Re: [Asterisk-Users] merchant account
You could have your customers call in and enter all of that -- then give them a confirmation number and they could fill out the rest online. --- trixter aka Bret McDanel [EMAIL PROTECTED] wrote: I am interested in hearing some user experiences of anyone using a merchant account. The constraints are that everything entered must be DTMF-able. Card number, CCV, exp, numeric portion of the street address, zipcode are all easy. name however is not so easy. How have others solved this problem? Or have they only set up systems where web access is required? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] DIDx error
I'm getting:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: REJECT
Timestamp: 9ms SCall: 8 DCall: 2
[66.98.180.77:4569]
CAUSE : No authority found
Under the DIDx number I'm putting IAX:
[EMAIL PROTECTED]/1567252(IAX)
where N is the rest of the numbers.
In iax.conf I have:
[default] ; DIDx
type=user
contact=DIDx-in
and in extensions.conf I have something like
[DIDx-in]
exten = 1567252,1,Noop(${DATETIME} ${CALLERID})
exten = 1567252,2,Answer
exten = 1567252,3,Goto(main-menu,s,2)
exten = 1567252,4,Hangup
What am I doing wrong?
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[Asterisk-Users] Invalid Extensions in Context but no invalid handler....
what does this mean and how do I fix it?
Channel 'SIP/3044-80e1' sent into invalid extension
'573486' in context 'redial-from-local', but no
invalid handler
Here is the
[redial-from-local]
_91NXXNXX,1,Macro(redial,${EXTEN})
_91NXXNXX,2,Congestion
_9NXXNXX,3,Macro(redial,${EXTEN})
_9NXXNXX,4,Congestion
a separate context passes this via:
Goto(redial-from-local,${lastcaller},1)
and then redial-from-local activates a macro called
redial
[macro-redial]
exten = s,1,SetCIDName(${OURCID}|a)
exten = s,2,SetCIDNum(${OURCIDN3}|a)
exten = s,3,Monitor(wav)
exten = s,4,Dial(IAX2/[EMAIL PROTECTED]/${ARG1})
exten = s,5,Congestion
exten = s,105,Dial(IAX2/[EMAIL PROTECTED]/${ARG1})
exten = s,106,Congestion
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Re: [Asterisk-Users] Call-in/Call-out
Ah It was a typo. It should work now! L:)
--- Erik Slooff [EMAIL PROTECTED] wrote:
snip
written by Crystal Stream, Incorporated
Here is my extensions.conf file. Things have been
left
out or changed to protect the innocent.
Why isn't it working when I call from the outside
that
when I press 124 it repeats the menu and doesn't
initiate DISA correctly to dial out?
[general]
static=yes
writeprotect=yes
[globals]
[voicepulse-in]
exten = ${OURVOIP1},1,Noop(${DATETIME}
${CALLERID})
exten = ${OURVOIP1},2,Answer
exten = ${OURVOIP1},3,Goto(main-menu,s,2)
exten = ${OURVOIP1},4,Hangup
[nufone-in]
exten = ${OURVOIP3},1,Noop(${DATETIME}
${CALLERID})
exten = ${OURVOIP3},2,Answer
exten = ${OURVOIP3},3,Goto(main-menu,s,2)
exten = ${OURVOIP3},4,Hangup
[incoming-sip]
include = voicepulse-in
include = nufone-in
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,Answer
exten = s,3,Goto(main-menu,s,6)
exten = s,4,Hangup
MAIN MENU ;;
[main-menu]
include = operator
include = queues
; if pressed 4-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for main menu selections:
exten = 1,1,Goto(office-day,1,1)
exten = 2,1,Goto(office-day,2,1)
exten = 3,1,Goto(office-day,3,1)
exten = 4,1,Goto(office-day,4,1)
exten = 5,1,Goto(office-day,5,1)
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,Wait(${WAIT_AFTER_ANSWER})
exten = s,3,SetCallerID(${CALLERID})
exten = s,4,DigitTimeout,2
exten = s,5,ResponseTimeout,7
exten =
s,6,Background(/usr/local/etc/asterisk/ivr/UPGRADEPHONES)
