Hi,
I have asterisk 10.12.2 on Centos 5.
sip.conf
[general]
debug=yes
autoprune=no
autoregister=yes
[gmail]
type=client
serverhost=talk.google.com
username=mygmail@gmail.com/Talk
secret=GmailPass
port=5222
buddy=myBuddy1@gmail.com
buddy=myBuddy2@gmail.com
statusmessage="Online via SIP Server"
timeout=100gtalk.conf
[general]
context=users
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=users
connection=gmail
[myBuddy1]
username=myBuddy1@gmail.com
disallow=all
allow=ulaw
allow=alaw
context=users
connection=gmail
[myBuddy2]
username=myBuddy2@gmail.com
disallow=all
allow=ulaw
allow=alaw
context=users
connection=gmail
[myBuddy3]
username=myBuddy3@gmail.com
disallow=all
allow=ulaw
allow=alaw
context=users
connection=gmailsip.conf
[mobile]
context=users
port=5060
type=friend
secret=mobileSecret
dtmfmode=rfc2833
nat=yes
host=dynamic
username=mobile
[laptop]
context=users
port=5060
type=friend
secret=laptopSecret
dtmfmode=rfc2833
nat=yes
host=dynamic
username=laptopextensions.conf
[docs:users]
[users]
include => longdistance2
include => parkedcalls
exten=>mygmail@gmail.com,1,Answer()
exten=>mygmail@gmail.com,n,Wait(1)
exten=>mygmail@gmail.com,n,SendDTMF(1)
exten=>mygmail@gmail.com,n,NoOp(Callerid ${CALLERID(name)})
exten=>mygmail@gmail.com,n,Set(CALLERID(num)=${SHIFT(CALLERID(name),@)})
exten=>mygmail@gmail.com,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten=>mygmail@gmail.com,n,Dial(SIP/laptop)
;exten=>mygmail@gmail.com,n,VoiceMail(603@vm,u)
exten=>603,1,Dial(SIP/mobile,20)
exten=>603,n,VoiceMail(603@vm,u)
exten=>604,1,Dial(SIP/laptop,20)
exten => 200,1,Dial(Gtalk/gmail/myBuddy3@gmail.com)
exten => 201,1,Dial(Gtalk/gmail/myBuddy2@gmail.com)
exten => 202,1,Dial(Gtalk/gmail/myBuddy1@gmail.com)
[longdistance2]
;testing USA+Canada
exten=> _91NXXNXXXXXX,1,Dial(Gtalk/gmail/+${EXTEN:1}@voice.google.com)in voicemail.conf, I have added:
[vm]
603=>secret,Adeel Ahmed,mygmail@gmail.com,attach=no|tz=mountian|maxmsg=10[color=#FF0000]PROBLEM AND LOG OUTPUT[/color]
My gmail is: mygmail@gmail.com
My another gmail for testing purpose is: myBuddy1@gmail.com
Other two: myBuddy2@gmail.com and myBuddy3@gmail.com are my friends/contacts of mygmail@gmail.com
On BlackBerry, I have SIP Account registered: laptop
CLI: sip show peers
*CLI>sip show peers
Name/username Host Dyn Forcerport ACL Port Status Description
laptop/laptop 42.83.85.74 D N 33975 Unmonitored
mobile/mobile (Unspecified) D N 0 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 1 offline]On my SYSTEM, I have Google Talk and I am logged in with myBuddy1@gmail.com
jabber show buddies shows list of online friends and one of the is myBuddy1@gmail.com
Buddy: myBuddy1darnexams@gmail.com
Resource: Talk.v105F6EFDD82
node: http://www.google.com/xmpp/client/caps
version: 1.0.0.105
Jingle capable: yes
Status: 1
Priority: 0
On Google Talk, mygmail@gmail.com is online and shows status: “Online via SIP Server”
When I call from Google Talk to mygmail@gmail.com , call gets answered automatically due to extensions.conf settings but after like 3 seconds, I hear tone… “beep beep beep beep” and then call disconnects.
