Difference between revisions of "Sipgate"

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m (Asterisk sipgate moved to Sipgate)
 
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<table width="100%"> This should be a short How-To for integrating sipgate with LinuxMCE-Asterisk. It describes the steps what i have done and where the problems are.
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[[Category: Tutorials]]
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[[Category: Telecom]]
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[[Category: Phone Lines]]
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[[Category: VoIP]]
  
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'''1. Step Install LinuxMCE Core/Hybrid'''
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http://www.sipgate.co.uk
  
 +
==Automatic Setup==
 +
The setup is automatic from the House Setup Wizard using the following script
  
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'''2. Step Add a Media Director'''
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  /usr/pluto/bin/create_amp_inphonex.pl
  
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==Manual Setup==
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After the this i a running LinuxMCE system with one Media Director and one Windows XP Orbiter.
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'''3. Step Add Phone Lines'''
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In LinuxMCE Web Admin go to Devices -> Phone Lines.  
 
In LinuxMCE Web Admin go to Devices -> Phone Lines.  
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</pre>
 
</pre>
  
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''I don't know if i have to change the settings in the first box!?!?!?''
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'''AMP Configuration'''
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'''4. Step Reload the router'''
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Don't know if the reload is important at this point, but i think it's better to do it.
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'''5. Step AMP Configuration'''
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In LinuxMCE Web Admin go to Advanced -> Configuration -> Phones Setup -> Setup. LinuxMCE has inserted everything to asterisk. But at this point i can't call my sipgate-number. So go to Trunks and choose my trunk ''Trunk SIP/sipgate''. On Sipgate.at i find a configuration file for asterisk, the main difference is that i need ''allow=alow&alaw&alaw&ulaw&g729&gsm&slinear'' this in the Incoming and Outgoing Settings. So i added it and it worked!
 
In LinuxMCE Web Admin go to Advanced -> Configuration -> Phones Setup -> Setup. LinuxMCE has inserted everything to asterisk. But at this point i can't call my sipgate-number. So go to Trunks and choose my trunk ''Trunk SIP/sipgate''. On Sipgate.at i find a configuration file for asterisk, the main difference is that i need ''allow=alow&alaw&alaw&ulaw&g729&gsm&slinear'' this in the Incoming and Outgoing Settings. So i added it and it worked!
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username=9627932
 
username=9627932
 
</pre>
 
</pre>
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==TroubleShooting==
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'''Problems'''
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I can call my sipgate number from my cell phone but i only here a female voice who tells me that all circuits are busy now.  
 
I can call my sipgate number from my cell phone but i only here a female voice who tells me that all circuits are busy now.  
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Jan  7 15:44:04 DEBUG[14531] chan_sip.c: Stopping retransmission on '203858ad3dfce9881ac7c7155bee5404@217.10.66.71' of Request 102: Match Found
 
Jan  7 15:44:04 DEBUG[14531] chan_sip.c: Stopping retransmission on '203858ad3dfce9881ac7c7155bee5404@217.10.66.71' of Request 102: Match Found
 
</pre>
 
</pre>
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[[Category: Tutorials]]
 
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[[Category: Telecom]]
 
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[[Category: Phone Lines]]
 

Latest revision as of 16:16, 18 October 2010


http://www.sipgate.co.uk

Automatic Setup

The setup is automatic from the House Setup Wizard using the following script 

  /usr/pluto/bin/create_amp_inphonex.pl

Manual Setup

In LinuxMCE Web Admin go to Devices -> Phone Lines.

I don't make changes in the first box: 

 When dialing a local number, automatically prepend the area code
Also prepend a digit (1 for US/Canada, 0 for Europe)
Local Number Length

I choose my Provider (sipgate.at) and entered my Datas. 

Type  	Data  	Password  	Username  	Host  	        Phone number
SIP 	sipgate XXXXXXX 	9627932 	sipgate.at 	0720726742

AMP Configuration

In LinuxMCE Web Admin go to Advanced -> Configuration -> Phones Setup -> Setup. LinuxMCE has inserted everything to asterisk. But at this point i can't call my sipgate-number. So go to Trunks and choose my trunk Trunk SIP/sipgate. On Sipgate.at i find a configuration file for asterisk, the main difference is that i need allow=alow&alaw&alaw&ulaw&g729&gsm&slinear this in the Incoming and Outgoing Settings. So i added it and it worked! Here are my configurations:

Dial Rules: 

112
411
911
9|.