exten =
s,7,Background(/usr/local/etc/asterisk/ivr/GREETING)
exten = s,8,WaitExten(1.2)
exten = s,9,SetGlobalVar(prompt_loops=0)
exten =
s,10,GotoIfTime(07:00-18:00|mon-thu|*|*?office-day,s,2)
exten =
s,11,GotoIfTime(10:00-16:30|fri|*|*?office-day,s,2)
exten = s,12,Goto(office-night,s,1)
exten = t,1,Goto(main-menu,#,1) ; If
they
take too long, go to hangup
; invalid
exten = i,1,Wait(1)
exten = i,2,Playback(invalid) ; That's not
valid,
try again
exten = i,3,Wait(1)
exten = i,4,Goto(s,6)
; #=hangup
exten = #,1,Wait(1)
exten = #,2,Playback(vm-goodbye)
exten = #,3,Wait(2)
exten = #,4,Hangup
[office-day]
include = operator
include = queues
; if pressed 3-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for accessing voicemail:
include = voicemail
exten = s,1,SetGlobalVar(prompt_loops=1)
exten = s,2,WaitExten(${BETWEEN_PROMPTS})
exten =
s,3,Background(/usr/local/etc/asterisk/ivr/MAINMENU)
exten = s,4,WaitExten(4)
exten =
s,5,SetGlobalVar(prompt_loops=$[${prompt_loops} +
1])
exten = s,6,GotoIf($[${prompt_loops}
${MAX_MENU_LOOPS}] ? 2:23)
exten = s,7,Goto(operator,0,1)
; invalid
exten = i,1,Playback(invalid) ; That's not
valid,
try again
exten = i,2,Wait(1)
exten = i,3,Goto(s,7)
; timeout
exten = t,1,Goto(operator,0,1)
[office-night]
include = operator
include = queues
; if pressed 3-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for accessing voicemail:
include = voicemail
exten = s,1,SetGlobalVar(prompt_loops=1)
exten = s,2,WaitExten(${BETWEEN_PROMPTS})
exten =
s,3,Background(/usr/local/etc/asterisk/ivr/MAINMENU)
exten = s,4,WaitExten(4)
exten =
s,5,SetGlobalVar(prompt_loops=$[${prompt_loops} +
1])
exten = s,6,GotoIf($[${prompt_loops}
${MAX_MENU_LOOPS}] ? 2:23)
exten = s,7,Goto(operator,0,1)
; invalid
exten = i,1,Wait(1)
exten = i,2,Playback(invalid) ; That's not
valid,
try again
exten = i,3,Wait(1)
exten = i,4,Goto(s,4)
; timeout
exten = t,1,Goto(main-menu,#,1) ; If
they
take too long, go to hangup
[local]
; Directory:
exten = 411,1,Directory(crystal-sip|local)
exten = 411,2,Hangup
; DISA
exten = 124,1,Answer
exten = 124,2,DigitTimeout(5)
exten = 124,3,ResponseTimeout(10)
exten = 124,4,Authenticate(16435679)
exten = 124,DISA(4376194673164379|crystal-sip)
and snip
---
Why did you not set a priority here or is it a typo?
Erik
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[Asterisk-Users] IPComms Setup
Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect attempt from 69.15.xxx.xx, request '[EMAIL PROTECTED]' does not exist __ Yahoo! for Good Donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPComms Setup
this isn't working
[IPComms-in]
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,SetCallerID(${CALLERID})
exten = s,3,Answer
exten = s,4,Goto(main-menu,s,2)
exten = s,5,Hangup
What I have is a block of 20 DIDs and I want to accept
calls from all of them.
It would be way to freaking complicated to do
exten = 2027575120,1,Noop(
.
exten = 2027575121,1,Noop(
et cetera
How do I get this done?
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RE: [Asterisk-Users] IPComms Setup
CURRENTLY in GLOBALS I have
IPCCIDN01 - IPCCIDN20 set to their respective numbers
How do I need to change the IPComms-in context to do
what you say
--- Kevin Walsh [EMAIL PROTECTED] wrote:
Crystal Stream, Incorporated
[EMAIL PROTECTED] wrote:
this isn't working
[IPComms-in]
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,SetCallerID(${CALLERID})
exten = s,3,Answer
exten = s,4,Goto(main-menu,s,2)
exten = s,5,Hangup
What I have is a block of 20 DIDs and I want to
accept calls from all of
them.