I am not receiving call on SIP account laptop which is registered on BlackBerry.
Here is the complete log in three parts (due to 60000 limit)
[color=#0000FF]x.x.x.x is my SERVER IP[/color]
[2013-08-12 21:32:23] VERBOSE[3344] logger.c: Asterisk Queue Logger restarted
[2013-08-12 21:32:28] VERBOSE[3297] res_jabber.c:
JABBER: gmail INCOMING: <iq to="mygmail@gmail.com/Talk1EA42B20" type="set" id="61" from="myBuddy1@gmail.com/Talk.v1051587ED6D"><session type="initiate" id="900097919"
initiator="myBuddy1@gmail.com/Talk.v1051587ED6D" xmlns="http://www.google.com/session"><description xmlns="http://www.google.com/session/phone"><payload-type id="103"
name="ISAC"/><payload-type id="97" name="IPCMWB"/><payload-type id="4" name="G723"/><payload-type id="100" name="EG711U"/><payload-type id="101"
name="EG711A"/><payload-type id="0" name="PCMU"/><payload-type id="8" name="PCMA"/><payload-type id="13" name="CN"/><payload-type id="102" name="iLBC"/><payload-type
id="117" name="red"/><payload-type id="106" name="audio/telephone-event"/></description></session></iq>
[2013-08-12 21:32:28] DEBUG[3297] res_jabber.c: JABBER: Handling paktype IQ
[2013-08-12 21:32:28] DEBUG[3297] chan_gtalk.c: The client is myBuddy1 for alloc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x95dfe10'
[2013-08-12 21:32:28] DEBUG[3297] res_rtp_asterisk.c: Allocated port 19530 for RTP instance '0x95dfe10'
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: RTP instance '0x95dfe10' is setup and ready to go
[2013-08-12 21:32:28] DEBUG[3297] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x95dfe10'
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 103 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 97 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 4 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 100 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 101 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 0 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 8 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 13 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 102 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 117 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Setting payload 106 based on m type on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 0 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 4 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 8 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 13 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 97 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 100 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 101 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 102 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 103 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 106 on 0x95dffbc
[2013-08-12 21:32:28] DEBUG[3297] rtp_engine.c: Incorporating payload 117 on 0x95dffbc
[2013-08-12 21:32:28] VERBOSE[3297] res_jabber.c:
JABBER: gmail OUTGOING: <iq type='result' from='mygmail@gmail.com/Talk1EA42B20' to='myBuddy1@gmail.com/Talk.v1051587ED6D' id='61'/>
[2013-08-12 21:32:28] DEBUG[3297] netsock2.c: Splitting 'google.com' into...
[2013-08-12 21:32:28] DEBUG[3297] netsock2.c: ...host 'google.com' and port ''.
[2013-08-12 21:32:28] DEBUG[3291] devicestate.c: No provider found, checking channel drivers for Gtalk - myBuddy1
[2013-08-12 21:32:28] DEBUG[3291] devicestate.c: Changing state for Gtalk/myBuddy1 - state 2 (In use)
[2013-08-12 21:32:28] DEBUG[3291] devicestate.c: device 'Gtalk/myBuddy1' state '2'
[2013-08-12 21:32:28] DEBUG[3327] app_queue.c: Device 'Gtalk/myBuddy1' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[2013-08-12 21:32:28] DEBUG[3675] pbx.c: Launching 'Answer'
[2013-08-12 21:32:28] VERBOSE[3675] pbx.c: -- Executing [mygmail@gmail.com@users:1] Answer("Gtalk/myBuddy1-7776", "") in new stack
[2013-08-12 21:32:28] DEBUG[3675] chan_gtalk.c: Answer!
