Sipgate - Revision history

Zaerc at 15:16, 18 October 2010

2010-10-18T15:16:56Z

@@ Line 1: / Line 1: @@
 http://www.sipgate.co.uk
@@ Line 374: / Line 379: @@
 Jan  7 15:44:04 DEBUG[14531] chan_sip.c: Stopping retransmission on '203858ad3dfce9881ac7c7155bee5404@217.10.66.71' of Request 102: Match Found
 </pre>

Rwilson131: clean up unneeded steps.

2007-10-14T05:09:39Z

clean up unneeded steps.

←Older revision		Revision as of 05:09, 14 October 2007
Line 1:		Line 1:
	~~<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~~.		http://www.sipgate.co.uk

	~~'''1. Step Install LinuxMCE Core/Hybrid'''~~		==Automatic Setup==
			The setup is automatic from the House Setup Wizard using the following script

			/usr/pluto/bin/create_amp_inphonex.pl

	~~'''2. Step Add a Media Director'''~~		==Manual Setup==


	~~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.		In LinuxMCE Web Admin go to Devices -> Phone Lines.
Line 27:		Line 24:
	</pre>		</pre>

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

	~~'''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!		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!
Line 91:		Line 81:
	username=9627932		username=9627932
	</pre>		</pre>
			==TroubleShooting==
	~~'''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 call my sipgate number from my cell phone but i only here a female voice who tells me that all circuits are busy now.

Rwilson131: Asterisk sipgate moved to Sipgate

2007-10-14T05:05:48Z

Asterisk sipgate moved to Sipgate

←Older revision	Revision as of 05:05, 14 October 2007
(No difference)

Chewi at 07:27, 12 October 2007

2007-10-12T07:27:49Z

←Older revision		Revision as of 07:27, 12 October 2007
Line 386:		Line 386:
	</pre>		</pre>
	[[Category: Tutorials]]		[[Category: Tutorials]]
			[[Category: Telecom]]
			[[Category: Phone Lines]]

Chewi at 10:34, 19 September 2007

2007-09-19T10:34:57Z

←Older revision		Revision as of 10:34, 19 September 2007
Line 97:		Line 97:
	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 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).		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:		Here is the log file from asterisk when i make a call from my cell phone:

Rwilson131 at 01:36, 6 September 2007

2007-09-06T01:36:23Z

←Older revision		Revision as of 01:36, 6 September 2007
Line 379:		Line 379:
	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>
			[[Category: Tutorials]]

Juliet at 09:10, 16 March 2007

2007-03-16T09:10:41Z

←Older revision		Revision as of 09:10, 16 March 2007
Line 1:		Line 1:
	<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.		<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!'''~~		'''1. Step Install LinuxMCE Core/Hybrid'''

	'''1. Step Install ~~Pluto~~ Core/Hybrid'''


Line 9:		Line 7:


	After the this i a running ~~Plutosystem~~ with one Media Director and one Windows XP Orbiter.		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.		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!		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:

Wikiadmin at 14:04, 7 January 2007

2007-01-07T14:04:37Z

Show changes

Wikiadmin at 10:56, 6 January 2007

2007-01-06T10:56:54Z

←Older revision		Revision as of 10:56, 6 January 2007
Line 88:		Line 88:
	nat=yes		nat=yes
	qualify=no		qualify=no
	secret=~~FHwsed~~		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 "~~069915324714~~" <9627932>") in new stack		-- 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 "~~069915324714~~" <9627932>") in new stack		-- 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

Wikiadmin at 10:55, 6 January 2007

2007-01-06T10:55:12Z

New page

<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.

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

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

<pre>
Type Data Password Username Host Phone number
SIP sipgate XXXXXXX 9627932 sipgate.at 0720726742
</pre>

''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:
<pre>
112
411
911
9|.
</pre>

Outgoing Settings:
<pre>
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
</pre>

Incoming Settings

<pre>
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=FHwsed
type=user
user=9627932
username=9627932
</pre>

'''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:

<pre>
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 "069915324714" <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 "069915324714" <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
</pre>