It would be way to freaking complicated to do
exten = 2027575120,1,Noop(
.
exten = 2027575121,1,Noop(
et cetera
How do I get this done?
You could wildcard your DDIs, replacing 20, 21 etc.
with [23][0-9], or
whatever. Alternatively, you could create a macro
that would look
a lot like the body of your [IPComms-in] context,
and then call that
from 20 separate DDI exten lines. I'd just go with
the wildcard.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K
e v i n W a l s h
_/ _/_/ _/ _/ _/_/ _/_/
[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_/_/ _/_/
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[Asterisk-Users] Help! Extensions
Hello How do I fix this
[IPComms-in]
exten = ${IPCCIDN01},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN01},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN02},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN02},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN03},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN03},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN04},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN04},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN05},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN05},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN06},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN06},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN07},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN07},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN08},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN08},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN09},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN09},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN10},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN10},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN11},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN11},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN12},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN12},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN13},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN13},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN14},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN14},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN15},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN15},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN16},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN16},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN17},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN17},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN18},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN18},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN19},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN19},2,SetCallerID(${CALLERID})
exten = ${IPCCIDN20},1,Noop(${DATETIME} ${CALLERID})
exten = ${IPCCIDN20},2,SetCallerID(${CALLERID})
exten = s,3,Answer
exten = s,4,Goto(main-menu,s,2)
exten = s,5,Hangup
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[Asterisk-Users] Two Questions
1) What do these two notices mean? Oct 4 09:34:30 NOTICE[49584]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 198.22.67.70, request '[EMAIL PROTECTED]' does not exist Oct 4 09:34:51 NOTICE[49584]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist 2) I have ran the registration utility and done the setup of g.729 codec. How do I enable that (settings are enabled in modules.conf) so that it will use that codec priority before ulaw/alaw, et cetera? I just have allow=g729 in the sip/iax configs Joshua __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Call-in/Call-out
Hello, How would I setup where I call into my number and press say 911 and then it would ask for a pass and would accept it and then would prompt for a number so I could call out of my number on the road? Joshua __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Number Restriction
Hello, I have a block of 25 DIDs and have 10 phones on the network. I want when a person tries to call out for * to pick a number for the CIDN and I want to make sure that the number isn't duplicated while it's in use. Joshua __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] G.729 Codec
Hello, How do I make sure the G.729 codec is being utilized fully and not just as a passthru? I've registered it and followed the install instructions __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] G.729 Codec
show g729 doesn't work --- Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: from the CLI, show g729 -- In my situation, with polycom 501s, if a phone calls another internal phone and canreinvite is set to yes, this does not count against your licenses 'cause the phones are now the only devices in the conversation. you can still find in my phones' status menu what codec it has negotiated for the current call. When asterisk drops from the loop for me, my phones remain on g729. Moj Crystal Stream, Incorporated wrote: Hello, How do I make sure the G.729 codec is being utilized fully and not just as a passthru? I've registered it and followed the install instructions __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Call-in/Call-out
Here is my extensions.conf file. Things have been left
out or changed to protect the innocent.
Why isn't it working when I call from the outside that
when I press 124 it repeats the menu and doesn't
initiate DISA correctly to dial out?
