Re: [asterisk-users] Question about TFTPD server
Check is: very good http://www.it4u2.com/asterisk2.htm#SIPmacaddress http://www.loligo.com/asterisk/cisco/79xx/current/ - Original Message - From: Edwin Lam [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 15, 2006 9:22 PM Subject: Re: [asterisk-users] Question about TFTPD server Christian wrote: I have installed this package onto my Debian and placed the files i want the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to work. Are ther any special settings I should do to this server? Many thanks for all your help, which tftp server package did you installed? make sure your /tftpboot directory and all the files inside is at least readable by everyone. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 407 proxy authentication
in the sip.conf insecure=very canreinvite=yes []'s - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: asterisk-dev@lists.digium.com Sent: Saturday, April 08, 2006 11:41 Subject: [Asterisk-Users] 407 proxy authentication Hello, look at this I can't receive calls from other domains I wish sip:[EMAIL PROTECTED] are forwarded to asterisk however this one spend its time to ask 407 proxy authentication. asterisk 1.2.5 + realtime how can i fix this problem what' wrong ? extension.conf [info] exten = info,1,Answer() exten = info,2,Dial(Sip/84,10) exten = info,3,Dial(Sip/85,10) exten = info,4,Hangup serveur1*CLI sip show user info load serveur1*CLI * Name : info Secret : Not set MD5Secret: Not set Context : info Language : fr AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : ACL : No Codec Order : (g729|ilbc|gsm|ulaw|alaw) harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID's
I need 10 DID's for it those country's NicaraguaEl salvadorCosta RicaPanamaHonduras Thank's João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID's
thank's - Original Message - From: Mike Pollitt To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, February 16, 2006 17:22 Subject: RE: [Asterisk-Users] DID's Wrong list. You want asterisk-biz. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURASent: Friday, 17 February 2006 9:06 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] DID's I need 10 DID's for it those country's NicaraguaEl salvadorCosta RicaPanamaHonduras Thank's João Carlos MouraNiNeTel Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_addon_mysql.so pb
I use the module that this in the [EMAIL PROTECTED] and functions very well. []'s jmoura - Original Message - From: Jonathan k. Creasy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, September 11, 2005 15:00 Subject: RE: [Asterisk-Users] cdr_addon_mysql.so pb Did you confirm that cdr_addon_mysql is indeed built and loading? I had missed that it wasn’t being built (I didn’t have mysql-devel installed) when I first tried to do that a few months ago. -Jonathan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of alexandre zhang Sent: Sunday, September 11, 2005 2:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cdr_addon_mysql.so pb hi I load cdr_addon_mysql.so without error configuration of cdr_mysql.conf [general] dbhost = localhost dbname = recharge dbuser = root dbpass = ast dbport = 3306 dbsock = /var/lib/mysql/mysql.sock But, I get nothing in the table of cdr of my database. Somebody have an idea ? Thanks you for your help best regards DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TE110P reset
Thank you for all Sorry my English Jmoura - Original Message - From: Jason Walker [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, September 10, 2005 21:40 Subject: RE: [Asterisk-Users] TE110P reset You are correct. I did not expand completely and stand corrected. An additional note...we have some Dialogic cards (not associated with *) that do the same thing on PRI. Question - is it somewhat standard to have b chans restart on PRI circuits when not explicitly configured to NOT reset? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, September 10, 2005 5:00 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TE110P reset On Saturday 10 September 2005 19:40, Jason Walker wrote: PRI channels will reset when not in use throughout the day. A reset on a channel should not happen when that channel is in use. This happens all the time on my PRI circuits (TE110P and TE410P). From what I gather, it's somewhat like a handshake for the D chan between the cpe and net sides. Not exactly. Digium's replicating the B channel resets someone noted in a particular situation. It's not required, but it shouldn't hurt. If it's causing trouble you can turn it off with resetinterval=0 in your zapata.conf. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TE110P reset
