[May 12 16:42:32] VERBOSE[21273] pbx.c: -- Executing [+441618bbbbbbb@keybm-incoming:1] Goto("SIP/openser-00043361", "keybm-internal,202,1") in new stack [May 12 16:42:32] VERBOSE[21273] pbx.c: -- Goto (keybm-internal,202,1) [May 12 16:42:32] DEBUG[21273] pbx.c: Launching 'Dial' [May 12 16:42:32] VERBOSE[21273] pbx.c: -- Executing [202@keybm-internal:1] Dial("SIP/openser-00043361", "SIP/keybm-202,15") in new stack [May 12 16:42:32] DEBUG[21273] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [May 12 16:42:32] VERBOSE[21273] netsock.c: == Using SIP RTP CoS mark 5 [May 12 16:42:32] DEBUG[21273] chan_sip.c: Allocating new SIP dialog for 35856d527478be3e76f43cf00b216e2c@yy.yy.yy.yy - INVITE (With RTP) [May 12 16:42:32] DEBUG[21273] chan_sip.c: Setting NAT on RTP to On [May 12 16:42:32] DEBUG[21273] chan_sip.c: OBPROXY: Not applying OBproxy to this call [May 12 16:42:32] DEBUG[21273] acl.c: Found IP address for this socket [May 12 16:42:32] DEBUG[21273] chan_sip.c: Setting SIP_TRANSPORT_UDP with address yy.yy.yy.yy:5060 [May 12 16:42:32] DEBUG[21273] frame.c: Could not find preferred codec - Going for the best codec [May 12 16:42:32] DEBUG[21273] chan_sip.c: *** Our native formats are 0x8 (alaw) [May 12 16:42:32] DEBUG[21273] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [May 12 16:42:32] DEBUG[21273] chan_sip.c: *** Our capabilities are 0x100 (g729) [May 12 16:42:32] DEBUG[20767] audiohook.c: Read factory 0xb348c5b0 was pretty quick last time, waiting for them. [May 12 16:42:32] DEBUG[21273] frame.c: Could not find preferred codec - Going for the best codec [May 12 16:42:32] DEBUG[21273] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [May 12 16:42:32] DEBUG[21273] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [May 12 16:42:32] DEBUG[21273] chan_sip.c: This channel will not be able to handle video. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable DIALEDTIME. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable ANSWEREDTIME. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable DIALEDPEERNAME. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable DIALEDPEERNUMBER. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable DIALSTATUS. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable AGISTATUS. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable MIXMONITOR_FILENAME. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable SIPCALLID. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable SIPDOMAIN. [May 12 16:42:32] DEBUG[21273] channel.c: Not copying variable SIPURI. [May 12 16:42:32] DEBUG[21273] chan_sip.c: Outgoing Call for keybm-202 [May 12 16:42:32] DEBUG[21273] chan_sip.c: Updating call counter for outgoing call [May 12 16:42:32] DEBUG[21273] chan_sip.c: Call to peer 'keybm-202' is 1 out of 5 [May 12 16:42:32] WARNING[21273] chan_sip.c: No audio format found to offer. Cancelling call to keybm-202 [May 12 16:42:32] DEBUG[21273] app_dial.c: ast call on peer returned -1 [May 12 16:42:32] VERBOSE[21273] app_dial.c: -- Couldn't call keybm-202 [May 12 16:42:32] DEBUG[27236] devicestate.c: No provider found, checking channel drivers for SIP - keybm-202 [May 12 16:42:32] DEBUG[21273] channel.c: Hanging up channel 'SIP/keybm-202-00043362' [May 12 16:42:32] DEBUG[27236] chan_sip.c: Checking device state for peer keybm-202 [May 12 16:42:32] DEBUG[21273] chan_sip.c: Hangup call SIP/keybm-202-00043362, SIP callid 0aaa14515bc05cb110c4b27e51b18f84@yy.yy.yy.yy [May 12 16:42:32] DEBUG[27236] devicestate.c: Changing state for SIP/keybm-202 - state 6 (Ringing) [May 12 16:42:32] DEBUG[27236] devicestate.c: device 'SIP/keybm-202' state '6' [May 12 16:42:32] DEBUG[21273] chan_sip.c: update_call_counter(keybm-202) - decrement call limit counter on hangup [May 12 16:42:32] DEBUG[21273] chan_sip.c: Updating call counter for outgoing call [May 12 16:42:32] DEBUG[21273] chan_sip.c: Call to peer 'keybm-202' removed from call limit 5 [May 12 16:42:32] DEBUG[21273] chan_sip.c: Hanging up channel in state Down (not UP) [May 12 16:42:32] DEBUG[27236] devicestate.c: No provider found, checking channel drivers for SIP - keybm-202 [May 12 16:42:32] DEBUG[27236] chan_sip.c: Checking device state for peer keybm-202 [May 12 16:42:32] DEBUG[27236] devicestate.c: Changing state for SIP/keybm-202 - state 1 (Not in use) [May 12 16:42:32] DEBUG[27254] app_queue.c: Device 'SIP/keybm-202' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [May 12 16:42:32] DEBUG[27236] devicestate.c: device 'SIP/keybm-202' state '1' [May 12 16:42:32] DEBUG[20764] audiohook.c: Write factory 0xb34ee040 was pretty quick last time, waiting for them. [May 12 16:42:32] DEBUG[27254] app_queue.c: Device 'SIP/keybm-202' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 12 16:42:32] VERBOSE[21273] chan_sip.c: Scheduling destruction of SIP dialog '0aaa14515bc05cb110c4b27e51b18f84@yy.yy.yy.yy' in 6400 ms (Method: INVITE) [May 12 16:42:32] DEBUG[19943] audiohook.c: Read factory 0xa034ea8 was pretty quick last time, waiting for them. [May 12 16:42:32] VERBOSE[21273] app_dial.c: == Everyone is busy/congested at this time (0:0/0/0) [May 12 16:42:32] DEBUG[21273] rtp.c: Channel '' has no RTP, not doing anything [May 12 16:42:32] DEBUG[21273] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. [May 12 16:42:32] VERBOSE[21273] pbx.c: -- Auto fallthrough, channel 'SIP/openser-00043361' status is 'CHANUNAVAIL' [May 12 16:42:32] DEBUG[27236] devicestate.c: No provider found, checking channel drivers for SIP - keybm-202 [May 12 16:42:32] DEBUG[27236] chan_sip.c: Checking device state for peer keybm-202 [May 12 16:42:32] DEBUG[27236] devicestate.c: Changing state for SIP/keybm-202 - state 1 (Not in use) [May 12 16:42:32] DEBUG[27236] devicestate.c: device 'SIP/keybm-202' state '1' [May 12 16:42:32] DEBUG[27254] app_queue.c: Device 'SIP/keybm-202' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 12 16:42:32] VERBOSE[21273] chan_sip.c: <--- Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060 ---> SIP/2.0 503 Service Unavailable^M Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK5f0f0f11;received=xx.xx.xx.xx;rport=5060^M From: "+441614911754" ;tag=as6970ffcd^M To: ;tag=as1469ba00^M Call-ID: 0998c4726d039d76508d8abd1a271f00@domain.co.uk^M CSeq: 102 INVITE^M Server: Asterisk PBX 1.6.2.7^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M Supported: replaces, timer^M Content-Length: 0^M ^M <------------>