[Problem] Nur ein Gespräch abgehend möglich

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astrakid

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Hi,

ich habe das Problem, dass seit Neuestem nur noch ein Gespräch abgehend möglich ist. Führe ich abgehend ein Gespräch, ist ein eingehendes Gespräch noch möglich.
Führe ich aber parallel ein 2. Gespräch, wird der Aufbau mit einer Fehlermeldung abgebrochen:

WARNING[17052]: chan_sip.c:21124 handle_response_invite: Received response: "Forbidden" from '"0301234567" <sip:[email protected]>;tag=as16992fc7'

ist die Fehleremdlung defintivi vom Provider, bei dem ich registriert bin? Oder kann die auch "intern" auftreten, d.h. von Asterisk?
Wie kann ich den Fehler weiter eingrenzen?

gruß und danke,
astrakid
 
Hi,

da das Thema etwas drängt, habe ich mal dei Debug-Logs analysiert. Ein bisschen bin ich weitergekommen, aber noch nicht am Ziel.
Ich vermute, dass es mit meiner NEtzwerkstruktur zusammenhängt, denn beim Vergleich der Calls, die ich nacheinander abgehend getätigt habe (sich überschneidend, um den Fehler nachzustellen), konnte ich Meldungen von einem meiner Router bzgl. siproxd sehen. Daher kurz meine Architektur:
Am Telefonanschluss hängt ein Speedport W723V (im folgenden DSL-Router genannt), der eigentlich nur als Modem fungieren soll. Da die "Modem-only" Konfiguration aber bei VDSL nicht funktioniert, muss ich den Router so weit wie möglich auf Durchzug stellen (das geht nicht komplett, da viele Ports reserviert sind und nicht weitergeleitet werden dürfen -> Fehlermeldung).
Die Weiterleitung geht auf meinen eigentlichen Router, auf dem Tomato läuft (im Folgenden TOMATO genannt). Daran hängt dann ein iConnect, wo Asterisk drauf läuft (im Folgenden Asterisk genannt).

Die Asterisk Konfiguration sieht folgendermaßen aus:
SIP-Port: 15070
RTP-Ports: 16000-16995

Diese Ports werden vom DSL-Router an TOMATO und von dort an Asterisk durchgegeben. Im Prinzip klappt ja auch alles (eingehend und ausgehend, Sprache etc. alles da).

Nun kann ich im Debug Log aber folgenden Eintrag sehen (nur im Schlechtfall):
Line 235: Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL

