Incoming Skype Calls gets disconnected
I setup a Skype gateway using the howto on this site. Outbound Skype calls work without any issue but I've a problem with inbound skype calls. It gets disconnected after around 10 secs. It doesn't matter whichever destination i choose in the Inbound Route
IVR,Local Extension,Phonebook - the result is always the same.
This is what i came across while debugging the call
[2009-07-16 22:59:00] WARNING[2943] chan_sip.c: Maximum retries exceeded on transmission 914870468026@127.0.0.1 for seqno 1 (Critical Response) [2009-07-16 22:59:00] WARNING[2943] chan_sip.c: Hanging up call 914870468026@127.0.0.1 - no reply to our critical packet.
Any idea why?
Here are my trunk settings,siptosip.cfg file,stack trace. I've CHECKED the SIGNAL RINGING option in Inbound Routes.
Trunk Settings
---------------------------------------------------------------
disallow=all
allow=ulaw&alaw&ilbc&speex
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
host=127.0.0.1
incominglimit=1
nat=never
port=5070
qualify=yes
secret=goodpasswd
type=peer
username=Skype_Caller
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.255
insecure=invite,port < - Without this line,calls never came insiptosis.cfg
# SipToSis configuration file
# ___________________________________________
#
#how opften in minutes to check for configuration changes (0=disable)
configWatchInterval=0
#Set to log_debug.properties for full debugging
logConfigFile=log.properties
#Files containing Authorization rules
siptoskypeauthfile=SipToSkypeAuth.props
skypetosipauthfile=SkypeToSipAuth.props
#Files containing dialing transforms
SkypeOutDialingRulesFile=SkypeOutDialingRules.props
SipOutDialingRulesFile=SipOutDialingRules.props
#location of ua.jar file
ua_jar=ua.jar
#increase audio threads processing priority (0-2) 0=normal
audioPriorityIncrease=0
#jitter enable and size (0=disable,1=small,2=medium,3=large,4=extra large)
jitterLevel=2
#Set to skype_connect=no to disable connection to skype client - for testing purposes
skype_connect=yes
#Following ports are used by skype to transfer audio to/from siptosis
# - use any unused ports - uses 2 ports per connection
skype_audioportbase=64432
#Set to yes to enable skype DTMF support - uses more cpu
enableSkypeDtmfDetector=yes
SkypeDtmfDetectorHitThreshold=90
SkypeDtmfDetectorSilenceThreshold=6
#Set to yes to regenerate SIP DTMF to Skype
sendSipDtmfToSkype=yes
#Set to yes to regenerate Skype DTMF to Sip
sendSkypeDtmfToSip=yes
#If using inband detectors, no to detect dtmf only during authentication (saves cpu)
inbandFullTimeDtmfDetection=yes
#special mode if using skype client manually and an outbound skype call is made, it will attempt a sip call and link the two
JoinManualSkypeOutboundCallToSip=no
#refuse,voicemail,ignore,transferto:skypeid
# If you are using a PBX with multiple clients/ids you probably want to use ignore
# or possibly transferto:nextskypeid to the next skypeclient in the chain
# then refuse on the last one
SkypeInboundAllChannelsBusyAction=refuse
#If an incoming skype call and sip destination is not available for any reason, what to do with skype call.
#Allowed options: ring or refuse - ring allows the skype client to continue ringing and be answered manually.
SkypeInboundSipDestUnavailableAction=refuse
#busy,transferto:sipurl
# If you want multiple outbound channels (AsteriskWin32 does not like this)
# use transferto:sipurl to the next SipToSis channel in the chain then busy on the last one
SipInboundAllChannelsBusyAction=busy
#enable if skype client can support multiple active calls at same time - trying to lie here won't work
#I have yet to find a client that can do this
skypeclientsupportsmulticalls=no
#For an ATA/SIP Phone set to 2 - this allows two total calls - one will be on hold.