[2013-08-12 21:32:28] VERBOSE[3675] res_jabber.c:
JABBER: gmail OUTGOING: <iq type='set' to='myBuddy1@gmail.com/Talk.v1051587ED6D' from='mygmail@gmail.com/Talk1EA42B20' id='aaaat'><session
xmlns='http://www.google.com/session' type='accept' initiator='myBuddy1@gmail.com/Talk.v1051587ED6D' id='900097919'><description
xmlns='http://www.google.com/session/phone' xml:lang='en'><payload-type id='0' name='PCMU' clockrate='8000' bitrate='64000'/><payload-type id='100' name='EG711U'
clockrate='8000' bitrate='64000'/><payload-type id='8' name='PCMA' clockrate='8000' bitrate='64000'/><payload-type id='101' name='EG711A' clockrate='8000'
bitrate='64000'/><payload-type id='101' name='telephone-event' clockrate='8000'/></description></session></iq>
[2013-08-12 21:32:28] DEBUG[3291] devicestate.c: No provider found, checking channel drivers for Gtalk - myBuddy1
[2013-08-12 21:32:28] DEBUG[3291] devicestate.c: Changing state for Gtalk/myBuddy1 - state 2 (In use)
[2013-08-12 21:32:28] DEBUG[3291] devicestate.c: device 'Gtalk/myBuddy1' state '2'
[2013-08-12 21:32:28] DEBUG[3327] app_queue.c: Device 'Gtalk/myBuddy1' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[2013-08-12 21:32:28] DEBUG[3675] chan_gtalk.c: Don't know how to indicate condition '-1'
[2013-08-12 21:32:28] DEBUG[3297] acl.c: Not an IPv4 nor IPv6 address, cannot get port.
[2013-08-12 21:32:28] DEBUG[3297] acl.c: For destination '173.194.46.37', our source address is 'x.x.x.x'.
[2013-08-12 21:32:28] VERBOSE[3297] res_jabber.c:
JABBER: gmail OUTGOING: <iq from='mygmail@gmail.com/Talk1EA42B20' to='myBuddy1@gmail.com/Talk.v1051587ED6D' type='set' id='aaaau'><session type='candidates'
id='900097919' initiator='myBuddy1@gmail.com/Talk.v1051587ED6D' xmlns='http://www.google.com/session'><candidate name='rtp' address='x.x.x.x' port='19530'
username='6e7c82b5023e7f37' password='6e004b1f590e23db' preference='1.00' protocol='udp' type='local' network='0' generation='0'/><transport
xmlns='http://www.google.com/transport/p2p'/></session></iq>
[2013-08-12 21:32:28] DEBUG[3297] res_jabber.c: XML parsing successful
[2013-08-12 21:32:28] VERBOSE[3297] res_jabber.c:
JABBER: gmail INCOMING: <iq type="result" to="mygmail@gmail.com/Talk1EA42B20" id="aaaat" from="myBuddy1@gmail.com/Talk.v1051587ED6D"/>
[2013-08-12 21:32:28] DEBUG[3297] res_jabber.c: JABBER: Handling paktype IQ
[2013-08-12 21:32:28] DEBUG[3297] res_jabber.c: XML parsing successful
[2013-08-12 21:32:28] VERBOSE[3297] res_jabber.c:
JABBER: gmail INCOMING: <iq type="result" to="mygmail@gmail.com/Talk1EA42B20" id="aaaau" from="myBuddy1@gmail.com/Talk.v1051587ED6D"/>
[2013-08-12 21:32:28] DEBUG[3297] res_jabber.c: JABBER: Handling paktype IQ
[2013-08-12 21:32:28] DEBUG[3297] res_jabber.c: XML parsing successful
[2013-08-12 21:32:28] DEBUG[3675] channel.c: Didn't receive a media frame from Gtalk/myBuddy1-7776 within 500 ms of answering. Continuing anyway
[2013-08-12 21:32:28] DEBUG[3675] pbx.c: Launching 'Wait'
[2013-08-12 21:32:28] VERBOSE[3675] pbx.c: -- Executing [mygmail@gmail.com@users:2] Wait("Gtalk/myBuddy1-7776", "1") in new stack
[2013-08-12 21:32:29] DEBUG[3675] pbx.c: Launching 'SendDTMF'
[2013-08-12 21:32:29] VERBOSE[3675] pbx.c: -- Executing [mygmail@gmail.com@users:3] SendDTMF("Gtalk/myBuddy1-7776", "1") in new stack
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Function result is 'myBuddy1@gmail.com/Talk.v1051587ED6D'
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Launching 'NoOp'
[2013-08-12 21:32:30] VERBOSE[3675] pbx.c: -- Executing [mygmail@gmail.com@users:4] NoOp("Gtalk/myBuddy1-7776", "Callerid myBuddy1@gmail.com/Talk.v1051587ED6D") in
new stack