Hat einer eine Idee, warum dieses Event generiert wird? Ich hab mal etwas mehr Logshier, vllt. kann jemand etwas erkennen, das wäre echt super...
Code: 
Search "chan.sip" (101 hits in 1 file)
  new  5 (101 hits)
	Line 6:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
	Line 14:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.2.60:5070
	Line 19:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
	Line 20:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Stopping retransmission on '1570266335@192_168_2_60' of Response 2: Match Found
	Line 25:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
	Line 34:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Setting NAT on RTP to On
	Line 35:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
	Line 36:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP o=12 16532 63 IN IP4 79.253.140.1... UNSUPPORTED OR FAILED.
	Line 37:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP s=Mapping... UNSUPPORTED OR FAILED.
	Line 40:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP c=IN IP4 79.253.140.1... OK.
	Line 41:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
	Line 49:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
	Line 50:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
	Line 51:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
	Line 52:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-32/8000... OK.
	Line 53:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 AAL2-G726-32/8000... OK.
	Line 54:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK.
	Line 55:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
	Line 56:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
	Line 66:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: We're settling with these formats: (ulaw|alaw|g722)
	Line 67:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Checking SIP call limits for device 12
	Line 68:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Updating call counter for incoming call
	Line 73:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: *** Our native formats are (g722)
	Line 74:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: *** Joint capabilities are (ulaw|alaw|g722)
	Line 75:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|g722)
	Line 76:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722
	Line 77:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: This channel will not be able to handle video.
	Line 78:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5070>
	Line 79:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: SIP/12-00000083: New call is still down.... Trying...
	Line 80:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.2.60:5070
	Line 91:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Asked to create a SIP channel with formats: (g722)
	Line 92:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Allocating new SIP dialog for [email protected]:15070 - INVITE (No RTP)
	Line 97:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Setting NAT on RTP to On
	Line 99:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.2.1:15070
	Line 100:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: *** Our native formats are (g722)
	Line 101:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: *** Joint capabilities are (g722)
	Line 102:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|g722)
	Line 103:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722
	Line 104:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: *** Our preferred formats from the incoming channel are (g722)
	Line 105:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: This channel will not be able to handle video.
	Line 117:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Outgoing Call for 08003301000
	Line 118:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Updating call counter for outgoing call
	Line 119:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
	Line 120:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: ** Our prefcodec: (g722)
	Line 121:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: -- Done with adding codecs to SDP
	Line 122:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|g722)
	Line 123:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Initializing initreq for method INVITE - callid [email protected]
	Line 124:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 217.0.17.230:5060
	Line 127:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Acked pending invite 102
	Line 128:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
	Line 129:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: SIP response 401 to standard invite
	Line 130:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Trying to put 'ACK sip:080' onto UDP socket destined for 217.0.17.230:5060
	Line 131:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Auth attempt 1 on INVITE
	Line 132:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
	Line 133:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: ** Our prefcodec: (g722)
	Line 134:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: -- Done with adding codecs to SDP
	Line 135:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|g722)
	Line 136:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 217.0.17.230:5060
	Line 141:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
	Line 142:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: SIP response 100 to standard invite
	Line 147:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
	Line 148:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: SIP response 181 to standard invite
	Line 149:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: build_route: Record-Route hop: <sip:[email protected]:56005;lr>
	Line 150:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Got redirecting from number 08003301000
	Line 151:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Got redirecting to number DTMTASP01
	Line 154:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 181' onto UDP socket destined for 192.168.2.60:5070
	Line 157:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
	Line 158:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: SIP response 183 to standard invite
	Line 159:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: build_route: Record-Route hop: <sip:[email protected]:56005;lr>
	Line 160:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
	Line 161:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP o=sems 1668105013 408862860 IN IP4 192.168.1.1... UNSUPPORTED OR FAILED.
	Line 162:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP s=announcement 6ff884ee9a2425d0c5498e1892bfcb4d... UNSUPPORTED OR FAILED.
	Line 165:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.1... OK.
	Line 166:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
	Line 170:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
	Line 171:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
	Line 172:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK.