#For a PBX - depending on if the skype client supports multi calls - if not set it to 1
# otherwise set based on your hardware/bandwidth limitations
concurrentcalllimit=1
#specify in 5 minute increments, 0=disable auto shutdown - siptosis will auto shutdown in x minutes when idle
autoShutdownMinutes=0
#Seconds caller has to enter the pin number
pintimeout=8
#Number of pin entry attempts before auto hangup
pinretrylimit=3
#Seconds caller has to enter the destination number
destinationtimeout=12
#Number of destination entry attempts before auto hangup
destinationretrylimit=3
#SIP authorization system recordings - make your own if you like (wav 16k 16bit mono).
pinFile=clips/enterPin.wav
destinationFile=clips/enterDest.wav
dialingFile=clips/dialing.wav
invalidPinFile=clips/invalidPin.wav
invalidDestFile=clips/invalidDest.wav
#Skype authorization system recordings - make your own if you like (wav 16k 16bit mono).
skypePinFile=clips/enterPin.wav
skypeDestinationFile=clips/enterDest.wav
skypeDialingFile=clips/dialing.wav
skypeInvalidPinFile=clips/invalidPin.wav
skypeInvalidDestFile=clips/invalidDest.wav
#Used for Skypeout only - transmit skype feedback sound during PSTN call attempt
handleEarlyMedia=yes
#Send Skype IM when calling skype users - not used for skypeout
sendSkypeIM=no
skypeimmessage=You are about to receive a Skype Voice call from [callerid].
#delay between the IM and the actual skype call in seconds.
sendSkypeImDelay=2
#transport_protocols=udp tcp
transport_protocols=udp
#via_addr=127.0.0.7
#outbound_proxy=127.0.0.2:5060
#Sample AUTO config with NO registration
# username and password not important in this mode
#Set to available port to transport SIP messages on siptosis computer
#host_port=5070
#username=skypests
#username=Skype_Caller
#passwd=goodpasswd
# --- end of NO registration example ---
#Sample config with NO registration - use if above auto config doesn't work - change 127.0.0.1 to ip address of computer running siptosis
# username and password not important in this mode
#Set to available port to transport SIP messages on siptosis computer
host_port=5070
#contact_url=sip:skypests@127.0.0.1:5070
from_url="SkypeCaller" <sip:Skype_Caller@127.0.0.1:5060>
username=Skype_Caller
passwd=goodpasswd
#realm=127.0.0.1
# --- end of NO registration example ---
#do_unregister=yes
#do_unregister_all=yes
#keepalive_time - set to zero to disable keep alives
keepalive_time=45000
audio=yes
#following is the SIP RTP port base - use an even port number
audio_port=63200
#auto hangup after no rtp packets received for ? seconds
noRtpReceivedAutoHangupSeconds=30
#only PCMU,PCMA,GSM (jmf lib),GSMTRI (tritonus libs) codecs currently supported - first one is the preferred codec
#You can append RW to the codec name to disable sample averaging - use need to use filter with those.
#Speex doesn't not work well at all - mass cpu usage
audio_codec=PCMU,PCMA,GSMTRI,ILBC,SPEEX
#PCMU/PCMA allow 160,240,320 - GSM allows 160 - ILBC allows 240, Speex allows 160, Speex16k allows 320 - need one size for each codec specified
audio_frame_size=240,240,160,240,160
#if using dynamic payload types (speex and ilbc) you must specify the payload number (asterisk uses 98,97), if not you can remove this parameter
audio_avp=-1,-1,-1,98,97
#Audio volume gains - 1 for each codec - (decimal number) 1=flat no gain, higher=louder, too high will clip or distort
#volume sip->skype
skype_audiooutgain=1,1,1,1,1
#volume skype->sip
skype_audioingain=1.5,1.5,1.5,1.5,1.5
#Filter skype audio before being downsampled and sent to SIP device.
#No Filtering
FilterParams=NONE
#RC lowpass filter
#RC,delay time (lower lowers cutoff),RC constant (higher lowers cutoff) - 50,40 is a good starting point
#FilterParams=RC,50,40
#FIR filter
#FIR,Order (higher sharper cutoff and more cpu),window type (RECTANGULAR,HANNING,HAMMING,BLACKMAN),filter type (LP,HP,BP),minFreq,maxFreq
#FilterParams=FIR,100,HANNING,LP,0,3200
#FilterParams=FIR,100,HANNING,HP,300,3200
#FilterParams=FIR,100,HANNING,BP,300,3200
#If yes, will send RTP packets to address received from the otherside
# instead of what was received in the session descriptor.
# This may help with one way audio problems.
enableSendRTPtoReceivedAddress=yes
#works with above setting - sending of rtpPackets can be redirected until receiving this number of packets. After that the address is locked.