My TE110P reset some times in the day. E this cause an interruption in the service. How I decide this problem? my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for E1 defaultzone=us loadzone=us my 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 -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 -- B-channel 0/7 successfully restarted on span 1 -- B-channel 0/8 successfully restarted on span 1 -- B-channel 0/9 successfully restarted on span 1 -- B-channel 0/10 successfully restarted on span 1 -- B-channel 0/11 successfully restarted on span 1 -- B-channel 0/12 successfully restarted on span 1 -- B-channel 0/13 successfully restarted on span 1 -- B-channel 0/14 successfully restarted on span 1 -- B-channel 0/15 successfully restarted on span 1 -- B-channel 0/16 successfully restarted on span 1 -- B-channel 0/17 successfully restarted on span 1 -- B-channel 0/18 successfully restarted on span 1 -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/20 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/22 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 Thank you João Carlos Moura ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] chan_sip.c: stale nonce received
I have this error Aug 27 13:31:46 NOTICE[2863] chan_sip.c: stale nonce received from '656720189sip:[EMAIL PROTECTED];user=phone' generated for two equipment hardwired in asterisk. Some friend can help me? Thank's João Carlos Moura ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Erro
Help... I am receiving this message and I do not know as to decide. Jul 27 13:06:47 NOTICE[26766] chan_sip.c: stale nonce received from . My asterisk: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-06-16 16:57:10 Thank you João Carlos Moura ___ 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] E-911
Hi there, Somebody knows a solution E911 for Asterisk? I can implement the E911 with a AGI? We have some customers with a DID in our termination into our Asterisk in different areas... Somebody know how can I send the address? Today if some of those users we have dial 911 the address we will appear will be our address for the PSAP... Some help?? Thanks in advance João Carlos Moura ___ 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
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 asterisk-users@lists.digium.com 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
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
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 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] 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
[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
[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
[Asterisk-Users] Termination South America
We offer termination in: Miami,USA u$s 0.019 Buenos Aires,ARGENTINA u$s 0.019 Fortaleza,BRAZIL u$s 0.029 Check our rates in CHILE Santiago, PARAGUAY Asuncion, URUGUAY Montevideo, Punta del Este, BRAZIL Rio de Janeiro, San Pablo, Goiania, Puerto Alegre. DID's u$s 5.50 each in all ours areas. Joao Carlos Moura [EMAIL PROTECTED] ___ 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] TDM04B - TDM PCI MASTER ABORT
Hi all I have four cards TDM04B with modules FXO. I am using the stabled version of the Asterisk in a Pentium IV with 1GB of RAM and RedHat 9 When I load the modules zaptel and wctdm, I receive the message: TDM PCI Master Abort. How I decide this problem? Thank's Joao Carlos Moura ___ 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] Not register SIP and IAX
Hi all, My Asterisk server is facing some problem that I can´t even find, any registrarion for that, into the error log file. It runs normally for while and suddenly stop registering even IAX and SIP. Acting like that all my softphones and equipments once registered stop working and the only way to start working again is applying a STOP NOW command. Is there anybody there has faced into this problem someday that could help me? Thank´s jmoura ___ 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] SIP ActiveX
I search a ActiveX to develop one softphone SIP with codec G723. Who can help me? Thank´s João Carlos Moura ___ 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] SIP / IAX ActiveX
Tks - Original Message - From: Tim Greiser [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 09, 2005 5:16 PM Subject: Re: [Asterisk-Users] SIP / IAX ActiveX Colin Anderson wrote: Actually, I have an application for a IAX ActiveX control myself. Rather use IAX because of the whole NAT-traversal issue. I found this: http://lists.digium.com/pipermail/asterisk-users/2002-August/003766.html But little else, and I can't find the source. Anyone else found this, or is the original author of the linked post on this list? I would recommend looking at the iaxclient lib. Somebody also made an OCX control for it. The code doesn't look very mature, but it worked great for my testing and it doesn't suffer from the silence suppression issues that SIP controls may have. iaxclient: http://iaxclient.sourceforge.net/ ocx: http://www.geocities.com/babarnazmi/index2.htm Tim Greiser -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.6 - Release Date: 2/7/2005 ___ 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] DTMF G729