Outgoing Settings: 

allow=alow&alaw&alaw&ulaw&g729&gsm&slinear
auth=md5
authuser=9627932
callerid=9627932
canreinvite=no
context=from-internal
disallow=all
dtmfmode=inband
fromdomain=sipgate.at
fromuser=9627932
host=sipgate.at
insecure=very
nat=yes
qualify=no
secret=XXXXXX
type=peer
user=9627932
username=9627932

Incoming Settings 

allow=alow&alaw&alaw&ulaw&g729&gsm&slinear
auth=md5
authuser=9627932
callerid=9627932
canreinvite=no
context=from-internal
disallow=all
dtmfmode=inband
fromdomain=sipgate.at
fromuser=9627932
host=sipgate.at
insecure=very
nat=yes
qualify=no
secret=XXXXX
type=user
user=9627932
username=9627932

TroubleShooting

I can call my sipgate number from my cell phone but i only here a female voice who tells me that all circuits are busy now. I can See in the report Panel of AMP that someone has called me and the call was answered but i don't get a Message on my orbiters. I even can't talk from one Orbiter (MD) to another (Hybrid).

Solution for me is: replace context=from-internal with context=from-trunk Please try to confirm and if it works for you as well, make the final changes to this page...


Here is the log file from asterisk when i make a call from my cell phone: 

Connected to Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q currently running on dcero                                                                             uter (pid = 16397)
Verbosity is at least 3
Core debug is at least 1
    -- Executing Macro("SIP/9627932-96e1", "dialout-trunk|2|627932|") in new stack
    -- Executing GotoIf("SIP/9627932-96e1", "1?3:2)") in new stack
    -- Goto (macro-dialout-trunk,s,3)
    -- Executing Macro("SIP/9627932-96e1", "user-callerid") in new stack
    -- Executing DBget("SIP/9627932-96e1", "AMPUSER=DEVICE/9627932/user") in new stack
    -- DBget: varname=AMPUSER, family=DEVICE, key=9627932/user
    -- DBget: Value not found in database.
    -- Executing DBget("SIP/9627932-96e1", "AMPUSERCIDNAME=AMPUSER//cidname") in new stack
    -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname
    -- DBget: Value not found in database.
    -- Executing GotoIf("SIP/9627932-96e1", "1?5") in new stack
    -- Goto (macro-user-callerid,s,5)
    -- Executing NoOp("SIP/9627932-96e1", "Using CallerID "0699153xxxxx" <9627932>") in new stack
    -- Executing Macro("SIP/9627932-96e1", "record-enable|9627932|OUT") in new stack
    -- Executing GotoIf("SIP/9627932-96e1", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/9627932-96e1", "recordingcheck|20070106-123650|asterisk-16397-1168083410.0") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20070106-123650|asterisk-16397-1168083410.0: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/9627932-96e1", "No recording needed") in new stack
    -- Executing Macro("SIP/9627932-96e1", "outbound-callerid|2") in new stack
    -- Executing GotoIf("SIP/9627932-96e1", "1?3") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing DBget("SIP/9627932-96e1", "USEROUTCID=AMPUSER/9627932/outboundcid") in new stack
    -- DBget: varname=USEROUTCID, family=AMPUSER, key=9627932/outboundcid
    -- DBget: Value not found in database.
    -- Executing GotoIf("SIP/9627932-96e1", "1?6") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing NoOp("SIP/9627932-96e1", "CallerID set to "0699153xxxxx" <9627932>") in new stack
    -- Executing SetGroup("SIP/9627932-96e1", "OUT_2") in new stack
    -- Executing CheckGroup("SIP/9627932-96e1", "") in new stack
    -- Executing SetVar("SIP/9627932-96e1", "DIAL_NUMBER=627932") in new stack
    -- Executing SetVar("SIP/9627932-96e1", "DIAL_TRUNK=2") in new stack
    -- Executing AGI("SIP/9627932-96e1", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing SetVar("SIP/9627932-96e1", "OUTNUM=627932") in new stack
    -- Executing Cut("SIP/9627932-96e1", "custom=OUT_2|:|1") in new stack
    -- Executing GotoIf("SIP/9627932-96e1", "0?16") in new stack
    -- Executing Dial("SIP/9627932-96e1", "SIP/sipgate/627932") in new stack
    -- Called sipgate/627932
    -- SIP/sipgate-8e34 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Goto("SIP/9627932-96e1", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing NoOp("SIP/9627932-96e1", "Dial failed due to CONGESTION") in new stack
    -- Executing Macro("SIP/9627932-96e1", "outisbusy") in new stack
    -- Executing Playback("SIP/9627932-96e1", "allison7/all-circuits-busy-now") in new stack
    -- Playing 'allison7/all-circuits-busy-now' (language 'en')
    -- Executing Playback("SIP/9627932-96e1", "allison7/pls-try-call-later") in new stack
    -- Playing 'allison7/pls-try-call-later' (language 'en')
    -- Executing Macro("SIP/9627932-96e1", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/9627932-96e1", "w") in new stack
    -- Executing NoCDR("SIP/9627932-96e1", "") in new stack
    -- Executing Wait("SIP/9627932-96e1", "5") in new stack
    -- Executing Hangup("SIP/9627932-96e1", "") in new stack