lockRtpSendAddressAfterPackets=1
#Set to -1 to disable rfc2833 some providers use 96 most use 101
dtmf2833payloadtype=101
#Use these for SIP INFO msg support - first is the most common type
#dtmfinfotype=application/dtmf-relay
#dtmfinfotype=application/dtmf
#Use only if rfc2833 and INFO are not supported - uses more cpu
enableSIPInbandDtmfDetector=no
SipDtmfDetectorHitThreshold=30
SipDtmfDetectorSilenceThreshold=6
#params to control sip response address handling
useViaRport=yes
useViaReceived=yes
#send all responses using outbound proxy - outbound proxy must be set up
sendResponseUsingOutboundProxy=no
#sip response for any uncovered reason
baseFailureResponse=403
#sip response if remote skype user refused call
skypeRefusedResponse=603
#sip response if skype call failed, invalid skype user, or no skype credit
skypeFailedResponse=404
#sip response if skype returned unplaced status
skypeUnPlacedResponse=408
#sip response if called party is busy
skypeBusyResponse=600
#network buffers for skype api audio transport (0=leave at OS default)
TcpRxBufferSize=8192
TcpTxBufferSize=8192
#network buffers for RTP audio transport (0=leave at OS default)
RtpRxBufferSize=8192
RtpTxBufferSize=8192
#*** register server settings below *** not required if registration is not needed for phone
#set to yes to turn on server registrar or no to disable
is_registrar=no
#set to yes to allow register of users not already in registar database (users.db)
register_new_users=yes
#set to domains of server - see mjsip doc
#domain_names=192.168.0.4 somedomain.com
allowMultiContactsPerUser=noStack Trace
[2009-07-16 22:58:19] VERBOSE[2943] logger.c: Reliably Transmitting (no NAT) to 127.0.0.1:5070:
OPTIONS sip:127.0.0.1:5070 SIP/2.0
Via: SIP/2.0/UDP 218.186.12.242:5060;branch=z9hG4bK069374d8
From: "Unknown" <sip:Unknown@218.186.12.242>;tag=as5edc492e
To: <sip:127.0.0.1:5070>
Contact: <sip:Unknown@218.186.12.242>
Call-ID: 4f88c51125ed427c676427143169c508@218.186.12.242
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 16 Jul 2009 14:58:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[2009-07-16 22:58:19] VERBOSE[2943] logger.c:
<--- SIP read from 127.0.0.1:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 218.186.12.242:5060;branch=z9hG4bK069374d8;received=127.0.0.1
To: <sip:127.0.0.1:5070>
From: "Unknown" <sip:Unknown@218.186.12.242>;tag=as5edc492e
Call-ID: 4f88c51125ed427c676427143169c508@218.186.12.242
CSeq: 102 OPTIONS
Server: mjsip stack 1.6 sts
Content-Length: 308
Content-Type: application/sdp
v=0
o=Skype_Caller 1247756299 0 IN IP4 127.0.0.1
s=Session SIP/SDP
c=IN IP4 127.0.0.1
t=0 0
m=audio 63200 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=silenceSupp:off
<------------->
[2009-07-16 22:58:19] VERBOSE[2943] logger.c: --- (9 headers 14 lines) ---
[2009-07-16 22:58:19] VERBOSE[2943] logger.c: Really destroying SIP dialog '4f88c51125ed427c676427143169c508@218.186.12.242' Method: OPTIONS
[2009-07-16 22:58:19] NOTICE[2943] chan_sip.c: -- Re-registration for 1xxxxxxxxxx@proxy01.sipphone.com
[2009-07-16 22:58:20] NOTICE[2943] chan_sip.c: -- Re-registration for 3XXXXXXX@sip.pfingo.com
[2009-07-16 22:58:20] NOTICE[2943] chan_sip.c: Outbound Registration: Expiry for sip.pfingo.com is 120 sec (Scheduling reregistration in 105 s)
[2009-07-16 22:58:21] NOTICE[2943] chan_sip.c: Outbound Registration: Expiry for proxy01.sipphone.com is 120 sec (Scheduling reregistration in 105 s)
[2009-07-16 22:58:21] WARNING[2943] chan_sip.c: Got 200 OK on REGISTER that isn't a register
[2009-07-16 22:58:35] VERBOSE[2943] logger.c:
<--- SIP read from 127.0.0.1:5070 --->
INVITE sip:75973@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057
Max-Forwards: 70
To: <sip:75973@127.0.0.1:5060>
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
Contact: <sip:Skype_Caller@127.0.0.1:5070>
Expires: 3600
User-Agent: mjsip stack 1.6 sts
Content-Length: 308
Content-Type: application/sdp
v=0
o=Skype_Caller 1247756315 0 IN IP4 127.0.0.1
s=Session SIP/SDP
c=IN IP4 127.0.0.1
t=0 0
m=audio 63200 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=silenceSupp:off
<------------->
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: --- (12 headers 14 lines) ---
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Sending to 127.0.0.1 : 5070 (NAT)
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Using INVITE request as basis request - 914870468026@127.0.0.1
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found peer 'Skype'
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found RTP audio format 0
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found RTP audio format 8
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found RTP audio format 98