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Evaluating 'CALLERID(name)' (from 'CALLERID(name)}' len 14)
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Function result is 'myBuddy1@gmail.com/Talk.v1051587ED6D'
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Function result is 'myBuddy1'
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Launching 'Set'
[2013-08-12 21:32:30] VERBOSE[3675] pbx.c: -- Executing [mygmail@gmail.com@users:5] Set("Gtalk/myBuddy1-7776", "CALLERID(num)=myBuddy1") in new stack
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Function result is 'myBuddy1'
[2013-08-12 21:32:30] DEBUG[3675] db.c: Unable to find key 'myBuddy1' in family 'cidname'
[2013-08-12 21:32:30] DEBUG[3675] func_db.c: DB: cidname/myBuddy1 not found in database.
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Function result is ''
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Launching 'Set'
[2013-08-12 21:32:30] VERBOSE[3675] pbx.c: -- Executing [mygmail@gmail.com@users:6] Set("Gtalk/myBuddy1-7776", "CALLERID(name)=") in new stack
[2013-08-12 21:32:30] DEBUG[3675] pbx.c: Launching 'Dial'
[2013-08-12 21:32:30] VERBOSE[3675] pbx.c: -- Executing [mygmail@gmail.com@users:7] Dial("Gtalk/myBuddy1-7776", "SIP/laptop") in new stack
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Asked to create a SIP channel with formats: (ulaw)
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Allocating new SIP dialog for 26a56b813f24e16f652e9c4b4db1e503@x.x.x.x:5060 - INVITE (No RTP)
[2013-08-12 21:32:30] DEBUG[3675] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x95eb348'
[2013-08-12 21:32:30] DEBUG[3675] res_rtp_asterisk.c: Allocated port 15562 for RTP instance '0x95eb348'
[2013-08-12 21:32:30] DEBUG[3675] rtp_engine.c: RTP instance '0x95eb348' is setup and ready to go
[2013-08-12 21:32:30] DEBUG[3675] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x95eb348'
[2013-08-12 21:32:30] VERBOSE[3675] netsock2.c: == Using SIP RTP CoS mark 5
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Setting NAT on RTP to On
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[2013-08-12 21:32:30] DEBUG[3675] acl.c: For destination '42.83.85.74', our source address is 'x.x.x.x'.
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Setting SIP_TRANSPORT_UDP with address x.x.x.x:5060
[2013-08-12 21:32:30] DEBUG[3675] format_pref.c: Could not find preferred codec - Going for the best codec
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: *** Our native formats are (ulaw)
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: *** Joint capabilities are (ulaw)
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw)
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw)
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: This channel will not be able to handle video.
[2013-08-12 21:32:30] DEBUG[3675] rtp_engine.c: Seeded SDP of 'SIP/laptop-00000004' with that of 'Gtalk/myBuddy1-7776'
[2013-08-12 21:32:30] DEBUG[3675] channel.c: Not copying variable DIALEDTIME.
[2013-08-12 21:32:30] DEBUG[3675] channel.c: Not copying variable ANSWEREDTIME.
[2013-08-12 21:32:30] DEBUG[3675] channel.c: Not copying variable DIALEDPEERNAME.
[2013-08-12 21:32:30] DEBUG[3675] channel.c: Not copying variable DIALEDPEERNUMBER.
[2013-08-12 21:32:30] DEBUG[3675] channel.c: Not copying variable DIALSTATUS.
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Outgoing Call for laptop
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Updating call counter for outgoing call
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: This call needs video offers, but there's no video support enabled!