This is the log (/etc/log/asterisk/full) when i call my sipgate number from my cell phone 

Jan  7 15:43:55 DEBUG[14531] chan_sip.c: Allocating new SIP dialog for 203858ad3dfce9881ac7c7155bee5404@217.10.66.71 - INVITE (With RTP)
Jan  7 15:43:55 DEBUG[14531] chan_sip.c: Setting NAT on RTP to 524288
Jan  7 15:43:55 DEBUG[14531] chan_sip.c: Checking SIP call limits for device 9627932
Jan  7 15:43:55 DEBUG[14531] chan_sip.c: build_route: Record-Route hop: <sip:217.116.119.252;lr=on>
Jan  7 15:43:55 DEBUG[14531] chan_sip.c: build_route: Record-Route hop: <sip:217.10.79.8;ftag=as032821cf;lr=on>
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Macro'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Macro("SIP/9627932-f491", "dialout-trunk|2|627932|") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Expression result is '1'
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'GotoIf'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing GotoIf("SIP/9627932-f491", "1?3:2)") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Goto (macro-dialout-trunk,s,3)
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Macro'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Macro("SIP/9627932-f491", "user-callerid") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'DBget'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing DBget("SIP/9627932-f491", "AMPUSER=DEVICE/9627932/user") in new stack
Jan  7 15:43:55 WARNING[15661] app_db.c: This application has been deprecated, please use the ${DB(family/key)} function instead.
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- DBget: varname=AMPUSER, family=DEVICE, key=9627932/user
Jan  7 15:43:55 DEBUG[15661] db.c: Unable to find key '9627932/user' in family 'DEVICE'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- DBget: Value not found in database.
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'DBget'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing DBget("SIP/9627932-f491", "AMPUSERCIDNAME=AMPUSER//cidname") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname
Jan  7 15:43:55 DEBUG[15661] db.c: Unable to find key '/cidname' in family 'AMPUSER'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- DBget: Value not found in database.
Jan  7 15:43:55 DEBUG[15661] pbx.c: Expression result is '1'
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'GotoIf'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing GotoIf("SIP/9627932-f491", "1?5") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Goto (macro-user-callerid,s,5)
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'NoOp'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing NoOp("SIP/9627932-f491", "Using CallerID "069915324714" <9627932>") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Macro'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Macro("SIP/9627932-f491", "record-enable|9627932|OUT") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Function result is '0'
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'GotoIf'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing GotoIf("SIP/9627932-f491", "0 > 0?2:4") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Goto (macro-record-enable,s,4)
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'AGI'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing AGI("SIP/9627932-f491", "recordingcheck|20070107-154355|asterisk-14485-1168181035.0") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
Jan  7 15:43:55 DEBUG[15662] app_queue.c: Device 'SIP/9627932' changed to state '4' (Invalid) but we don't care because they're not a member of any queue.
Jan  7 15:43:55 DEBUG[15661] db.c: Unable to find key '9627932/recording' in family 'AMPUSER'
Jan  7 15:43:55 VERBOSE[15661] logger.c:   recordingcheck|20070107-154355|asterisk-14485-1168181035.0: Outbound recording not enabled
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- AGI Script recordingcheck completed, returning 0
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'NoOp'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing NoOp("SIP/9627932-f491", "No recording needed") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Macro'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Macro("SIP/9627932-f491", "outbound-callerid|2") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Expression result is '1'