	Line 178:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: We're settling with these formats: (gsm|alaw)
	Line 179:  Oct 30 19:13:00] DEBUG[31885][C-0000003a] chan_sip.c: We have an owner, now see if we need to change this call
	Line 185:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Setting framing from config on incoming call
	Line 186:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: ** Our capability: (ulaw|alaw|g722) Video flag: True Text flag: True
	Line 187:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: ** Our prefcodec: (nothing)
	Line 188:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: -- Done with adding codecs to SDP
	Line 189:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|g722)
	Line 190:  Oct 30 19:13:00] DEBUG[32176][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 192.168.2.60:5070
	Line 235:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL
	Line 238:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1570266335@192_168_2_60
	Line 240:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 192.168.2.60:5070
	Line 241:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.60:5070
	Line 245:  Oct 30 19:13:07] DEBUG[32176][C-0000003a] chan_sip.c: Hangup call SIP/telekom-00000084, SIP callid [email protected]
	Line 246:  Oct 30 19:13:07] DEBUG[32176][C-0000003a] chan_sip.c: Hanging up channel in state Down (not UP)
	Line 248:  Oct 30 19:13:07] DEBUG[32176][C-0000003a] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
	Line 249:  Oct 30 19:13:07] DEBUG[32176][C-0000003a] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 217.0.17.230:5060
	Line 254:  Oct 30 19:13:07] DEBUG[32176][C-0000003a] chan_sip.c: Hangup call SIP/12-00000083, SIP callid 1570266335@192_168_2_60
	Line 290:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: Acked pending invite 103
	Line 291:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: Stopping retransmission on '[email protected]' of Request 103: Match Found
	Line 296:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
	Line 297:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: Stopping retransmission on '1570266335@192_168_2_60' of Response 3: Match Found
	Line 302:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: Stopping retransmission on '[email protected]' of Request 103: Match Found
	Line 303:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: SIP response 403 to standard invite
	Line 304:  Oct 30 19:13:07] DEBUG[31885][C-0000003a] chan_sip.c: Trying to put 'ACK sip:DTM' onto UDP socket destined for 217.0.17.230:5060
Search "00000039" (327 hits in 1 file)
  new  4 (327 hits)
	Line 29: [Oct 30 19:13:00] DEBUG[31885] logger.c: CALL_ID [C-00000039] created by thread.
	Line 33: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 33: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 34: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
	Line 35: [Oct 30 19:13:00] DEBUG[31885][C-00000039] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces"
	Line 36: [Oct 30 19:13:00] DEBUG[31885][C-00000039] sip/reqresp_parser.c: Found SIP option: -replaces-
	Line 37: [Oct 30 19:13:00] DEBUG[31885][C-00000039] sip/reqresp_parser.c: Matched SIP option: replaces
	Line 38: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.2.60:5070' into...
	Line 39: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.2.60' and port '5070'.
	Line 40: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.2.1' into...
	Line 41: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.2.1' and port ''.
	Line 42: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.2.60:5070
	Line 43: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 43: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 45: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 45: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 46: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
	Line 47: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Stopping retransmission on '296512860@192_168_2_60' of Response 2: Match Found
	Line 48: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 48: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 54: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 54: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 55: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
	Line 56: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.2.60:5070' into...
	Line 57: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.2.60' and port '5070'.
	Line 58: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.2.1' into...
	Line 59: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.2.1' and port ''.
	Line 60: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb4e0eeb4'
	Line 61: [Oct 30 19:13:00] DEBUG[31885][C-00000039] res_rtp_asterisk.c: Allocated port 16810 for RTP instance '0xb4e0eeb4'
	Line 62: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: RTP instance '0xb4e0eeb4' is setup and ready to go
	Line 63: [Oct 30 19:13:00] DEBUG[31885][C-00000039] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb4e0eeb4'
	Line 64: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Setting NAT on RTP to On
	Line 65: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
	Line 66: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP o=11 16530 126 IN IP4 79.253.140.1... UNSUPPORTED OR FAILED.
	Line 67: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP s=Mapping... UNSUPPORTED OR FAILED.
	Line 68: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: Splitting '79.253.140.1' into...
	Line 69: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: ...host '79.253.140.1' and port ''.
	Line 70: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP c=IN IP4 79.253.140.1... OK.
	Line 71: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
	Line 72: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 9 based on m type on 0xb51bfdc8
	Line 73: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 8 based on m type on 0xb51bfdc8
	Line 74: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 0 based on m type on 0xb51bfdc8
	Line 75: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 96 based on m type on 0xb51bfdc8
	Line 76: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 97 based on m type on 0xb51bfdc8
	Line 77: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 2 based on m type on 0xb51bfdc8
	Line 78: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 18 based on m type on 0xb51bfdc8
	Line 79: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 101 based on m type on 0xb51bfdc8
	Line 80: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
	Line 81: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
	Line 82: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
	Line 83: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-32/8000... OK.
	Line 84: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 AAL2-G726-32/8000... OK.
	Line 85: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK.
	Line 86: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