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found RTP audio format 97
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found RTP audio format 101
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Peer audio RTP is at port 127.0.0.1:63200
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found audio description format PCMU for ID 0
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found audio description format PCMA for ID 8
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found audio description format iLBC for ID 98
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found audio description format speex for ID 97
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Found audio description format telephone-event for ID 101
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Capabilities: us - 0x60c (ulaw|alaw|speex|ilbc), peer - audio=0x60c (ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x60c (ulaw|alaw|speex|ilbc)
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Peer audio RTP is at port 127.0.0.1:63200
[2009-07-16 22:58:35] DEBUG[2943] chan_sip.c: Call from peer 'Skype' is 1 out of 1
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: Looking for 75973 in from-trunk (domain 127.0.0.1)
[2009-07-16 22:58:35] VERBOSE[2943] logger.c: list_route: hop: <sip:Skype_Caller@127.0.0.1:5070>
[2009-07-16 22:58:35] VERBOSE[2943] logger.c:
<--- Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Length: 0
<------------>
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [75973@from-trunk:1] Set("SIP/Skype_Caller-091faeb8", "__FROM_DID=75973") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [75973@from-trunk:2] ExecIf("SIP/Skype_Caller-091faeb8", "0 |Set|CALLERID(name)=Skype_Caller") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [75973@from-trunk:3] Ringing("SIP/Skype_Caller-091faeb8", "") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c:
<--- Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Length: 0
<------------>
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [75973@from-trunk:4] Set("SIP/Skype_Caller-091faeb8", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [75973@from-trunk:5] SetCallerPres("SIP/Skype_Caller-091faeb8", "allowed_not_screened") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [75973@from-trunk:6] Goto("SIP/Skype_Caller-091faeb8", "from-did-direct|5000|1") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Goto (from-did-direct,5000,1)
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [5000@from-did-direct:1] Macro("SIP/Skype_Caller-091faeb8", "exten-vm|novm|5000") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/Skype_Caller-091faeb8", "user-callerid") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/Skype_Caller-091faeb8", "AMPUSER=Skype_Caller") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/Skype_Caller-091faeb8", "0?report") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/Skype_Caller-091faeb8", "1|Set|REALCALLERIDNUM=Skype_Caller") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: ExecIf
[2009-07-16 22:58:35] DEBUG[4584] func_db.c: DB: DEVICE/Skype_Caller/user not found in database.
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/Skype_Caller-091faeb8", "AMPUSER=") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] DEBUG[4584] func_db.c: DB: AMPUSER//cidname not found in database.
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/Skype_Caller-091faeb8", "AMPUSERCIDNAME=") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/Skype_Caller-091faeb8", "1?report") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Goto (macro-user-callerid,s,10)
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/Skype_Caller-091faeb8", "0?continue") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:11] Set("SIP/Skype_Caller-091faeb8", "__TTL=64") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/Skype_Caller-091faeb8", "1?continue") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Goto (macro-user-callerid,s,19)
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/Skype_Caller-091faeb8", "Using CallerID "skype2sipgate" <Skype_Caller>") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Noop
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Macro
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:2] Set("SIP/Skype_Caller-091faeb8", "RingGroupMethod=none") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:3] Set("SIP/Skype_Caller-091faeb8", "VMBOX=novm") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:4] Set("SIP/Skype_Caller-091faeb8", "EXTTOCALL=5000") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] DEBUG[4584] func_db.c: DB: CFU/5000 not found in database.
