News & Reviews
More How-To's & Tips More News
More Reviews Device Configuration Tools
No account yet? Create one
Forgot your Username or Password?

Welcome to the Voxilla VoIP Forum.

Voxilla has been a trusted source for accurate, up-to-date information on the IP Communications industry since 2002. A dedicated staff of reporters and engineers produce feature articles and product reviews to keep industry watchers abreast of the people, companies, and trends driving a fast moving market.

You are currently viewing our boards as a guest which gives you limited access to view most discussions and access our other features. By joining our free community you will have access to post topics, communicate privately with other members (PM), respond to polls, upload content and access many other special features. Registration is fast, simple and absolutely free so please, join our community today!

If you have any problems with the registration process or your account login, please contact contact us.





Closed Thread
 
LinkBack Thread Tools Rate Thread Display Modes
  #1 (permalink)  
Old March 31st, 2005, 09:19 PM
pat_delaney pat_delaney is offline
Junior Member
 
Join Date: Mar 2005
Posts: 3
pat_delaney
Default Asterisk & Definity G3 PBX integration

I have setup a 4 analog line hunt grout in our Lucent G3 switch. Those 4 lines are connected to my asterisk server via 4 port TDM card. In looking at the * CLI is see that some of the ZAP channels will mysteriously startup as if a call was coming in. A call was forwrded to my softphone and when I answered it, it should like a wave-off tone. I suspect that * may not be hanging up the line correctly and the G3 gets confused. The analog lines are coming off the G3, would * consider them kewl_start loop_start, or ground_start?

Any thoughts

Here is a snipit from the log:


-- Starting simple switch on 'Zap/1-1'
-- Executing GotoIf("Zap/1-1", "0?from-pstn-reghours|s|1:") in new stack
-- Executing GotoIf("Zap/1-1", "0?from-pstn-afthours|s|1:") in new stack
-- Executing GotoIfTime("Zap/1-1", "7:55-17:05|mon-fri|*|*?from-pstn-reghours|s|1:") in new stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf("Zap/1-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack
-- Goto (from-pstn-reghours,s,2)
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing Wait("Zap/1-1", "1") in new stack
-- Executing SetVar("Zap/1-1", "intype=EXT-2014") in new stack
-- Executing Cut("Zap/1-1", "intype=intype|-|1") in new stack
-- Executing GotoIf("Zap/1-1", "1?7:9") in new stack
-- Goto (from-pstn-reghours,s,7)
-- Executing Wait("Zap/1-1", "3") in new stack
-- Executing Goto("Zap/1-1", "ext-local|2014|1") in new stack
-- Goto (ext-local,2014,1)
-- Executing Macro("Zap/1-1", "exten-vm|novm|2014") in new stack
-- Executing GotoIf("Zap/1-1", "1?novm|1") in new stack
-- Goto (macro-exten-vm,novm,1)
-- Executing Macro("Zap/1-1", "dial|120|tr|2014") in new stack
-- Executing AGI("Zap/1-1", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: priority = 1
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode =
-- dialparties.agi: uniqueid = 1112303666.40
-- dialparties.agi: channel = Zap/1-1
-- dialparties.agi: callerid = <0>
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = Zap
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = unknown
dialparties.agi: Caller ID name is '' number is '0'
-- dialparties.agi: Added extension 2014 to extension map
-- dialparties.agi: Extension 2014 cf is disabled
-- dialparties.agi: Extension 2014 do not disturb is disabled

[top] Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Extension 2014 has call waiting disabled