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: ** Our capability: (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: ** Our prefcodec: (ulaw)
[2013-08-12 21:32:30] VERBOSE[3675] chan_sip.c: Audio is at 15562
[2013-08-12 21:32:30] VERBOSE[3675] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2013-08-12 21:32:30] VERBOSE[3675] chan_sip.c: Adding codec 100002 (gsm) to SDP
[2013-08-12 21:32:30] VERBOSE[3675] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2013-08-12 21:32:30] VERBOSE[3675] chan_sip.c: Adding codec 100017 (testlaw) to SDP
[2013-08-12 21:32:30] VERBOSE[3675] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: -- Done with adding codecs to SDP
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|h263|testlaw)
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Initializing initreq for method INVITE - callid 0d9685d52efbf8664d60dd3a0a1608c7@x.x.x.x:5060
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 0 [ 43]: INVITE sip:laptop@10.3.93.149:55996 SIP/2.0
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0fefe9ed;rport
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 3 [ 62]: From: "myBuddy1" <sip:myBuddy1@x.x.x.x>;tag=as7c4ddd4a
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 4 [ 34]: To: <sip:laptop@10.3.93.149:55996>
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 5 [ 43]: Contact: <sip:myBuddy1@x.x.x.x:5060>
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 6 [ 60]: Call-ID: 0d9685d52efbf8664d60dd3a0a1608c7@x.x.x.x:5060
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 10.12.2
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 9 [ 35]: Date: Mon, 12 Aug 2013 17:32:30 GMT
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
[2013-08-12 21:32:30] VERBOSE[3675] chan_sip.c: Reliably Transmitting (NAT) to 42.83.85.74:49619:
INVITE sip:laptop@10.3.93.149:55996 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0fefe9ed;rport
Max-Forwards: 70
From: "myBuddy1" <sip:myBuddy1@x.x.x.x>;tag=as7c4ddd4a
To: <sip:laptop@10.3.93.149:55996>
Contact: <sip:myBuddy1@x.x.x.x:5060>
Call-ID: 0d9685d52efbf8664d60dd3a0a1608c7@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.12.2
Date: Mon, 12 Aug 2013 17:32:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1040529696 1040529696 IN IP4 x.x.x.x
s=Asterisk PBX 10.12.2
c=IN IP4 x.x.x.x
t=0 0
m=audio 15562 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #32
[2013-08-12 21:32:30] DEBUG[3675] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 42.83.85.74:49619
[2013-08-12 21:32:30] VERBOSE[3675] app_dial.c: -- Called SIP/laptop
[2013-08-12 21:32:30] DEBUG[3301] chan_sip.c: SIP TIMER: Rescheduling retransmission #32 (1) INVITE - 5
[2013-08-12 21:32:30] DEBUG[3301] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #32))
[2013-08-12 21:32:30] VERBOSE[3301] chan_sip.c: Retransmitting #1 (NAT) to 42.83.85.74:49619:
INVITE sip:laptop@10.3.93.149:55996 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0fefe9ed;rport
Max-Forwards: 70
From: "myBuddy1" <sip:myBuddy1@x.x.x.x>;tag=as7c4ddd4a
To: <sip:laptop@10.3.93.149:55996>
Contact: <sip:myBuddy1@x.x.x.x:5060>
Call-ID: 0d9685d52efbf8664d60dd3a0a1608c7@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.12.2
Date: Mon, 12 Aug 2013 17:32:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1040529696 1040529696 IN IP4 x.x.x.x
s=Asterisk PBX 10.12.2
c=IN IP4 x.x.x.x
t=0 0
m=audio 15562 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2013-08-12 21:32:30] DEBUG[3301] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 42.83.85.74:49619