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:5] Set("SIP/Skype_Caller-091faeb8", "CFUEXT=") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] DEBUG[4584] func_db.c: DB: CFB/5000 not found in database.
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:6] Set("SIP/Skype_Caller-091faeb8", "CFBEXT=") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:7] Set("SIP/Skype_Caller-091faeb8", "RT=""") in new stack
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: Set
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/Skype_Caller-091faeb8", "record-enable|5000|IN") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/Skype_Caller-091faeb8", "1?check") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Goto (macro-record-enable,s,4)
[2009-07-16 22:58:35] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/Skype_Caller-091faeb8", "recordingcheck|20090716-225835|1247756315.54") in new stack
[2009-07-16 22:58:35] VERBOSE[4584] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: recordingcheck|20090716-225835|1247756315.54: Inbound recording not enabled
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- AGI Script recordingcheck completed, returning 0
[2009-07-16 22:58:36] DEBUG[4584] app_macro.c: Executed application: AGI
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/Skype_Caller-091faeb8", "") in new stack
[2009-07-16 22:58:36] DEBUG[4584] app_macro.c: Executed application: Macro
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/Skype_Caller-091faeb8", "dial||tr|5000") in new stack
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- Executing [s@macro-dial:1] GotoIf("SIP/Skype_Caller-091faeb8", "1?dial") in new stack
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- Goto (macro-dial,s,3)
[2009-07-16 22:58:36] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- Executing [s@macro-dial:3] AGI("SIP/Skype_Caller-091faeb8", "dialparties.agi") in new stack
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: dialparties.agi: Starting New Dialparties.agi
[2009-07-16 22:58:36] VERBOSE[4587] logger.c: == Parsing '/etc/asterisk/manager.conf': [2009-07-16 22:58:36] VERBOSE[4587] logger.c: Found
[2009-07-16 22:58:36] VERBOSE[4587] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [2009-07-16 22:58:36] VERBOSE[4587] logger.c: Found
[2009-07-16 22:58:36] VERBOSE[4587] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [2009-07-16 22:58:36] VERBOSE[4587] logger.c: Found
[2009-07-16 22:58:36] VERBOSE[4587] logger.c: == Manager 'admin' logged on from 127.0.0.1
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: dialparties.agi: Caller ID name is 'skype2sipgate' number is 'Skype_Caller'
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: dialparties.agi: USE_CONFIRMATION: 'FALSE'
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: dialparties.agi: RINGGROUP_INDEX: ''
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: dialparties.agi: Methodology of ring is 'none'
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- dialparties.agi: Added extension 5000 to extension map
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: > dialparties.agi: Extension 5000 has call screening off
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- dialparties.agi: Extension 5000 cf is disabled
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- dialparties.agi: Extension 5000 do not disturb is disabled
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: > dialparties.agi: extnum 5000 has: cw: 1; hascfb: 0 [] hascfu: 0 []
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: > dialparties.agi: ExtensionState: 0
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- dialparties.agi: DbDel CALLTRACE/5000 - Caller ID is not defined
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- dialparties.agi: Filtered ARG3: 5000
[2009-07-16 22:58:36] VERBOSE[4587] logger.c: == Manager 'admin' logged off from 127.0.0.1
