Incoming Skype Calls gets disconnected

ninelives · July 16, 2009, 3:27pm

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 [email protected] for seqno 1 (Critical Response) [2009-07-16 22:59:00] WARNING[2943] chan_sip.c: Hanging up call [email protected] - 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 in

siptosis.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:[email protected]:5070
from_url=“SkypeCaller” sip:[email protected]: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=no

Stack 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" ;tag=as5edc492e 
To:  
Contact:  
Call-ID: [email protected] 
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:[email protected];tag=as5edc492e

Call-ID: [email protected]

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 ‘[email protected]’ Method: OPTIONS

[2009-07-16 22:58:19] NOTICE[2943] chan_sip.c: – Re-registration for [email protected]

[2009-07-16 22:58:20] NOTICE[2943] chan_sip.c: – Re-registration for [email protected]

[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:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5070;rport;branch=z9hG4bK429057

Max-Forwards: 70

To: sip:[email protected]:5060

From: “skype2sipgate” sip:[email protected]:5060;tag=z9hG4bK26357349

Call-ID: [email protected]

CSeq: 1 INVITE

Contact: sip:[email protected]: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 - [email protected]

[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:[email protected]: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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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 [email protected]

[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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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:[email protected]:5060;tag=z9hG4bK26357349

To: sip:[email protected]:5060;tag=as4a815a65

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: sip:[email protected]

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 [email protected] for seqno 1 (Critical Response)

[2009-07-16 22:59:00] WARNING[2943] chan_sip.c: Hanging up call [email protected] - 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 [email protected]

[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 ‘[email protected]’ 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

vmikhelson · July 18, 2009, 4:08pm

Hi,

From what I see your gateway packets come from 192.168.1.147:

From: “skype2sipgate” sip:[email protected]:5060;tag=z9hG4bK26357349
To: sip:[email protected]: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

ninelives · July 19, 2009, 6:19am

I tried that.I’ve now changed the registration section of siptosis.cfg file to

host_port=5070
contact_url=sip:[email protected]:5070
from_url=“SkypeCaller” sip:[email protected]:5070
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?

MrHanMan · August 17, 2010, 3:46pm

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?

ninelives · August 18, 2010, 11:14am

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:[email protected]:5070
from_url=“SkypeCaller” sip:[email protected]:5070
#from_url="Skype_Caller"
username=Skype_Caller
passwd=whateverthepwdis
realm=192.168.1.147

system · June 4, 2014, 6:51pm