[general]
static=yes
writeprotect=yes
[globals]
[voicepulse-in]
exten = ${OURVOIP1},1,Noop(${DATETIME} ${CALLERID})
exten = ${OURVOIP1},2,Answer
exten = ${OURVOIP1},3,Goto(main-menu,s,2)
exten = ${OURVOIP1},4,Hangup
[nufone-in]
exten = ${OURVOIP3},1,Noop(${DATETIME} ${CALLERID})
exten = ${OURVOIP3},2,Answer
exten = ${OURVOIP3},3,Goto(main-menu,s,2)
exten = ${OURVOIP3},4,Hangup
[incoming-sip]
include = voicepulse-in
include = nufone-in
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,Answer
exten = s,3,Goto(main-menu,s,6)
exten = s,4,Hangup
MAIN MENU ;;
[main-menu]
include = operator
include = queues
; if pressed 4-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for main menu selections:
exten = 1,1,Goto(office-day,1,1)
exten = 2,1,Goto(office-day,2,1)
exten = 3,1,Goto(office-day,3,1)
exten = 4,1,Goto(office-day,4,1)
exten = 5,1,Goto(office-day,5,1)
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,Wait(${WAIT_AFTER_ANSWER})
exten = s,3,SetCallerID(${CALLERID})
exten = s,4,DigitTimeout,2
exten = s,5,ResponseTimeout,7
exten =
s,6,Background(/usr/local/etc/asterisk/ivr/UPGRADEPHONES)
exten =
s,7,Background(/usr/local/etc/asterisk/ivr/GREETING)
exten = s,8,WaitExten(1.2)
exten = s,9,SetGlobalVar(prompt_loops=0)
exten =
s,10,GotoIfTime(07:00-18:00|mon-thu|*|*?office-day,s,2)
exten =
s,11,GotoIfTime(10:00-16:30|fri|*|*?office-day,s,2)
exten = s,12,Goto(office-night,s,1)
exten = t,1,Goto(main-menu,#,1) ; If they
take too long, go to hangup
; invalid
exten = i,1,Wait(1)
exten = i,2,Playback(invalid) ; That's not valid,
try again
exten = i,3,Wait(1)
exten = i,4,Goto(s,6)
; #=hangup
exten = #,1,Wait(1)
exten = #,2,Playback(vm-goodbye)
exten = #,3,Wait(2)
exten = #,4,Hangup
[office-day]
include = operator
include = queues
; if pressed 3-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for accessing voicemail:
include = voicemail
exten = s,1,SetGlobalVar(prompt_loops=1)
exten = s,2,WaitExten(${BETWEEN_PROMPTS})
exten =
s,3,Background(/usr/local/etc/asterisk/ivr/MAINMENU)
exten = s,4,WaitExten(4)
exten =
s,5,SetGlobalVar(prompt_loops=$[${prompt_loops} + 1])
exten = s,6,GotoIf($[${prompt_loops}
${MAX_MENU_LOOPS}] ? 2:23)
exten = s,7,Goto(operator,0,1)
; invalid
exten = i,1,Playback(invalid) ; That's not valid,
try again
exten = i,2,Wait(1)
exten = i,3,Goto(s,7)
; timeout
exten = t,1,Goto(operator,0,1)
[office-night]
include = operator
include = queues
; if pressed 3-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for accessing voicemail:
include = voicemail
exten = s,1,SetGlobalVar(prompt_loops=1)
exten = s,2,WaitExten(${BETWEEN_PROMPTS})
exten =
s,3,Background(/usr/local/etc/asterisk/ivr/MAINMENU)
exten = s,4,WaitExten(4)
exten =
s,5,SetGlobalVar(prompt_loops=$[${prompt_loops} + 1])
exten = s,6,GotoIf($[${prompt_loops}
${MAX_MENU_LOOPS}] ? 2:23)
exten = s,7,Goto(operator,0,1)
; invalid
exten = i,1,Wait(1)
exten = i,2,Playback(invalid) ; That's not valid,
try again
exten = i,3,Wait(1)
exten = i,4,Goto(s,4)
; timeout
exten = t,1,Goto(main-menu,#,1) ; If they
take too long, go to hangup
[local]
; Directory:
exten = 411,1,Directory(crystal-sip|local)
exten = 411,2,Hangup
; DISA
exten = 124,1,Answer
exten = 124,2,DigitTimeout(5)
exten = 124,3,ResponseTimeout(10)
exten = 124,4,Authenticate(16435679)
exten = 124,DISA(4376194673164379|crystal-sip)
; # REAL LOCAL EXTENSIONS
###
exten = 9050,1,Macro(sipline,${SIP9050})
exten = 9061,1,Macro(sipline,${SIP9061})
exten = 9072,1,Macro(sipline,${SIP9072})
exten = 9083,1,Macro(sipline,${SIP9083})
exten = 9094,1,Macro(sipline,${SIP9094})
exten = 8005,1,Macro(sipline,${SIP8005})
exten = 8016,1,Macro(sipline,${SIP8016})
; invalid
exten = i,1,Playback(invalid) ; That's not valid,
try again
exten = i,2,Wait(1)
exten = i,3,Goto(0,2)
[agent_con]
exten = _3XXX,1,SetGroup(${EXTEN})
exten = _3XXX,2,CheckGroup(1)
exten = _3XXX,3,Dial(SIP/${EXTEN})
exten = _3XXX,103,Busy
[agents]
exten =
1000,1,AgentCallbackLogin([EMAIL PROTECTED])
exten = 1000,2,Hangup