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- AGI Script dialparties.agi completed, returning 0
[2009-07-16 22:58:36] DEBUG[4584] app_macro.c: Executed application: AGI
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- Executing [s@macro-dial:7] Dial("SIP/Skype_Caller-091faeb8", "SIP/5000||tr") in new stack
[2009-07-16 22:58:36] DEBUG[4584] chan_sip.c: Call to peer '5000' is 1 out of 50
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- Called 5000
[2009-07-16 22:58:36] VERBOSE[4584] logger.c:
<--- Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Length: 0
<------------>
[2009-07-16 22:58:36] VERBOSE[4584] logger.c: -- SIP/5000-091e6ff0 is ringing
[2009-07-16 22:58:50] DEBUG[2943] chan_sip.c: Strict routing enforced for session 125ceb677aa251030b3d30c75b29a84b@192.168.1.147
[2009-07-16 22:58:50] VERBOSE[4584] logger.c: -- SIP/5000-091e6ff0 answered SIP/Skype_Caller-091faeb8
[2009-07-16 22:58:50] VERBOSE[4584] logger.c: Audio is at 218.186.12.242 port 15986
[2009-07-16 22:58:50] VERBOSE[4584] logger.c: Adding codec 0x4 (ulaw) to SDP
[2009-07-16 22:58:50] VERBOSE[4584] logger.c: Adding codec 0x8 (alaw) to SDP
[2009-07-16 22:58:50] VERBOSE[4584] logger.c: Adding codec 0x400 (ilbc) to SDP
[2009-07-16 22:58:50] VERBOSE[4584] logger.c: Adding codec 0x200 (speex) to SDP
[2009-07-16 22:58:50] VERBOSE[4584] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[2009-07-16 22:58:50] VERBOSE[4584] logger.c:
<--- Reliably Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 2922 2922 IN IP4 218.186.12.242
s=session
c=IN IP4 218.186.12.242
t=0 0
m=audio 15986 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
[2009-07-16 22:58:50] VERBOSE[2943] logger.c: Retransmitting #1 (no NAT) to 127.0.0.1:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 2922 2922 IN IP4 218.186.12.242
s=session
c=IN IP4 218.186.12.242
t=0 0
m=audio 15986 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2009-07-16 22:58:50] VERBOSE[2943] logger.c: Retransmitting #2 (no NAT) to 127.0.0.1:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 2922 2922 IN IP4 218.186.12.242
s=session
c=IN IP4 218.186.12.242
t=0 0
m=audio 15986 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2009-07-16 22:58:51] VERBOSE[2943] logger.c: Retransmitting #3 (no NAT) to 127.0.0.1:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 2922 2922 IN IP4 218.186.12.242
s=session
c=IN IP4 218.186.12.242
t=0 0
m=audio 15986 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2009-07-16 22:58:51] VERBOSE[2943] logger.c: Retransmitting #4 (no NAT) to 127.0.0.1:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 2922 2922 IN IP4 218.186.12.242
s=session
c=IN IP4 218.186.12.242
t=0 0
m=audio 15986 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2009-07-16 22:58:53] VERBOSE[2943] logger.c: Retransmitting #5 (no NAT) to 127.0.0.1:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 2922 2922 IN IP4 218.186.12.242
s=session
c=IN IP4 218.186.12.242
t=0 0
m=audio 15986 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2009-07-16 22:58:56] VERBOSE[2943] logger.c: Retransmitting #6 (no NAT) to 127.0.0.1:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057;received=127.0.0.1
From: "skype2sipgate" <sip:Skype_Caller@192.168.1.147:5060>;tag=z9hG4bK26357349
To: <sip:75973@127.0.0.1:5060>;tag=as4a815a65
Call-ID: 914870468026@127.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:75973@218.186.12.242>
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 2922 2922 IN IP4 218.186.12.242
s=session
c=IN IP4 218.186.12.242
t=0 0
m=audio 15986 RTP/AVP 0 8 98 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2009-07-16 22:59:00] WARNING[2943] chan_sip.c: Maximum retries exceeded on transmission 914870468026@127.0.0.1 for seqno 1 (Critical Response)
[2009-07-16 22:59:00] WARNING[2943] chan_sip.c: Hanging up call 914870468026@127.0.0.1 - no reply to our critical packet.