Manager 'admin' logged off from 127.0.0.1
-- dialparties.agi: DbDel CALLTRACE/2014 - Caller ID is not defined
dialparties.agi: About to execute Dial(SIP/2014|120|tr)
-- AGI Script Executing Application: (Dial) Options: (SIP/2014|120|tr)
-- Called 2014
-- SIP/2014-e39f is ringing
-- SIP/2014-e39f answered Zap/1-1
dialparties.agi: Dial return value was -1 and dialstring was SIP/2014|120|tr
dialparties.agi: Setting Priority to 22 from 2
-- AGI Script dialparties.agi completed, returning 0
-- Executing Macro("Zap/1-1", "hangupcall") in new stack
-- Executing ResetCDR("Zap/1-1", "w") in new stack
-- Executing NoCDR("Zap/1-1", "") in new stack
-- Executing Wait("Zap/1-1", "5") in new stack
-- Executing Hangup("Zap/1-1", "") in new stack
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/2-1'
-- Executing GotoIf("Zap/2-1", "0?from-pstn-reghours|s|1:") in new stack
-- Executing GotoIf("Zap/2-1", "0?from-pstn-afthours|s|1:") in new stack
-- Executing GotoIfTime("Zap/2-1", "7:55-17:05|mon-fri|*|*?from-pstn-reghours|s|1:") in new stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf("Zap/2-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack
-- Goto (from-pstn-reghours,s,2)
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Wait("Zap/2-1", "1") in new stack
-- Executing SetVar("Zap/2-1", "intype=EXT-2014") in new stack
-- Executing Cut("Zap/2-1", "intype=intype|-|1") in new stack
-- Executing GotoIf("Zap/2-1", "1?7:9") in new stack
-- Goto (from-pstn-reghours,s,7)
-- Executing Wait("Zap/2-1", "3") in new stack
-- Executing Goto("Zap/2-1", "ext-local|2014|1") in new stack
-- Goto (ext-local,2014,1)
-- Executing Macro("Zap/2-1", "exten-vm|novm|2014") in new stack
-- Executing GotoIf("Zap/2-1", "1?novm|1") in new stack
-- Goto (macro-exten-vm,novm,1)
-- Executing Macro("Zap/2-1", "dial|120|tr|2014") in new stack
-- Executing AGI("Zap/2-1", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: priority = 1
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode =
-- dialparties.agi: uniqueid = 1112303737.42
-- dialparties.agi: channel = Zap/2-1
-- dialparties.agi: callerid = <0>
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = Zap
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = unknown
dialparties.agi: Caller ID name is '' number is '0'
-- dialparties.agi: Added extension 2014 to extension map
-- dialparties.agi: Extension 2014 cf is disabled
-- dialparties.agi: Extension 2014 do not disturb is disabled

[top] Manager 'admin' logged on from 127.0.0.1


Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 2014 has call waiting disabled
-- dialparties.agi: DbDel CALLTRACE/2014 - Caller ID is not defined
dialparties.agi: About to execute Dial(SIP/2014|120|tr)
-- AGI Script Executing Application: (Dial) Options: (SIP/2014|120|tr)
-- Called 2014
-- SIP/2014-1948 is ringing
-- SIP/2014-1948 answered Zap/2-1
dialparties.agi: Dial return value was -1 and dialstring was SIP/2014|120|tr
dialparties.agi: Setting Priority to 22 from 2
-- AGI Script dialparties.agi completed, returning 0
-- Executing Macro("Zap/2-1", "hangupcall") in new stack
-- Executing ResetCDR("Zap/2-1", "w") in new stack
-- Executing NoCDR("Zap/2-1", "") in new stack
-- Executing Wait("Zap/2-1", "5") in new stack
-- Executing Hangup("Zap/2-1", "") in new stack
-- Hungup 'Zap/2-1'
asterisk1*CLI>
Digg this Post!Add Post to del.icio.usBookmark Post in TechnoratiFurl this Post!
  #2 (permalink)  
Old April 1st, 2005, 12:22 AM
mberlant's Avatar
mberlant mberlant is offline
Senior Member
 
Join Date: Aug 2004
Location: USA or Japan
Posts: 5,013
mberlant is an unknown quantity at this point
Default RE: Asterisk & Definity G3 PBX integration

Typically, CO trunks in this situation are configured as loop start. Loop start is the way an ordinary phone connects with a switch. When you pick up the phone it closes an electrical loop, which tells the switch to deliver a Dial Tone.

Unfortunately, in the PABX world this opens up a security hole. Have you ever made a call, have the other guy hang up the phone and you don't hang up? With many switches you will get a new Dial Tone and be able to make another call. On a private phone this isn't a problem, but as a PABX trunk it's enormous. An unscrupulous employee could pick up the phone, call a friend on a local number, have the friend hang up, wait for the new Dial Tone and place a Long Distance or international call. This new call would not be checked against the user's permissions (maybe he isn't permitted international dialing) and would not be captured on the PABX's billing system.

