[Asterisk-Users] T1 - incomplete calls
Hi people, I think our problem was the (30) seg, we extend the time and we think is already resolved, Thanks for your cooperation, We will testing and if we find any other problem, we will send another message. João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
Was this an exmple of your incomplete calls? From the trace it appears that you issued the disconnect while the call was in process. On Jul 22, 2005, at 9:32 AM, JOAO CARLOS MOURA wrote: My debug Thank you for help. Verbosity is at least 5 -- Accepting AUTHENTICATED call from > requested format = g729, > requested prefs = (), > actual format = gsm, > host prefs = (gsm), > priority = mine -- Executing AbsoluteTimeout("IAX2/[EMAIL PROTECTED]", "3600") in new stack -- Set Absolute Timeout to 3600 -- Executing SetCallerID("IAX2/[EMAIL PROTECTED]", "9545569050") in new stack -- Executing Ringing("IAX2/[EMAIL PROTECTED]", "") in new stack -- Executing Dial("IAX2/[EMAIL PROTECTED]", "ZAP/ g1/0115491140583282|60|tr") in new stack -- Making new call for cr 42038 -- Requested transfer capability: 0x00 - SPEECH > Protocol Discriminator: Q.931 (8) len=52 > Call Ref: len= 2 (reference 9270/0x2436) (Originator) > Message type: SETUP (5) > [04 03 80 90 a2] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) > Ext: 1 User information layer 1: u- Law (34) > [18 03 a9 83 81] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 >ChanSel: Reserved > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 1 ] > [1e 02 80 83] > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) > Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] > [6c 0c 21 81 39 35 34 35 35 36 39 30 35 30] > Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) > Presentation: Presentation permitted, user number passed network screening (1) '9545569050' ] > [70 11 a1 30 31 31 35 34 39 31 31 34 30 35 38 33 32 38 32] > Called Number (len=19) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0115491140583282' ] -- Called g1/0115491140583282 < Protocol Discriminator: Q.931 (8) len=10 < Call Ref: len= 2 (reference 9270/0x2436) (Terminator) < Message type: CALL PROCEEDING (2) < [18 03 a9 83 81] < Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 <ChanSel: Reserved < Ext: 1 Coding: 0 Number Specified Channel Type: 3 < Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) < Protocol Discriminator: Q.931 (8) len=9 < Call Ref: len= 2 (reference 9270/0x2436) (Terminator) < Message type: PROGRESS (3) < [1e 02 8a 81] < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) < Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] -- Processing IE 30 (cs0, Progress Indicator) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Outgoing call Proceeding, peerstate Incoming Call Proceeding > Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 9270/0x2436) (Originator) > Message type: DISCONNECT (69) > [08 02 81 90] > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == Spawn extension (qvox, 0115491140583282, 4) exited non-zero on 'IAX2/[EMAIL PROTECTED]' -- Hungup 'IAX2/[EMAIL PROTECTED]' < Protocol Discriminator: Q.931 (8) len=5 < Call Ref: len= 2 (reference 9270/0x2436) (Terminator) < Message type: RELEASE (77) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request > Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 9270/0x2436) (Originator) > Message type: RELEASE COMPLETE (90) > [08 02 81 90] > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent
Re: [Asterisk-Users] T1 - incomplete calls
It would be helpful to capture a complete ISDN call setup. On the cli type "pri debug span 1" Then place a call and turn off debug with "pri no debug span 1" You will then have a complete listing of the signalling between your co and your * for this time period. Good Luck On Jul 22, 2005, at 9:34 AM, JOAO CARLOS MOURA wrote: pri show span 1 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 thks - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 11:33 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 21, 2005 4:14 PM Subject: RE: [Asterisk-Users] T1 - incomplete calls Incomplete meaning "never connected" or "connected then disconnected abruptly?" Are the calls inbound or outbound? All calls or just some calls? If just some, about what percentage are problem calls? Try setting Switchtype = 5ess, 4ess, etc. Let me know what you notice, if anything is different. Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Thursday, July 21, 2005 17:56 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel => 1-23 Thank you João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