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'GotoIf'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing GotoIf("SIP/9627932-f491", "1?3") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Goto (macro-outbound-callerid,s,3)
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'DBget'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing DBget("SIP/9627932-f491", "USEROUTCID=AMPUSER/9627932/outboundcid") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- DBget: varname=USEROUTCID, family=AMPUSER, key=9627932/outboundcid
Jan  7 15:43:55 DEBUG[15661] db.c: Unable to find key '9627932/outboundcid' in family 'AMPUSER'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- DBget: Value not found in database.
Jan  7 15:43:55 DEBUG[15661] pbx.c: Expression result is '1'
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'GotoIf'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing GotoIf("SIP/9627932-f491", "1?6") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Goto (macro-outbound-callerid,s,6)
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'NoOp'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing NoOp("SIP/9627932-f491", "CallerID set to "069915324714" <9627932>") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'SetGroup'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing SetGroup("SIP/9627932-f491", "OUT_2") in new stack
Jan  7 15:43:55 WARNING[15661] app_groupcount.c: The SetGroup application has been deprecated, please use the GROUP() function.
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'CheckGroup'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing CheckGroup("SIP/9627932-f491", "") in new stack
Jan  7 15:43:55 WARNING[15661] app_groupcount.c: The CheckGroup application has been deprecated, please use a combination of the GotoIf application and the GROUP_COUNT() function.
Jan  7 15:43:55 WARNING[15661] app_groupcount.c: CheckGroup requires an argument(max[@category][|options])
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'SetVar'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing SetVar("SIP/9627932-f491", "DIAL_NUMBER=627932") in new stack
Jan  7 15:43:55 WARNING[15661] pbx.c: SetVar is deprecated, please use Set instead.
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'SetVar'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing SetVar("SIP/9627932-f491", "DIAL_TRUNK=2") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'AGI'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing AGI("SIP/9627932-f491", "fixlocalprefix") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- AGI Script fixlocalprefix completed, returning 0
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'SetVar'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing SetVar("SIP/9627932-f491", "OUTNUM=627932") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Cut'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Cut("SIP/9627932-f491", "custom=OUT_2|:|1") in new stack
Jan  7 15:43:55 WARNING[15661] app_cut.c: The application Cut is deprecated.  Please use the CUT() function instead.
Jan  7 15:43:55 WARNING[15661] ast_expr2.y: non-numeric argument
Jan  7 15:43:55 DEBUG[15661] pbx.c: Expression result is '0'
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'GotoIf'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing GotoIf("SIP/9627932-f491", "0?16") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Not taking any branch
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Dial'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Dial("SIP/9627932-f491", "SIP/sipgate/627932") in new stack
Jan  7 15:43:55 DEBUG[15661] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
Jan  7 15:43:55 DEBUG[15661] chan_sip.c: Setting NAT on RTP to 524288
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-14.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable MACRO_DEPTH.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-13.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable custom.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-12.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable OUTNUM.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-11.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-10.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable DIAL_TRUNK.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-9.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable DIAL_NUMBER.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-8.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-7.