[2013-08-12 21:32:31] DEBUG[3301] chan_sip.c: SIP TIMER: Rescheduling retransmission #32 (2) INVITE - 5
[2013-08-12 21:32:31] DEBUG[3301] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #32))
[2013-08-12 21:32:31] VERBOSE[3301] chan_sip.c: Retransmitting #2 (NAT) to 42.83.85.74:49619:
INVITE sip:laptop@10.3.93.149:55996 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0fefe9ed;rport
Max-Forwards: 70
From: "myBuddy1" <sip:myBuddy1@x.x.x.x>;tag=as7c4ddd4a
To: <sip:laptop@10.3.93.149:55996>
Contact: <sip:myBuddy1@x.x.x.x:5060>
Call-ID: 0d9685d52efbf8664d60dd3a0a1608c7@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.12.2
Date: Mon, 12 Aug 2013 17:32:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1040529696 1040529696 IN IP4 x.x.x.x
s=Asterisk PBX 10.12.2
c=IN IP4 x.x.x.x
t=0 0
m=audio 15562 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2013-08-12 21:32:31] DEBUG[3301] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 42.83.85.74:49619
[2013-08-12 21:32:33] VERBOSE[3297] res_jabber.c:
JABBER: gmail INCOMING: <presence from="myBuddy1@gmail.com/Talk.v1051587ED6D" to="mygmail@gmail.com"><status/><priority>0</priority><c
node="http://www.google.com/xmpp/client/caps" ver="1.0.0.105" ext="voice-v1" xmlns="http://jabber.org/protocol/caps"/><x stamp="20130812T17:32:27"
xmlns="jabber:x:delay"/><x xmlns="vcard-temp:x:update"><photo/></x></presence>
[2013-08-12 21:32:33] DEBUG[3297] res_jabber.c: JABBER: I am available ^_* 13
[2013-08-12 21:32:33] DEBUG[3297] res_jabber.c: JABBER: type is available
[2013-08-12 21:32:33] DEBUG[3297] res_jabber.c: JABBER: Handling paktype PRESENCE
[2013-08-12 21:32:33] DEBUG[3297] res_jabber.c: XML parsing successful
[2013-08-12 21:32:33] DEBUG[3301] chan_sip.c: SIP TIMER: Rescheduling retransmission #32 (3) INVITE - 5
[2013-08-12 21:32:33] DEBUG[3301] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #32))
[2013-08-12 21:32:33] VERBOSE[3301] chan_sip.c: Retransmitting #3 (NAT) to 42.83.85.74:49619:
INVITE sip:laptop@10.3.93.149:55996 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0fefe9ed;rport
Max-Forwards: 70
From: "myBuddy1" <sip:myBuddy1@x.x.x.x>;tag=as7c4ddd4a
To: <sip:laptop@10.3.93.149:55996>
Contact: <sip:myBuddy1@x.x.x.x:5060>
Call-ID: 0d9685d52efbf8664d60dd3a0a1608c7@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.12.2
Date: Mon, 12 Aug 2013 17:32:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1040529696 1040529696 IN IP4 x.x.x.x
s=Asterisk PBX 10.12.2
c=IN IP4 x.x.x.x
t=0 0
m=audio 15562 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2013-08-12 21:32:33] DEBUG[3301] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 42.83.85.74:49619
[2013-08-12 21:32:37] DEBUG[3301] chan_sip.c: SIP TIMER: Rescheduling retransmission #32 (4) INVITE - 5
[2013-08-12 21:32:37] DEBUG[3301] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 8000 ms (t1 500 ms (Retrans id #32))
[2013-08-12 21:32:37] VERBOSE[3301] chan_sip.c: Retransmitting #4 (NAT) to 42.83.85.74:49619:
INVITE sip:laptop@10.3.93.149:55996 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0fefe9ed;rport
Max-Forwards: 70
From: "myBuddy1" <sip:myBuddy1@x.x.x.x>;tag=as7c4ddd4a
To: <sip:laptop@10.3.93.149:55996>
Contact: <sip:myBuddy1@x.x.x.x:5060>
Call-ID: 0d9685d52efbf8664d60dd3a0a1608c7@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.12.2
Date: Mon, 12 Aug 2013 17:32:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1040529696 1040529696 IN IP4 x.x.x.x
s=Asterisk PBX 10.12.2
c=IN IP4 x.x.x.x
t=0 0
m=audio 15562 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