My debugThank you for help. Verbosity is at least 5 -- Accepting AUTHENTICATED call from > requested format = g729, > requested prefs = (), > actual format = gsm, > host prefs = (gsm), > priority = mine -- Executing AbsoluteTimeout("IAX2/[EMAIL PROTECTED]", "3600") in new stack -- Set Absolute Timeout to 3600 -- Executing SetCallerID("IAX2/[EMAIL PROTECTED]", "9545569050") in new stack -- Executing Ringing("IAX2/[EMAIL PROTECTED]", "") in new stack -- Executing Dial("IAX2/[EMAIL PROTECTED]", "ZAP/g1/0115491140583282|60|tr") in new stack-- Making new call for cr 42038 -- Requested transfer capability: 0x00 - SPEECH> Protocol Discriminator: Q.931 (8) len=52> Call Ref: len= 2 (reference 9270/0x2436) (Originator)> Message type: SETUP (5)> [04 03 80 90 a2]> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)> Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)> Ext: 1 User information layer 1: u-Law (34)> [18 03 a9 83 81]> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0> ChanSel: Reserved> Ext: 1 Coding: 0 Number Specified Channel Type: 3> Ext: 1 Channel: 1 ]> [1e 02 80 83]> Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0)> Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ]> [6c 0c 21 81 39 35 34 35 35 36 39 30 35 30]> Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)> Presentation: Presentation permitted, user number passed network screening (1) '9545569050' ]> [70 11 a1 30 31 31 35 34 39 31 31 34 30 35 38 33 32 38 32]> Called Number (len=19) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0115491140583282' ] -- Called g1/0115491140583282< Protocol Discriminator: Q.931 (8) len=10< Call Ref: len= 2 (reference 9270/0x2436) (Terminator)< Message type: CALL PROCEEDING (2)< [18 03 a9 83 81]< Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0< ChanSel: Reserved< Ext: 1 Coding: 0 Number Specified Channel Type: 3< Ext: 1 Channel: 1 ]-- Processing IE 24 (cs0, Channel Identification)< Protocol Discriminator: Q.931 (8) len=9< Call Ref: len= 2 (reference 9270/0x2436) (Terminator)< Message type: PROGRESS (3)< [1e 02 8a 81]< Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10)< Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ]-- Processing IE 30 (cs0, Progress Indicator)NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Outgoing call Proceeding, peerstate Incoming Call Proceeding> Protocol Discriminator: Q.931 (8) len=9> Call Ref: len= 2 (reference 9270/0x2436) (Originator)> Message type: DISCONNECT (69)> [08 02 81 90]> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1)> Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == Spawn extension (qvox, 0115491140583282, 4) exited non-zero on 'IAX2/[EMAIL PROTECTED]' -- Hungup 'IAX2/[EMAIL PROTECTED]'< Protocol Discriminator: Q.931 (8) len=5< Call Ref: len= 2 (reference 9270/0x2436) (Terminator)< Message type: RELEASE (77)NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request> Protocol Discriminator: Q.931 (8) len=9> Call Ref: len= 2 (reference 9270/0x2436) (Originator)> Message type: RELEASE COMPLETE (90)> [08 02 81 90]> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1)> Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 11:33 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing
Re: [Asterisk-Users] T1 - incomplete calls
pri show span 1 Primary D-channel: 24Status: Provisioned, Up, ActiveSwitchtype: National ISDNType: CPEWindow Length: 0/7Sentrej: 0SolicitFbit: 0Retrans: 0Busy: 0Overlap Dial: 0T200 Timer: 1000T203 Timer: 1T305 Timer: 3T308 Timer: 4000T313 Timer: 4000N200 Counter: 3 thks - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 11:33 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 21, 2005 4:14 PM Subject: RE: [Asterisk-Users] T1 - incomplete calls Incomplete meaning "never connected" or "connected then disconnected abruptly?" Are the calls inbound or outbound? All calls or just some calls? If just some, about what percentage are problem calls? Try setting Switchtype = 5ess, 4ess, etc. Let me know what you notice, if anything is different. Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURASent: Thursday, July 21, 2005 17:56To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration?Here our configuration Zaptel.conf span=1,1,0,esf,b8zsbchan=1-23 dchan=24 defaultzone=usloadzone=us === Zapata.conf [channels]language=ensignalling=pri_cpeswitchtype=nationalechocancel=yesechocancelwhenbridged=yesechotraining=200 ; Asterisk trains to the beginning of the call, number is in millisecondscallerid=000busydetect=yesbusycount=5group=1callgroup=1pickupgroup=1callreturn=yescontext=pstnchannel => 1-23 Thank you João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
Hi there, Our problem is with outgoing calls... And the problem is some calls do not complete...the asterisk show the ring...but doesnt complete some calls...we dont have dropped calls... thank you - Original Message - From: "Paul Belanger" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, July 21, 2005 6:45 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls Are your problems with incoming calls to your PRI or outgoing calls? Are the calls being dropped or just not hitting your asterisk box? PB JOAO CARLOS MOURA wrote: Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel => 1-23 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
I am observing. The problem is in the outbound calls. Some are not completed. Thank you - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 11:33 PM Subject: Re: [Asterisk-Users] T1 - incomplete calls pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 21, 2005 4:14 PM Subject: RE: [Asterisk-Users] T1 - incomplete calls Incomplete meaning "never connected" or "connected then disconnected abruptly?" Are the calls inbound or outbound? All calls or just some calls? If just some, about what percentage are problem calls? Try setting Switchtype = 5ess, 4ess, etc. Let me know what you notice, if anything is different. Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Thursday, July 21, 2005 17:56 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel => 1-23 Thank you João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