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable GROUP.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-6.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable MACRO_PRIORITY.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable MACRO_CONTEXT.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable MACRO_EXTEN.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable ARG1.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-outbound-callerid-s-6.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-outbound-callerid-s-4.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable DBGETSTATUS.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-outbound-callerid-s-3.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-outbound-callerid-s-1.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-5.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable ARG2.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-record-enable-s-5.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-record-enable-s-4.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-record-enable-s-1.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-4.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-user-callerid-s-5.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-user-callerid-s-3.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-user-callerid-s-2.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-user-callerid-s-1.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-3.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-macro-dialout-trunk-s-1.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable ARG3.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable STACK-from-internal-9627932-1.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable SIPCALLID.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable SIPUSERAGENT.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable SIPDOMAIN.
Jan  7 15:43:55 DEBUG[15661] channel.c: Not copying variable SIPURI.
Jan  7 15:43:55 DEBUG[15661] chan_sip.c: Outgoing Call for 627932
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Called sipgate/627932
Jan  7 15:43:55 DEBUG[15661] channel.c: Set channel SIP/sipgate-2324 to read format alaw
Jan  7 15:43:55 DEBUG[15661] channel.c: Set channel SIP/9627932-f491 to write format alaw
Jan  7 15:43:55 DEBUG[15661] channel.c: Set channel SIP/9627932-f491 to read format alaw
Jan  7 15:43:55 DEBUG[15661] channel.c: Set channel SIP/sipgate-2324 to write format alaw
Jan  7 15:43:55 DEBUG[14531] chan_sip.c: Acked pending invite 102
Jan  7 15:43:55 DEBUG[14531] chan_sip.c: Stopping retransmission on '5eadbba35f58de8346e2df7e4868c51d@sipgate.at' of Request 102: Match Found
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- SIP/sipgate-2324 is circuit-busy
Jan  7 15:43:55 DEBUG[15661] channel.c: Hanging up channel 'SIP/sipgate-2324'
Jan  7 15:43:55 DEBUG[15661] chan_sip.c: Hangup call SIP/sipgate-2324, SIP callid 5eadbba35f58de8346e2df7e4868c51d@sipgate.at)
Jan  7 15:43:55 DEBUG[15661] chan_sip.c: update_call_counter(627932) - decrement call limit counter
Jan  7 15:43:55 VERBOSE[15661] logger.c:   == Everyone is busy/congested at this time (1:0/1/0)
Jan  7 15:43:55 DEBUG[15661] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Goto'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Goto("SIP/9627932-f491", "s-CONGESTION|1") in new stack
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Goto (macro-dialout-trunk,s-CONGESTION,1)
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'NoOp'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing NoOp("SIP/9627932-f491", "Dial failed due to CONGESTION") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Macro'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Macro("SIP/9627932-f491", "outisbusy") in new stack
Jan  7 15:43:55 DEBUG[15661] pbx.c: Launching 'Playback'
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Executing Playback("SIP/9627932-f491", "allison7/all-circuits-busy-now") in new stack
Jan  7 15:43:55 DEBUG[15661] chan_sip.c: sip_answer(SIP/9627932-f491)
Jan  7 15:43:55 DEBUG[15669] app_queue.c: Device 'SIP/sipgate' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
Jan  7 15:43:55 DEBUG[15670] app_queue.c: Device 'SIP/9627932' changed to state '4' (Invalid) but we don't care because they're not a member of any queue.
Jan  7 15:43:55 DEBUG[15661] channel.c: Set channel SIP/9627932-f491 to write format gsm
