Difference between revisions of "Sipgate"

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<table width="100%"> <tr><td bgcolor="#FFCFCF">This page was written by Pluto and imported with their permission when LinuxMCE branched off in February, 2007.  In general any information should apply to LinuxMCE.  However, this page should be edited to reflect changes to LinuxMCE and remove old references to Pluto.</td></tr> </table>This should be a short How-To for integrating sipgate with Pluto-Asterisk. It describes the steps what i have done and where the problems are.
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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.
  
'''This Page is under construction!'''
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'''1. Step Install LinuxMCE Core/Hybrid'''
 
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'''1. Step Install Pluto Core/Hybrid'''
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Line 9: Line 7:
  
  
After the this i a running Plutosystem with one Media Director and one Windows XP Orbiter.  
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After the this i a running LinuxMCE system with one Media Director and one Windows XP Orbiter.  
  
 
'''3. Step Add Phone Lines'''
 
'''3. Step Add Phone Lines'''
  
In Pluto Web Admin go to Devices -> Phone Lines.  
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In LinuxMCE Web Admin go to Devices -> Phone Lines.  
  
 
I don't make changes in the first box:
 
I don't make changes in the first box:
Line 38: Line 36:
 
'''5. Step AMP Configuration'''
 
'''5. Step AMP Configuration'''
  
In Pluto Web Admin go to Advanced -> Configuration -> Phones Setup -> Setup. Pluto 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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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:
 
Here are my configurations:
  

Revision as of 09:10, 16 March 2007

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. 1. Step Install LinuxMCE Core/Hybrid2. Step Add a Media Director After the this i a running LinuxMCE system with one Media Director and one Windows XP Orbiter. 3. Step Add Phone Lines 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

I don't know if i have to change the settings in the first box!?!?!?

4. Step Reload the router

Don't know if the reload is important at this point, but i think it's better to do it.


5. Step 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

Problems

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).

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