	Line 87: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK.
	Line 88: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
	Line 89: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
	Line 90: [Oct 30 19:13:00] DEBUG[31885][C-00000039] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4e0eeb4'
	Line 91: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 0 from 0xb51bfdc8 to 0xb4e0f060
	Line 92: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 2 from 0xb51bfdc8 to 0xb4e0f060
	Line 93: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 8 from 0xb51bfdc8 to 0xb4e0f060
	Line 94: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 9 from 0xb51bfdc8 to 0xb4e0f060
	Line 95: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 18 from 0xb51bfdc8 to 0xb4e0f060
	Line 96: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 96 from 0xb51bfdc8 to 0xb4e0f060
	Line 97: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 97 from 0xb51bfdc8 to 0xb4e0f060
	Line 98: [Oct 30 19:13:00] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 101 from 0xb51bfdc8 to 0xb4e0f060
	Line 99: [Oct 30 19:13:00] DEBUG[31885][C-00000039] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb4e0eeb4'
	Line 100: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: We're settling with these formats: (ulaw|alaw|g722)
	Line 101: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Checking SIP call limits for device 11
	Line 102: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Updating call counter for incoming call
	Line 103: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.2.1' into...
	Line 104: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.2.1' and port ''.
	Line 105: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.2.1' into...
	Line 106: [Oct 30 19:13:00] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.2.1' and port ''.
	Line 107: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: *** Our native formats are (g722)
	Line 108: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: *** Joint capabilities are (ulaw|alaw|g722)
	Line 109: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|g722)
	Line 110: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722
	Line 111: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: This channel will not be able to handle video.
	Line 112: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5070>
	Line 113: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: SIP/11-00000081: New call is still down.... Trying...
	Line 114: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.2.60:5070
	Line 115: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 115: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 120: [Oct 30 19:13:00] DEBUG[32175][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 120: [Oct 30 19:13:00] DEBUG[32175][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 121: [Oct 30 19:13:00] DEBUG[32175][C-00000039] pbx.c: Result of 'EXTEN' is '08003301028'
	Line 122: [Oct 30 19:13:00] DEBUG[32175][C-00000039] pbx.c: Launching 'Log'
	Line 123: [Oct 30 19:13:00] DEBUG[32175][C-00000039] pbx.c: Launching 'Set'
	Line 124: [Oct 30 19:13:00] DEBUG[32175][C-00000039] pbx.c: Launching 'Set'
	Line 125: [Oct 30 19:13:00] DEBUG[32175][C-00000039] pbx.c: Result of 'EXTEN' is '08003301028'
	Line 126: [Oct 30 19:13:00] DEBUG[32175][C-00000039] pbx.c: Launching 'Dial'
	Line 127: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Asked to create a SIP channel with formats: (g722)
	Line 128: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Allocating new SIP dialog for [email protected]:15070 - INVITE (No RTP)
	Line 129: [Oct 30 19:13:00] DEBUG[32175][C-00000039] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x3bd0b4'
	Line 130: [Oct 30 19:13:00] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Allocated port 16230 for RTP instance '0x3bd0b4'
	Line 131: [Oct 30 19:13:00] DEBUG[32175][C-00000039] rtp_engine.c: RTP instance '0x3bd0b4' is setup and ready to go
	Line 132: [Oct 30 19:13:00] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x3bd0b4'
	Line 133: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Setting NAT on RTP to On
	Line 134: [Oct 30 19:13:00] DEBUG[32175][C-00000039] acl.c: For destination '217.0.17.230', our source address is '192.168.2.1'.
	Line 135: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.2.1:15070
	Line 136: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: *** Our native formats are (g722)
	Line 137: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: *** Joint capabilities are (g722)
	Line 138: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|g722)
	Line 139: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722
	Line 140: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: *** Our preferred formats from the incoming channel are (g722)
	Line 141: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: This channel will not be able to handle video.
	Line 142: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel_internal_api.c: Channel Call ID changing from [C-00000039] to [C-00000039]
	Line 142: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel_internal_api.c: Channel Call ID changing from [C-00000039] to [C-00000039]
	Line 142: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel_internal_api.c: Channel Call ID changing from [C-00000039] to [C-00000039]
	Line 143: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel.c: Not copying variable DIALEDTIME.
	Line 144: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel.c: Not copying variable ANSWEREDTIME.
	Line 145: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel.c: Not copying variable DIALEDPEERNAME.
	Line 146: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel.c: Not copying variable DIALEDPEERNUMBER.
	Line 147: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel.c: Not copying variable DIALSTATUS.
	Line 148: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel.c: Not copying variable SIPCALLID.
	Line 149: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel.c: Not copying variable SIPDOMAIN.
	Line 150: [Oct 30 19:13:00] DEBUG[32175][C-00000039] channel.c: Not copying variable SIPURI.
	Line 151: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Outgoing Call for 08003301028
	Line 152: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Updating call counter for outgoing call
	Line 153: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
	Line 154: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: ** Our prefcodec: (g722)
	Line 155: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: -- Done with adding codecs to SDP
	Line 156: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|g722)
	Line 157: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Initializing initreq for method INVITE - callid [email protected]
	Line 158: [Oct 30 19:13:00] DEBUG[32175][C-00000039] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 217.0.17.230:5060
