIONS - callid 1ec3037471ac917e6c28f9586e1493c8@83.208.99.107:5060
[Dec 30 16:36:12] DEBUG[23816] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 40.85.102.118:5060
[Dec 30 16:36:13] NOTICE[14595][C-0000005f] app_sms.c: channel hangup
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] pbx.c: Launching 'Hangup'
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] channel.c: Soft-Hanging (0x20) up channel 'SIP/telefon-0000000d'
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] pbx.c: Spawn extension (outgoing,*880,5) exited non-zero on 'SIP/telefon-0000000d'
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] channel.c: Soft-Hanging (0x10) up channel 'SIP/telefon-0000000d'
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] channel.c: Hanging up channel 'SIP/telefon-0000000d'
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] chan_sip.c: Hangup call SIP/telefon-0000000d, SIP callid 4264793084@10_10_10_22
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fe7ec004670'
[Dec 30 16:36:13] DEBUG[14595][C-0000005f] chan_sip.c: Trying to put 'BYE sip:tel' onto UDP socket destined for 10.10.10.22:5060