Difference between revisions of "Sipgate"
From LinuxMCE
| Line 88: | Line 88: | ||
nat=yes | nat=yes | ||
qualify=no | qualify=no | ||
| − | secret= | + | secret=XXXXX |
type=user | type=user | ||
user=9627932 | user=9627932 | ||
| Line 118: | Line 118: | ||
-- Executing GotoIf("SIP/9627932-96e1", "1?5") in new stack | -- Executing GotoIf("SIP/9627932-96e1", "1?5") in new stack | ||
-- Goto (macro-user-callerid,s,5) | -- Goto (macro-user-callerid,s,5) | ||
| − | -- Executing NoOp("SIP/9627932-96e1", "Using CallerID " | + | -- 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 Macro("SIP/9627932-96e1", "record-enable|9627932|OUT") in new stack | ||
-- Executing GotoIf("SIP/9627932-96e1", "0 > 0?2:4") in new stack | -- Executing GotoIf("SIP/9627932-96e1", "0 > 0?2:4") in new stack | ||
| Line 135: | Line 135: | ||
-- Executing GotoIf("SIP/9627932-96e1", "1?6") in new stack | -- Executing GotoIf("SIP/9627932-96e1", "1?6") in new stack | ||
-- Goto (macro-outbound-callerid,s,6) | -- Goto (macro-outbound-callerid,s,6) | ||
| − | -- Executing NoOp("SIP/9627932-96e1", "CallerID set to " | + | -- Executing NoOp("SIP/9627932-96e1", "CallerID set to "0699153xxxxx" <9627932>") in new stack |
-- Executing SetGroup("SIP/9627932-96e1", "OUT_2") in new stack | -- Executing SetGroup("SIP/9627932-96e1", "OUT_2") in new stack | ||
-- Executing CheckGroup("SIP/9627932-96e1", "") in new stack | -- Executing CheckGroup("SIP/9627932-96e1", "") in new stack | ||
Revision as of 11:56, 6 January 2007
| 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. |
This Page is under construction!
1. Step Install Pluto Core/Hybrid
2. Step Add a Media Director
After the this i a running Plutosystem with one Media Director and one Windows XP Orbiter.
3. Step Add Phone Lines
In Pluto 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 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! 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