	Line 160: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 160: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 161: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Acked pending invite 102
	Line 162: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
	Line 163: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: SIP response 401 to standard invite
	Line 164: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Trying to put 'ACK sip:080' onto UDP socket destined for 217.0.17.230:5060
	Line 165: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Auth attempt 1 on INVITE
	Line 166: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
	Line 167: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: ** Our prefcodec: (g722)
	Line 168: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: -- Done with adding codecs to SDP
	Line 169: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|g722)
	Line 170: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 217.0.17.230:5060
	Line 171: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 171: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 173: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 173: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 174: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
	Line 175: [Oct 30 19:13:00] DEBUG[31885][C-00000039] chan_sip.c: SIP response 100 to standard invite
	Line 176: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 176: [Oct 30 19:13:00] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 438: [Oct 30 19:13:02] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 438: [Oct 30 19:13:02] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 439: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
	Line 440: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: SIP response 183 to standard invite
	Line 441: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: build_route: Record-Route hop: <sip:[email protected]:56005;lr>
	Line 442: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
	Line 443: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP o=hiQ9200 2700520120930191303 1814954094 IN IP4 192.168.1.1... UNSUPPORTED OR FAILED.
	Line 444: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP s=Phone Call via hiQ9200 SIPCA... UNSUPPORTED OR FAILED.
	Line 445: [Oct 30 19:13:02] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.1.1' into...
	Line 446: [Oct 30 19:13:02] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.1.1' and port ''.
	Line 447: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.1... OK.
	Line 448: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
	Line 449: [Oct 30 19:13:02] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 8 based on m type on 0xb51c0200
	Line 450: [Oct 30 19:13:02] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 0 based on m type on 0xb51c0200
	Line 451: [Oct 30 19:13:02] DEBUG[31885][C-00000039] rtp_engine.c: Setting payload 101 based on m type on 0xb51c0200
	Line 452: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
	Line 453: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=fmtp:8 vad=no... OK.
	Line 454: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
	Line 455: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=fmtp:0 vad=no... OK.
	Line 456: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
	Line 457: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
	Line 458: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=sqn: 0... UNSUPPORTED OR FAILED.
	Line 459: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
	Line 460: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
	Line 461: [Oct 30 19:13:02] DEBUG[31885][C-00000039] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x3bd0b4'
	Line 462: [Oct 30 19:13:02] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 0 from 0xb51c0200 to 0x3bd260
	Line 463: [Oct 30 19:13:02] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 8 from 0xb51c0200 to 0x3bd260
	Line 464: [Oct 30 19:13:02] DEBUG[31885][C-00000039] rtp_engine.c: Copying payload 101 from 0xb51c0200 to 0x3bd260
	Line 465: [Oct 30 19:13:02] DEBUG[31885][C-00000039] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x3bd0b4'
	Line 466: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: We're settling with these formats: (ulaw|alaw)
	Line 467: [Oct 30 19:13:02] DEBUG[31885][C-00000039] chan_sip.c: We have an owner, now see if we need to change this call
	Line 468: [Oct 30 19:13:02] DEBUG[31885][C-00000039] channel.c: Set channel SIP/telekom-00000082 to read format g722
	Line 469: [Oct 30 19:13:02] DEBUG[31885][C-00000039] channel.c: Set channel SIP/telekom-00000082 to write format g722
	Line 470: [Oct 30 19:13:02] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 470: [Oct 30 19:13:02] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 471: [Oct 30 19:13:02] DEBUG[32175][C-00000039] rtp_engine.c: Setting early bridge SDP of 'SIP/11-00000081' with that of 'SIP/telekom-00000082'
	Line 472: [Oct 30 19:13:02] DEBUG[32175][C-00000039] chan_sip.c: Setting framing from config on incoming call
	Line 473: [Oct 30 19:13:02] DEBUG[32175][C-00000039] chan_sip.c: ** Our capability: (ulaw|alaw|g722) Video flag: True Text flag: True
	Line 474: [Oct 30 19:13:02] DEBUG[32175][C-00000039] chan_sip.c: ** Our prefcodec: (nothing)
	Line 475: [Oct 30 19:13:02] DEBUG[32175][C-00000039] chan_sip.c: -- Done with adding codecs to SDP
	Line 476: [Oct 30 19:13:02] DEBUG[32175][C-00000039] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|g722)
	Line 477: [Oct 30 19:13:02] DEBUG[32175][C-00000039] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 192.168.2.60:5070
	Line 478: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- start learning mode pass with addr = 192.168.2.60:16530
	Line 479: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- probation = 4, seq = 0
	Line 480: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- Condition for learning hasn't exited, so reject the frame.
	Line 481: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- start learning mode pass with addr = 192.168.2.60:16530
	Line 482: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- probation = 3, seq = 1
	Line 483: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- Condition for learning hasn't exited, so reject the frame.
	Line 484: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- start learning mode pass with addr = 192.168.2.60:16530
	Line 485: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- probation = 2, seq = 2
	Line 486: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- Condition for learning hasn't exited, so reject the frame.
	Line 487: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- start learning mode pass with addr = 192.168.2.60:16530
	Line 488: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- probation = 1, seq = 3
	Line 489: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 192.168.2.60:16530
	Line 490: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4e0eeb4'
	Line 491: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw
	Line 492: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160