Jan  7 15:43:55 DEBUG[15661] rtp.c: Ooh, format changed from unknown to alaw
Jan  7 15:43:55 VERBOSE[15661] logger.c:     -- Playing 'allison7/all-circuits-busy-now' (language 'en')
Jan  7 15:43:55 DEBUG[14531] chan_sip.c: Stopping retransmission on '203858ad3dfce9881ac7c7155bee5404@217.10.66.71' of Response 102: Match Found
Jan  7 15:43:57 DEBUG[15661] channel.c: Set channel SIP/9627932-f491 to write format alaw
Jan  7 15:43:57 DEBUG[15661] pbx.c: Launching 'Playback'
Jan  7 15:43:57 VERBOSE[15661] logger.c:     -- Executing Playback("SIP/9627932-f491", "allison7/pls-try-call-later") in new stack
Jan  7 15:43:57 DEBUG[15661] channel.c: Set channel SIP/9627932-f491 to write format gsm
Jan  7 15:43:57 VERBOSE[15661] logger.c:     -- Playing 'allison7/pls-try-call-later' (language 'en')
Jan  7 15:43:59 DEBUG[15661] channel.c: Set channel SIP/9627932-f491 to write format alaw
Jan  7 15:43:59 DEBUG[15661] pbx.c: Launching 'Macro'
Jan  7 15:43:59 VERBOSE[15661] logger.c:     -- Executing Macro("SIP/9627932-f491", "hangupcall") in new stack
Jan  7 15:43:59 DEBUG[15661] pbx.c: Launching 'ResetCDR'
Jan  7 15:43:59 VERBOSE[15661] logger.c:     -- Executing ResetCDR("SIP/9627932-f491", "w") in new stack
Jan  7 15:43:59 DEBUG[15661] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Jan  7 15:43:59 DEBUG[15661] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2007-01-07 15:43:55','\"069915324714\" <9627932>','9627932','9627932','from-internal', 'SIP/9627932-f491','SIP/sipgate-2324','ResetCDR','w',4,4,'ANSWERED',3,'')
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '"069915324714" <9627932>'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '9627932'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '9627932'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is 'from-internal'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is 'SIP/9627932-f491'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is 'SIP/sipgate-2324'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is 'ResetCDR'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is 'w'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '2007-01-07 15:43:55'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '2007-01-07 15:43:55'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '2007-01-07 15:43:59'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '4'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '4'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is 'ANSWERED'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is 'DOCUMENTATION'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '(null)'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is 'asterisk-14485-1168181035.0'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Function result is '(null)'
Jan  7 15:43:59 DEBUG[15661] pbx.c: Launching 'NoCDR'
Jan  7 15:43:59 VERBOSE[15661] logger.c:     -- Executing NoCDR("SIP/9627932-f491", "") in new stack
Jan  7 15:43:59 WARNING[15661] cdr.c: CDR on channel 'SIP/9627932-f491' not posted
Jan  7 15:43:59 WARNING[15661] cdr.c: CDR on channel 'SIP/9627932-f491' lacks end
Jan  7 15:43:59 DEBUG[15661] pbx.c: Launching 'Wait'
Jan  7 15:43:59 VERBOSE[15661] logger.c:     -- Executing Wait("SIP/9627932-f491", "5") in new stack
Jan  7 15:44:04 DEBUG[15661] pbx.c: Launching 'Hangup'
Jan  7 15:44:04 VERBOSE[15661] logger.c:     -- Executing Hangup("SIP/9627932-f491", "") in new stack
Jan  7 15:44:04 DEBUG[15661] app_macro.c: Spawn extension (macro-hangupcall,s,4) exited non-zero on 'SIP/9627932-f491' in macro 'hangupcall'
Jan  7 15:44:04 DEBUG[15661] app_macro.c: Spawn extension (macro-hangupcall,s,4) exited non-zero on 'SIP/9627932-f491' in macro 'outisbusy'
Jan  7 15:44:04 DEBUG[15661] pbx.c: Spawn extension (macro-hangupcall,s,4) exited non-zero on 'SIP/9627932-f491'
Jan  7 15:44:04 DEBUG[15661] channel.c: Hanging up channel 'SIP/9627932-f491'
Jan  7 15:44:04 DEBUG[15661] chan_sip.c: Hangup call SIP/9627932-f491, SIP callid 203858ad3dfce9881ac7c7155bee5404@217.10.66.71)
Jan  7 15:44:04 DEBUG[15661] chan_sip.c: update_call_counter(9627932) - decrement call limit counter
Jan  7 15:44:04 DEBUG[15985] app_queue.c: Device 'SIP/9627932' changed to state '4' (Invalid) but we don't care because they're not a member of any queue.
Jan  7 15:44:04 DEBUG[14531] chan_sip.c: Stopping retransmission on '203858ad3dfce9881ac7c7155bee5404@217.10.66.71' of Request 102: Match Found
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