[asterisk-users] Re: Alert: Steering Committee Reminder and Agenda
OK. Have you had an opportunity to obtain the userid and password for the DEC Chair at the FDP website? We have changes to make there on the site, i.e. new officers. or The passcode on the DEC telephone, so that we can retrieve voice messages? SAL On Thursday 04 January 2007 11:40, [EMAIL PROTECTED] wrote: Looks like a very good agenda. We can't just keep the same Audit Comm. however, we can re-elect the same people if they are willing to serve. I think the members now are Susan Hunter, Doug Burleson and Noojin walker (?). Noojin can't be both a Steering Comm. member and an Audit Comm. member and we must elect these people whoever it turns out to be. For an outside audit, we can get John Jackson to do that. He is Treasurer of the DWC and has done audits for the DEC before. Evalyn From: Flash [EMAIL PROTECTED] Date: 2007/01/03 Wed PM 10:59:12 EST To: Undisclosed-Recipient:; Subject: Alert: Steering Committee Reminder and Agenda ___ --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] How do I diagnose the problem in this Asterisk test session with FWD?
=
SJphone Log
Outgoing SIP session
Respondent: (sip:[EMAIL PROTECTED])
Remote client:
Started: May 26 16:33
Accepted: no
Ended: May 26 16:34
End reason: Call rejected: 503 Service Unavailable
===
Asterisk Debug
Executing Dial(SIP/2201-a83e, IAX2/FWDNUMBER:@iax2.fwdnet.net/612|60|r)
in new stack
-- Called FWDNUMBER:@iax2.fwdnet.net/612
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00014ms SCall: 3 DCall: 0 [65.39.205.121:4569]
VERSION : 2
CALLED NUMBER : 612
CALLING NAME: MyName
LANGUAGE: en
USERNAME: FWDNUMBER
FORMAT : 4
CAPABILITY : 63494
ADSICPE : 2
DATE TIME : 179995695
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 00019ms SCall: 1 DCall: 0 [65.39.205.121:4569]
USERNAME: FWDNUMBER
REFRESH : 60
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00014ms SCall: 3 DCall: 0 [65.39.205.121:4569]
VERSION : 2
CALLED NUMBER : 612
CALLING NAME: MyName
LANGUAGE: en
USERNAME: FWDNUMBER
FORMAT : 4
CAPABILITY : 63494
ADSICPE : 2
DATE TIME : 179995695
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10019ms SCall: 1 DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10006ms SCall: 3 DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10019ms SCall: 1 DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 00019ms SCall: 1 DCall: 0 [65.39.205.121:4569]
USERNAME: FWDNUMBER
REFRESH : 60
=
IAX CONF
register = FWDNUMBER:[EMAIL PROTECTED]
[iaxfwd]
type=user
auth=rsa
inkeys=freeworlddialup
disallow=all
context=fromiaxfwd
context=fwd-out
EXTENSIONS CONF
FWDUSERID1=FWDNUMBER ; your calling number
FWDCIDNAME=MyName; your caller id
FWDPASSWORD=secret ; your password
FWDRINGS=sip/2201 ; the phone to ring
FWDVMBOX=2201 ; the VM box for this user
FWDPREFIX=8
[inbound-fwd]
include = daytime-fwd|9:00-21:00|*|*
include = nighttime-fwd|21:00-9:00|*|*
;
; Outbound
;
[fwd-out]
exten = _8.,1,SetCallerID(${FWDCIDNAME})
exten = _8.,2,Dial(IAX2/${FWDUSERID1}:
[EMAIL PROTECTED]/${EXTEN:1},60,r)
exten = _8.,3,Congestion
exten = _8.,4,Macro(fastbusy)
exten = _8.,5,Hangup
[fromiaxfwd]
exten = ${FWDUSERID1},1,Goto(inbound-fwd,s,1)
exten = ${FWDUSERID1},2,Voicemail2,u${FWDVMBOX}
exten = ${FWDUSERID1},102,Voicemail2,b${FWDVMBOX}
exten = ${FWDUSERID1},3,Hangup
[dialout]
include = fwd-out
include = fromiaxfwd
Thanks
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[Asterisk-Users] Please explain this streaming example to me
I believe that I found the example below while reading at the voip site. I understand the elements i.e. extensions, conference room assignment, etc. What I can not conceptualize is how do individuals using exten 1000 join the stream at exten 1002. ; Icecast streaming ; ; Conference Room exten = 1000,1,Answer exten = 1000,2,Wait(1) exten = 1000,3,MeetMe2(1000) ; Music On Hold exten = 1001,1,Answer exten = 1001,2,Wait(1) exten = 1001,3,MusicOnHold(random) ; Stream exten = 1002,1,Answer exten = 1002,2,Wait(1) exten = 1002,3,Ices(/etc/ices.xml) Thanks Flash ___ 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] Using patch -p0 meetme-diff-cbmysql_1.txt produces 'malformed patch' message
I have googled for several hours and have read several threads but I have not found an answer yet. I have downloaded asterisk-1.0.7 and WebMeetMe-Gui. I have tried to use the diff file 'meetme-diff-cbmysql_1.txt' to add the changes needed for WebMeetMe-Gui. Using 'patch -p0 meetme-diff-cbmysql_1.txt' in the apps directory returns: patching file app_meetme.c patch: malformed patch at line 157: @@ -680,7 +178,7 @@ I am beginning to think that I am using the wrong version of asterisk with this patch for app_meetme. Do anyone know which version of asterisk is compatible with WebMeetMe-Gui_v1.3.2 and its diff file 'meetme-diff-cbmysql_1.txt'? Thanks ___ 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