[2009-07-16 22:59:00] DEBUG[4584] chan_sip.c: Call to peer '5000' removed from call limit 50
[2009-07-16 22:59:00] DEBUG[4584] chan_sip.c: Strict routing enforced for session 125ceb677aa251030b3d30c75b29a84b@192.168.1.147
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/Skype_Caller-091faeb8' in macro 'dial'
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/Skype_Caller-091faeb8' in macro 'exten-vm'
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/Skype_Caller-091faeb8'
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Executing [h@macro-dial:1] Macro("SIP/Skype_Caller-091faeb8", "hangupcall") in new stack
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/Skype_Caller-091faeb8", "vw") in new stack
[2009-07-16 22:59:00] DEBUG[4584] app_macro.c: Executed application: ResetCDR
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/Skype_Caller-091faeb8", "") in new stack
[2009-07-16 22:59:00] DEBUG[4584] app_macro.c: Executed application: NoCDR
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/Skype_Caller-091faeb8", "1?skiprg") in new stack
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Goto (macro-hangupcall,s,6)
[2009-07-16 22:59:00] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/Skype_Caller-091faeb8", "1?skipblkvm") in new stack
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Goto (macro-hangupcall,s,9)
[2009-07-16 22:59:00] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/Skype_Caller-091faeb8", "1?theend") in new stack
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Goto (macro-hangupcall,s,11)
[2009-07-16 22:59:00] DEBUG[4584] app_macro.c: Executed application: GotoIf
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/Skype_Caller-091faeb8", "") in new stack
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/Skype_Caller-091faeb8' in macro 'hangupcall'
[2009-07-16 22:59:00] VERBOSE[4584] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/Skype_Caller-091faeb8'
[2009-07-16 22:59:00] DEBUG[4584] chan_sip.c: Call from peer 'Skype' removed from call limit 1
[2009-07-16 22:59:00] VERBOSE[2943] logger.c: Really destroying SIP dialog '914870468026@127.0.0.1' Method: INVITE
[2009-07-16 22:59:14] VERBOSE[4592] logger.c: == Parsing '/etc/asterisk/manager.conf': [2009-07-16 22:59:14] VERBOSE[4592] logger.c: Found
[2009-07-16 22:59:14] VERBOSE[4592] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [2009-07-16 22:59:14] VERBOSE[4592] logger.c: Found
[2009-07-16 22:59:14] VERBOSE[4592] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [2009-07-16 22:59:14] VERBOSE[4592] logger.c: Found
[2009-07-16 22:59:14] VERBOSE[4592] logger.c: == Manager 'admin' logged on from 127.0.0.1
[2009-07-16 22:59:14] VERBOSE[4592] logger.c: == Manager 'admin' logged off from 127.0.0.1
[2009-07-16 22:59:15] VERBOSE[4594] logger.c: == Parsing '/etc/asterisk/manager.conf': [2009-07-16 22:59:15] VERBOSE[4594] logger.c: Found
[2009-07-16 22:59:15] VERBOSE[4594] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [2009-07-16 22:59:15] VERBOSE[4594] logger.c: Found
[2009-07-16 22:59:15] VERBOSE[4594] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [2009-07-16 22:59:15] VERBOSE[4594] logger.c: Found
[2009-07-16 22:59:15] VERBOSE[4594] logger.c: == Manager 'admin' logged on from 127.0.0.1__________________
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- Where is externip stored?
- Can i just change the IP Address?
- Can not make or receive calls.. Please help..
- Cannot Transfer outbound calls
- Error Exit:1
- Logged in: USERNAME (Logout) link missing
- Upgrade Asterisk 1.4.42 to 1.8
- Problem
- [Working!] Cisco XML Directory with LDAP lookup to iCloud
- Codac g729 error
- Call Forwarding issue (DAHDI)
- Turkey indications
Make sure the localhost address is used consistently
Hi,
From what I see your gateway packets come from 192.168.1.147:
From: "skype2sipgate" sip:Skype_Caller@192.168.1.147:5060;tag=z9hG4bK26357349
To: sip:75973@127.0.0.1:5060;tag=as4a815a65
Your trunk specifically allows 127.0.0.1:
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.255
Try fixing that and see what happens.
HTH,
Vladimir
tried that..no luck
I tried that.I've now changed the registration section of siptosis.cfg file to
host_port=5070
contact_url=sip:Skype_Caller@127.0.0.1:5070
from_url="SkypeCaller"
username=Skype_Caller
passwd=goodpassword
and have removed the deny & permit lines in the trunk settings..
my * server uses my externalip in the FROM request to 127.0.0.1.do you think this maybe why? is it possible to use 127.0.0.1 in the FROM request?
Let VoIP set you free...
Same here
I have this same problem. Removing the deny and allow IP entries didn't change the behavior. Did you ever find a resolution to this problem?
solved
yup, i gave my PBX'es LAN IP address instead of 127.x.x.x.
#Sample config with NO registration - use if above auto config doesn't work - change 127.0.0.1 to ip address of computer running siptosis
# username and password not important in this mode
#Set to available port to transport SIP messages on siptosis computer
host_port=5070
contact_url=sip:Skype_Caller@192.168.1.147:5070
from_url="SkypeCaller"
#from_url="Skype_Caller"
username=Skype_Caller
passwd=whateverthepwdis
realm=192.168.1.147
Let VoIP set you free...