	Line 493: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x3bd0b4'
	Line 494: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- start learning mode pass with addr = 192.168.2.60:16530
	Line 495: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- probation = 4, seq = 4
	Line 496: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- Condition for learning hasn't exited, so reject the frame.
	Line 497: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- start learning mode pass with addr = 192.168.2.60:16530
	Line 498: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- probation = 3, seq = 5
	Line 499: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- Condition for learning hasn't exited, so reject the frame.
	Line 500: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- start learning mode pass with addr = 192.168.1.1:7072
	Line 501: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- probation = 4, seq = 24421
	Line 502: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- Condition for learning hasn't exited, so reject the frame.
	Line 503: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- start learning mode pass with addr = 192.168.2.60:16530
	Line 504: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- probation = 2, seq = 6
	Line 505: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- Condition for learning hasn't exited, so reject the frame.
	Line 506: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- start learning mode pass with addr = 192.168.1.1:7072
	Line 507: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- probation = 3, seq = 24422
	Line 508: [Oct 30 19:13:02] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- Condition for learning hasn't exited, so reject the frame.
	Line 509: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- start learning mode pass with addr = 192.168.2.60:16530
	Line 510: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- probation = 1, seq = 7
	Line 511: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0xb4e133c0 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 192.168.2.60:16530
	Line 512: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- start learning mode pass with addr = 192.168.1.1:7072
	Line 513: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- probation = 2, seq = 24423
	Line 514: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- Condition for learning hasn't exited, so reject the frame.
	Line 515: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- start learning mode pass with addr = 192.168.1.1:7072
	Line 516: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- probation = 1, seq = 24424
	Line 517: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: 0x44fa38 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 192.168.1.1:7072
	Line 518: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Ooh, format changed from unknown to g722
	Line 519: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Created smoother: format: g722 ms: 20 len: 160
	Line 521: [Oct 30 19:13:03] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 521: [Oct 30 19:13:03] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 522: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: Acked pending invite 103
	Line 523: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: Stopping retransmission on '[email protected]' of Request 103: Match Found
	Line 524: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: SIP response 200 to standard invite
	Line 525: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
	Line 526: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: Call [email protected] responded to our reinvite without changing SDP version; ignoring SDP.
	Line 527: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: Updating call counter for outgoing call
	Line 528: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: build_route: Record-Route hop: <sip:[email protected]:56005;lr>
	Line 529: [Oct 30 19:13:03] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.1.1:56005' into...
	Line 530: [Oct 30 19:13:03] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.1.1' and port '56005'.
	Line 531: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: Trying to put 'ACK sip:DTM' onto UDP socket destined for 217.0.17.230:5060
	Line 532: [Oct 30 19:13:03] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 532: [Oct 30 19:13:03] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 533: [Oct 30 19:13:03] DEBUG[32175][C-00000039] rtp_engine.c: Setting early bridge SDP of 'SIP/11-00000081' with that of 'SIP/telekom-00000082'
	Line 534: [Oct 30 19:13:03] DEBUG[32175][C-00000039] chan_sip.c: SIP answering channel: SIP/11-00000081
	Line 535: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Setting the marker bit due to a source update
	Line 544: [Oct 30 19:13:03] DEBUG[32175][C-00000039] chan_sip.c: Setting framing from config on incoming call
	Line 545: [Oct 30 19:13:03] DEBUG[32175][C-00000039] chan_sip.c: ** Our capability: (ulaw|alaw|g722) Video flag: True Text flag: True
	Line 546: [Oct 30 19:13:03] DEBUG[32175][C-00000039] chan_sip.c: ** Our prefcodec: (nothing)
	Line 547: [Oct 30 19:13:03] DEBUG[32175][C-00000039] chan_sip.c: -- Done with adding codecs to SDP
	Line 548: [Oct 30 19:13:03] DEBUG[32175][C-00000039] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|g722)
	Line 549: [Oct 30 19:13:03] DEBUG[32175][C-00000039] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.60:5070
	Line 550: [Oct 30 19:13:03] DEBUG[32175][C-00000039] features.c: bridge answer set, chan answer set
	Line 551: [Oct 30 19:13:03] DEBUG[32175][C-00000039] features.c: Removing dialed interfaces datastore on SIP/telekom-00000082 since we're bridging
	Line 552: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Setting the marker bit due to a source update
	Line 553: [Oct 30 19:13:03] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Setting the marker bit due to a source update
	Line 554: [Oct 30 19:13:03] DEBUG[32175][C-00000039] rtp_engine.c: rtp-engine-local-bridge: Oooh, formats changed, backing out
	Line 556: [Oct 30 19:13:03] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 556: [Oct 30 19:13:03] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 557: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
	Line 558: [Oct 30 19:13:03] DEBUG[31885][C-00000039] chan_sip.c: Stopping retransmission on '296512860@192_168_2_60' of Response 3: Match Found
	Line 559: [Oct 30 19:13:03] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 559: [Oct 30 19:13:03] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 653: [Oct 30 19:13:08] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 653: [Oct 30 19:13:08] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 654: [Oct 30 19:13:08] DEBUG[31885][C-00000039] chan_sip.c: **** Received BYE (8) - Command in SIP BYE
	Line 655: [Oct 30 19:13:08] DEBUG[31885][C-00000039] netsock2.c: Splitting '192.168.2.60:5070' into...
	Line 656: [Oct 30 19:13:08] DEBUG[31885][C-00000039] netsock2.c: ...host '192.168.2.60' and port '5070'.