For this reason Ground Start was created. Ground Start trunks will not provide Dial Tone until two things happen simultaneously. The loop needs to be closed and one side of the loop needs to be momentarily shorted to ground. This prevents the problem, since the PABX will only short the loop to ground after it has authorized the placement of the call.

For your purposes it is your choice. It may make your lines more stable to switch to Ground Start. One thing you will need to check is if these particular G3 trunks are Ground Start capable or not.
__________________
Please do not send technical questions via PM.
Please post all questions to the forum.
Digg this Post!Add Post to del.icio.usBookmark Post in TechnoratiFurl this Post!
  #3 (permalink)  
Old April 17th, 2005, 12:06 PM
marner marner is offline
Junior Member
 
Join Date: Jan 2005
Location: Gold Coast, Australia
Posts: 24
marner
Send a message via ICQ to marner
Default RE: Asterisk & Definity G3 PBX integration

If I read it right, you have connected 4 analogue station ports (FXS) from the Lucent into a 4 port TDM card (FXO).

On the Lucent, analogue station ports by default have an automatic Missed Call Indication function turned on in the station form. This function is performed every x mins (usually 5 or 15 minutes). A normal analogue phone does not detect this function, but the TDM card, and some panasonic and uniden cordless phone do. On the cordless phone, it appears as a phantom ring, on a TDM card, it would possibly appear as a phantom call. It is similiar to a MWI (Message Waiting Indicator) signal to turn on MW light on phone, but is a feature related to indicate Missed Calls (the PBX keeps track of last x number of Missed Calls, along with the CLID number) to the phone.
The feature can be turned off on a station by station basis in the station form.

This 'may' be the cause of your * ZAP channels mysteriously starting up as if a call was coming in.

HTH

marner
Digg this Post!Add Post to del.icio.usBookmark Post in TechnoratiFurl this Post!
  #4 (permalink)  
Old April 26th, 2005, 10:30 PM
patdelaney patdelaney is offline
Junior Member
 
Join Date: Mar 2005
Posts: 20
patdelaney
Default RE: Asterisk & Definity G3 PBX integration

Thanks for your reply. I will check that out. Do you know the name of the "feature" that I should be turning off?

Pat
Digg this Post!Add Post to del.icio.usBookmark Post in TechnoratiFurl this Post!
  #5 (permalink)  
Old April 30th, 2005, 08:01 AM
marner marner is offline
Junior Member
 
Join Date: Jan 2005
Location: Gold Coast, Australia
Posts: 24
marner
Send a message via ICQ to marner
Default RE: Asterisk & Definity G3 PBX integration

Pat,

Message Waiting Indication for External Calls

on the change system-parameters features form (note, you may need a high level password to be able to change this)

y or n, lights the message waiting light on the handset if you have 'missed' any external calls. System remembers the last 15 callers CLID, and if using a display phone, allows you to scroll back thru those missed calls.
Depending on the message waiting type of indication, this can cause the 2500 station to ring to indicate message waiting.

Hope this helps, of coarse it may be totally unrelated, but you gotta try everything

Mark
Digg this Post!Add Post to del.icio.usBookmark Post in TechnoratiFurl this Post!
Old April 30th, 2005, 08:01 AM
Closed Thread


Thread Tools
Display Modes Rate This Thread
Rate This Thread:



Similar Threads for: Asterisk & Definity G3 PBX integration
Thread Thread Starter Forum Replies Last Post
Cisco and Asterisk Integration (Voicemail related) Robert.Bell1 Asterisk Support Forum 0 March 21st, 2006 01:42 AM
asterisk & CTI software integration yaman Asterisk Support Forum 0 January 11th, 2006 08:41 AM
Integration Asterisk - Siemens Legacy david_znsp Asterisk Support Forum 1 October 14th, 2005 03:35 PM
Asterisk & Jive Messenger - IM Integration muppetmaster Asterisk Support Forum 0 October 11th, 2005 06:22 PM
SPA3000: gateway between LAN & PBX? vroem Linksys (Sipura) VoIP Support Forum 1 June 27th, 2004 06:52 AM



All times are GMT. The time now is 08:09 AM.


vBulletin, Copyright ©2000 - 2009, Jelsoft Enterprises Ltd. SEO by vBSEO 3.0.0 ©2007, Crawlability, Inc. Logos and trademarks are the property of Voxilla or their respective owner. All other content © 2003-2007 by Voxilla, Inc.