exten =
1001,1,AgentCallbackLogin(2101|[EMAIL PROTECTED])
exten = 1001,2,Hangup
exten =
1002,1,AgentCallbackLogin(2202|[EMAIL PROTECTED])
exten = 1002,2,Hangup
exten =
1003,1,AgentCallbackLogin(2303|[EMAIL PROTECTED])
exten = 1003,2,Hangup
exten = 1091,1,AgentCallbackLogin(2401)
exten = 1091,2,Hangup
exten = 1092,1,AgentCallbackLogin(2502)
exten = 1092,2,Hangup
exten =
Re: [Asterisk-Users] G.729 Codec
show g729 doesn't work still but see this: crystalstream*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - 5 2 2 5 2 1 8 23627 gsm 2 - 2 2 5 2 1 8 23627 ulaw 2 5 - 1 5 2 1 8 23627 alaw 2 5 1 - 5 2 1 8 23627 g726 5 8 5 5 - 5 4 11 53930 adpcm 2 5 2 2 5 - 1 8 23627 slin 1 4 1 1 4 1 - 7 13526 lpc10 4 7 4 4 7 4 3 - 43829 g729 2 5 2 2 5 2 1 8 -3627 speex 2 5 2 2 5 2 1 8 2 -27 ilbc101310101310 9 161044 - --- Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: On my 1.2.0b1: pbx*CLI show g729 0/0 encoders/decoders of 10 licensed channels are currently in use pbx*CLI as bill mentioned, try show translation to see if g729 is registered. if there are '-' in the columns and rows, something's not right. There should be small numbers there, like 10-20ms in my experience. I'm not familiar with wy show g729 doesn't work if the g729 codec is registered, however Crystal Stream, Incorporated wrote: show g729 doesn't work --- Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: from the CLI, show g729 -- In my situation, with polycom 501s, if a phone calls another internal phone and canreinvite is set to yes, this does not count against your licenses 'cause the phones are now the only devices in the conversation. you can still find in my phones' status menu what codec it has negotiated for the current call. When asterisk drops from the loop for me, my phones remain on g729. Moj Crystal Stream, Incorporated wrote: Hello, How do I make sure the G.729 codec is being utilized fully and not just as a passthru? I've registered it and followed the install instructions __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] RECAP: 3?
Just to recap my other three questions: 1) I have a block of 25 DIDs and have 10 phones on the network. I want when a person tries to call out for * to pick a number for the CIDN and I want to make sure that the number isn't duplicated while it's in use. 2) What do these two notices mean? Oct 4 09:34:30 NOTICE[49584]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 198.22.67.70, request '[EMAIL PROTECTED]' does not exist Oct 4 09:34:51 NOTICE[49584]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist 3) I have ran the registration utility and done the setup of g.729 codec. How do I enable that (settings are enabled in modules.conf) so that it will use that codec priority before ulaw/alaw, et cetera? I just have allow=g729 in the sip/iax configs Joshua __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Forcing Codec Usage
Hello, I have VPC (Voice Pulse Connect) and NuFone for providers and I have setup modules.conf with the registered (Digium) G.729 Codec such as: load = codec_g729a.so load = res_crypto.so With both sip/iax2 configuration disallow=all is first and then allow=g729 is next (allow=ulaw,allow=alaw,allow=gsm are next after allow=g729) and it always dials via ulaw. Why is this happening? Josh __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Forcing Codec Usage
Okay. I learned that if I comment ulaw out it works with NuFone kind of it connects and then I can't hear anything. Voice Pulse Connect flags Congestion. --- Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: maybe your phones (polycom 50?'s if I remember right) have the priority configured to place ulaw above g729? It's in your ipmid.cfg (if you do use polycoms) or maybe your sip.cfg if you've upgraded to SIP 1.5.2 and migrated your config files Crystal Stream, Incorporated wrote: Hello, I have VPC (Voice Pulse Connect) and NuFone for providers and I have setup modules.conf with the registered (Digium) G.729 Codec such as: load = codec_g729a.so load = res_crypto.so With both sip/iax2 configuration disallow=all is first and then allow=g729 is next (allow=ulaw,allow=alaw,allow=gsm are next after allow=g729) and it always dials via ulaw. Why is this happening? Josh __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Nufone