	Line 657: [Oct 30 19:13:08] DEBUG[31885][C-00000039] chan_sip.c: Setting SIP_ALREADYGONE on dialog 296512860@192_168_2_60
	Line 658: [Oct 30 19:13:08] DEBUG[31885][C-00000039] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4e0eeb4'
	Line 659: [Oct 30 19:13:08] DEBUG[31885][C-00000039] chan_sip.c: Received bye, issuing owner hangup
	Line 660: [Oct 30 19:13:08] DEBUG[31885][C-00000039] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.2.60:5070
	Line 661: [Oct 30 19:13:08] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 661: [Oct 30 19:13:08] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 662: [Oct 30 19:13:08] DEBUG[32175][C-00000039] channel.c: Didn't get a frame from channel: SIP/11-00000081
	Line 663: [Oct 30 19:13:08] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Setting the marker bit due to a source update
	Line 664: [Oct 30 19:13:08] DEBUG[32175][C-00000039] channel.c: Bridge stops bridging channels SIP/11-00000081 and SIP/telekom-00000082
	Line 665: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CSV_QUOTE(${CDR(start)})' (from 'CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(end)})},v,${CSV_QUOTE(${CDR(src)})},a,${CSV_QUOTE(${CDR(dst)})},ctxt,${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(answer)})},talktime,${CSV_QUOTE(${CDR(billsec,f)})},gdauer,${CSV_QUOTE(${CDR(duration,f)})}
	Line 667: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CDR(start)' (from 'CDR(start)})' len 10)
	Line 668: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '2012-10-30 19:13:00'
	Line 669: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '"2012-10-30 19:13:00"'
	Line 670: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CSV_QUOTE(${CDR(end)})' (from 'CSV_QUOTE(${CDR(end)})},v,${CSV_QUOTE(${CDR(src)})},a,${CSV_QUOTE(${CDR(dst)})},ctxt,${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(answer)})},talktime,${CSV_QUOTE(${CDR(billsec,f)})},gdauer,${CSV_QUOTE(${CDR(duration,f)})}
	Line 672: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CDR(end)' (from 'CDR(end)})' len 8)
	Line 673: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '2012-10-30 19:13:08'
	Line 674: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '"2012-10-30 19:13:08"'
	Line 675: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CSV_QUOTE(${CDR(src)})' (from 'CSV_QUOTE(${CDR(src)})},a,${CSV_QUOTE(${CDR(dst)})},ctxt,${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(answer)})},talktime,${CSV_QUOTE(${CDR(billsec,f)})},gdauer,${CSV_QUOTE(${CDR(duration,f)})}
	Line 677: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CDR(src)' (from 'CDR(src)})' len 8)
	Line 678: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '03474472888'
	Line 679: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '"03474472888"'
	Line 680: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dst)})' (from 'CSV_QUOTE(${CDR(dst)})},ctxt,${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(answer)})},talktime,${CSV_QUOTE(${CDR(billsec,f)})},gdauer,${CSV_QUOTE(${CDR(duration,f)})}
	Line 682: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CDR(dst)' (from 'CDR(dst)})' len 8)
	Line 683: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '08003301028'
	Line 684: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '"08003301028"'
	Line 685: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CSV_QUOTE(${CDR(dcontext)})' (from 'CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(answer)})},talktime,${CSV_QUOTE(${CDR(billsec,f)})},gdauer,${CSV_QUOTE(${CDR(duration,f)})}
	Line 687: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CDR(dcontext)' (from 'CDR(dcontext)})' len 13)
	Line 688: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is 'eltern'
	Line 689: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '"eltern"'
	Line 690: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CSV_QUOTE(${CDR(answer)})' (from 'CSV_QUOTE(${CDR(answer)})},talktime,${CSV_QUOTE(${CDR(billsec,f)})},gdauer,${CSV_QUOTE(${CDR(duration,f)})}
	Line 692: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CDR(answer)' (from 'CDR(answer)})' len 11)
	Line 693: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '2012-10-30 19:13:03'
	Line 694: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '"2012-10-30 19:13:03"'
	Line 695: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CSV_QUOTE(${CDR(billsec,f)})' (from 'CSV_QUOTE(${CDR(billsec,f)})},gdauer,${CSV_QUOTE(${CDR(duration,f)})}
	Line 697: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CDR(billsec,f)' (from 'CDR(billsec,f)})' len 14)
	Line 698: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '4.728200'
	Line 699: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '"4.728200"'
	Line 700: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CSV_QUOTE(${CDR(duration,f)})' (from 'CSV_QUOTE(${CDR(duration,f)})}
	Line 702: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Evaluating 'CDR(duration,f)' (from 'CDR(duration,f)})' len 15)
	Line 703: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '8.177052'
	Line 704: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Function result is '"8.177052"'
	Line 705: [Oct 30 19:13:08] DEBUG[32175][C-00000039] channel.c: Hanging up channel 'SIP/telekom-00000082'
	Line 706: [Oct 30 19:13:08] DEBUG[32175][C-00000039] chan_sip.c: Hangup call SIP/telekom-00000082, SIP callid [email protected]
	Line 707: [Oct 30 19:13:08] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x3bd0b4'
	Line 708: [Oct 30 19:13:08] DEBUG[32175][C-00000039] netsock2.c: Splitting '192.168.1.1:56005' into...
	Line 709: [Oct 30 19:13:08] DEBUG[32175][C-00000039] netsock2.c: ...host '192.168.1.1' and port '56005'.
	Line 710: [Oct 30 19:13:08] DEBUG[32175][C-00000039] chan_sip.c: Trying to put 'BYE sip:DTM' onto UDP socket destined for 217.0.17.230:5060
	Line 715: [Oct 30 19:13:08] DEBUG[32175][C-00000039] app_dial.c: Exiting with DIALSTATUS=ANSWER.
	Line 716: [Oct 30 19:13:08] DEBUG[32175][C-00000039] pbx.c: Spawn extension (eltern,08003301028,4) exited non-zero on 'SIP/11-00000081'
	Line 717: [Oct 30 19:13:08] DEBUG[32175][C-00000039] channel.c: Soft-Hanging up channel 'SIP/11-00000081'
	Line 718: [Oct 30 19:13:08] DEBUG[32175][C-00000039] channel.c: Hanging up channel 'SIP/11-00000081'
	Line 719: [Oct 30 19:13:08] DEBUG[32175][C-00000039] chan_sip.c: Hangup call SIP/11-00000081, SIP callid 296512860@192_168_2_60
	Line 720: [Oct 30 19:13:08] DEBUG[32175][C-00000039] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4e0eeb4'
	Line 726: [Oct 30 19:13:08] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 726: [Oct 30 19:13:08] DEBUG[31885][C-00000039] logger.c: CALL_ID [C-00000039] bound to thread.
	Line 727: [Oct 30 19:13:08] DEBUG[31885][C-00000039] chan_sip.c: Stopping retransmission on '[email protected]' of Request 104: Match Found
	Line 728: [Oct 30 19:13:08] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.
	Line 728: [Oct 30 19:13:08] DEBUG[31885][C-00000039] logger.c: Call_ID [C-00000039] being removed from thread.