pri debug span 1 output? - Original Message - From: Thomas Christie To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 21, 2005 4:14 PM Subject: RE: [Asterisk-Users] T1 - incomplete calls Incomplete meaning "never connected" or "connected then disconnected abruptly?" Are the calls inbound or outbound? All calls or just some calls? If just some, about what percentage are problem calls? Try setting Switchtype = 5ess, 4ess, etc. Let me know what you notice, if anything is different. Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURASent: Thursday, July 21, 2005 17:56To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration?Here our configuration Zaptel.conf span=1,1,0,esf,b8zsbchan=1-23 dchan=24 defaultzone=usloadzone=us === Zapata.conf [channels]language=ensignalling=pri_cpeswitchtype=nationalechocancel=yesechocancelwhenbridged=yesechotraining=200 ; Asterisk trains to the beginning of the call, number is in millisecondscallerid=000busydetect=yesbusycount=5group=1callgroup=1pickupgroup=1callreturn=yescontext=pstnchannel => 1-23 Thank you João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 - incomplete calls
Incomplete meaning "never connected" or "connected then disconnected abruptly?" Are the calls inbound or outbound? All calls or just some calls? If just some, about what percentage are problem calls? Try setting Switchtype = 5ess, 4ess, etc. Let me know what you notice, if anything is different. Thomas Christie There are 10 types of people in the world: those who understand binary and those who don't. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURASent: Thursday, July 21, 2005 17:56To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration?Here our configuration Zaptel.conf span=1,1,0,esf,b8zsbchan=1-23 dchan=24 defaultzone=usloadzone=us === Zapata.conf [channels]language=ensignalling=pri_cpeswitchtype=nationalechocancel=yesechocancelwhenbridged=yesechotraining=200 ; Asterisk trains to the beginning of the call, number is in millisecondscallerid=000busydetect=yesbusycount=5group=1callgroup=1pickupgroup=1callreturn=yescontext=pstnchannel => 1-23 Thank you João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 - incomplete calls
Are your problems with incoming calls to your PRI or outgoing calls? Are the calls being dropped or just not hitting your asterisk box? PB JOAO CARLOS MOURA wrote: >Hi All >Help. > >We are using a T1 with Paetec Telecom in the Miami area, with a Digium card >into our Asterisk >software, and in the last week we are experience a large quantities of >incomplete calls, even local and international, what do you think, >the problem are into the T1 or into our configuration? >Here our configuration > > >Zaptel.conf >span=1,1,0,esf,b8zs >bchan=1-23 >dchan=24 > >defaultzone=us >loadzone=us > >=== > >Zapata.conf >[channels] >language=en >signalling=pri_cpe >switchtype=national >echocancel=yes >echocancelwhenbridged=yes >echotraining=200 ; Asterisk trains to the beginning of the call, number is in >milliseconds >callerid=000 >busydetect=yes >busycount=5 >group=1 >callgroup=1 >pickupgroup=1 >callreturn=yes >context=pstn >channel => 1-23 > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 - incomplete calls
Do some debug on the calls and see what you get…. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Thursday, July 21, 2005 2:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel => 1-23 Thank you João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 - incomplete calls
Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration?Here our configuration Zaptel.conf span=1,1,0,esf,b8zsbchan=1-23 dchan=24 defaultzone=usloadzone=us === Zapata.conf [channels]language=ensignalling=pri_cpeswitchtype=nationalechocancel=yesechocancelwhenbridged=yesechotraining=200 ; Asterisk trains to the beginning of the call, number is in millisecondscallerid=000busydetect=yesbusycount=5group=1callgroup=1pickupgroup=1callreturn=yescontext=pstnchannel => 1-23 Thank you João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 - incomplete calls
I have the same setup. With Paetec and in Miami also.. You can call me to discuss if you like. 305-503-3000 ext 122 Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Wednesday, July 20, 2005 1:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel => 1-23 Thank you João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 - incomplete calls
Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration?Here our configuration Zaptel.conf span=1,1,0,esf,b8zsbchan=1-23 dchan=24 defaultzone=usloadzone=us === Zapata.conf [channels]language=ensignalling=pri_cpeswitchtype=nationalechocancel=yesechocancelwhenbridged=yesechotraining=200 ; Asterisk trains to the beginning of the call, number is in millisecondscallerid=000busydetect=yesbusycount=5group=1callgroup=1pickupgroup=1callreturn=yescontext=pstnchannel => 1-23 Thank you João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users