I just installed G729amy Asterisk. I am facing some problems on DTMFMODE=INBAND. I just can t transfer my calls. Is there anybody out there who could give me WHICH DTMFMODE to use?P.S - I already tried DTMFMODE = RFC2833 and did not work!Thank you.Jmoura ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] DTMF G729
Thank you Michael, I tried to use RFC2838 without success. Which another type? Thank you jmoura - Original Message - From: Michael Bielicki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 22, 2004 6:14 PM Subject: Re: [Asterisk-Users] DTMF G729 Inband dtmf only works on alaw/ulaw. Use any other mode and it should work On Fri, 22 Oct 2004 17:02:00 -0200, JOAO CARLOS MOURA [EMAIL PROTECTED] wrote: I just installed G729a my Asterisk. I am facing some problems on DTMFMODE=INBAND. I just can t transfer my calls. Is there anybody out there who could give me WHICH DTMFMODE to use? P.S - I already tried DTMFMODE = RFC2833 and did not work! Thank you. Jmoura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The previous reload command didn't finish yet
Helo all, Some times meeting my asterisk with the message: The previous reload command didn't finish yet. Therefore, it loses the communication with the telephones. What necessary to make to decide this problem? Thank you JMoura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not register
Hi all, use the brought up to date version of the Asterisk and I have the following problem: Mine asterisk stop to register the extensions and I do not obtain to execute the command Stop Now. I do not see no message of error in logs. Somebody can help me? Thank's, JMoura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Balance my customers
Hi all How I configure the Asterisk to work in set with another Asterisk? I want to balance my customers in some computers with the Asterisk rounding. Thank You, Joao Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ammount of packages
Making use of Asterisk´s resources I can see that when 2 connections between users is active, this activity generates a huge ammount of packages on server interface where Asterisk is running. So, I can see that Asterisk controls the calls system usage. Is there a way to set up Asterisk to avoid this ammont of traffic control? Thank You João Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registers
How many registers SIP I can place in the Asterisk? Thank's Jmoura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Register Asterisk
Necessary to create a register in the Asterisk, more it has that to send the information: username, password, sip proxy, outboundproxy, domain/real. Help to decide this problem me? Thank´s Joao Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help Asterisk - SIP Proxy
Hi, I am developing ASTERISK as my SIP Proxy server. So, questions arise when you begin adding new users, being like this, I would like to know if ASTERISK working as SIP PROXY has a limit of REGISTERS and if the answer is : YES this limit is caused by HARDWARE - NETWORK or else... I´m making this questions because I am facing some problems with my working ASTERISK and sometimes I got some informations associated to Sip Register informed below like: 1. UNREACHABLE or, 2. LAGGED and one of my most often problems is right after a communication is established, simply goes interrupted without any specific reason. So if someone from the list faced the same problem or have some usual information regarded to this subject that could help me solve this INTERRUPTED communications PROBLEM, please do it. Below my ASTERISK informations and HARDWARE either: Asterisk version is : ASTERISK CVS - 03/02/04 --- HARDWARE INFORMATIONS --- 1. Memory : 256 ddr pc 21000 2. Processor : 1.900 + Athlon XP 3. Red Hat 9.0 João Carlos Moura NiNeTel Telecommunications +55 85 264-9039 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Basic authentication
How can I settup a way for Asterisk doesn´t make any use of DIGEST AUTHENTICATION method? I don t want ASTERISK to check out any username or password of my users. Thank you Joao Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Register Call-ID
Hi all, Need to register a SIP Server into Asterisk. But I must, before, to send a Call-ID to the service provider like: [EMAIL PROTECTED] Anyone here knows how to implement this? Thank You Joao Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk MySQL
I need to develop an web interface to include clients automatically in Asterisk. So, to make this possible I need that all my peers and exten being at a database (Mysql). Where do I find doc´s regarded for it? Thank you very much, J Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] register request time out
Hi all I am with problem to register a SIP proxy in asterisk. Necessary to send a Call-ID with the following description: [EMAIL PROTECTED] of the Server that I want to register. What I must make? Thank you jmoura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