Zur Ergänzung:192.168.2.60 ist die Telefonstation. Dort habe ich 2 Geräte mit eigenen Anmeldeeinstellungen eingetragen, so dass die Handgeräte separat ansteuerbar sind. Dazu habe ich aber auch schon mit meinem Smartphone und SIP-Client den Fehler nachgestellt, um hier eine Fehlkonfiguration auszuschließen...

Ich bin echt am Verzweifeln...

gruß,
astrakid
 
So, jetzt habe ich das ganze mal mit Sipgate getestet - und da sind zwei parallele ausgehende Gespräche kein Problem... Ist das dann doch ein Problem bei meinem Provider?
Hier mal die sip.conf für den ausgehenden Teil:

sipgate:
[3747257]
type=peer
defaultuser=3747257
callerid=3747257
fromuser=3747257 ;WIRD VON SIPGATE BENOETIGT!!!
secret=passwort
host=sipgate.de
fromdomain=sipgate.de
insecure=port,invite
nat=force_rport
directmedia=no

[telekom]
type=peer
nat=force_rport
;nat=yes
[email protected]
[email protected]
md5secret=7f92f1111fa46bf9baa13452a1111111
host=tel.t-online.de
fromuser=03474472888
fromdomain=tel.t-online.de
insecure=port,invite
canreinvite=no
directmedia=no ;testweise benutzt, hat aber auch nichts gebracht


kann ich dann wohl doch ein Ticket bei der Telekom aufmachen (ist schon offen, aber dann kann ich da noch etwas Druck ausüben)?

gruß,
astrakid
 
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