After -- IAX2/NuFone/3 is making progress passing it to SIP/3044-bcd0 I'm getting a Busy tone and it's not even connecting the call. -- Executing Macro(SIP/3044-bcd0, outvoip-2|1800759) in new stack -- Executing SetCIDName(SIP/3044-bcd0, X X X|a) in new stack -- Executing SetCIDNum(SIP/3044-bcd0, 8663xx3|a) in new stack -- Executing Authenticate(SIP/3044-bcd0, xx) in new stack -- Playing 'agent-pass' (language 'en') -- Playing 'auth-thankyou' (language 'en') -- Executing Monitor(SIP/3044-bcd0, wav) in new stack -- Executing Ringing(SIP/3044-bcd0, ) in new stack -- Executing Wait(SIP/3044-bcd0, 2) in new stack -- Executing Dial(SIP/3044-bcd0, IAX2/[EMAIL PROTECTED]/1800759) in new stack -- Called [EMAIL PROTECTED]/1800759 -- Call accepted by 66.225.202.72 (format ulaw) -- Format for call is ulaw -- IAX2/NuFone/3 is making progress passing it to SIP/3044-bcd0 -- Hungup 'IAX2/NuFone/3' == Spawn extension (macro-outvoip-2, s, 7) exited non-zero on 'SIP/3044-bcd0' in macro 'outvoip-2' == Spawn extension (crystal-sip, 8800759, 1) exited non-zero on 'SIP/3044-bcd0' x*CLI iax2 show peers Name/UsernameHost Mask Port Status voicepulse2/Fbg 66.234.228.166 (S) 255.255.255.255 4569 Unmonitored voicepulse1/Fbg 66.234.228.160 (S) 255.255.255.255 4569 Unmonitored NuFone 66.225.202.72 (S) 255.255.255.255 4569 Unmonitored __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Nufone
crystalstream*CLI -- Executing Macro(SIP/3044-5300, outvoip-2|1800759) in new stack -- Executing SetCIDName(SIP/3044-5300, CRYSTAL STREAM NET|a) in new st ack -- Executing SetCIDNum(SIP/3044-5300, 866xxx|a) in new stack -- Executing Authenticate(SIP/3044-5300, 123987) in new stack -- Playing 'agent-pass' (language 'en') -- Playing 'auth-thankyou' (language 'en') -- Executing Monitor(SIP/3044-5300, wav) in new stack -- Executing Ringing(SIP/3044-5300, ) in new stack -- Executing Wait(SIP/3044-5300, 2) in new stack -- Executing Dial(SIP/3044-5300, IAX2/[EMAIL PROTECTED]/1800759 ) in new stack -- Called [EMAIL PROTECTED]/1800759 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 1 DCall: 0 [66.225.202.72:4569] VERSION : 2 CALLED NUMBER : 1800759 CALLING NUMBER : 8663113060 LANGUAGE: en USERNAME: username-hidden FORMAT : 4 CAPABILITY : 63502 ADSICPE : 2 DATE TIME : 188966086 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 00061 DCall: 1 [66.225.202.72:4569] AUTHMETHODS : 2 CHALLENGE : 150617580 USERNAME: username-hidden Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00033ms SCall: 1 DCall: 00061 [66.225.202.72:4569] MD5 RESULT : c8214533976d4dec8b233543dac0eaac Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00036ms SCall: 00061 DCall: 1 [66.225.202.72:4569] FORMAT : 4 -- Call accepted by 66.225.202.72 (format ulaw) -- Format for call is ulaw Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00036ms SCall: 1 DCall: 00061 [66.225.202.72:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00060ms SCall: 1 DCall: 00061 [66.225.202.72:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00080ms SCall: 00061 DCall: 1 [66.225.202.72:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00080ms SCall: 1 DCall: 00061 [66.225.202.72:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00060ms SCall: 00061 DCall: 1 [66.225.202.72:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: CONTROL Subclass: (15?) Timestamp: 00123ms SCall: 00061 DCall: 1 [66.225.202.72:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 00123ms SCall: 1 DCall: 00061 [66.225.202.72:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass: (14?) Timestamp: 01423ms SCall: 00061 DCall: 1 [66.225.202.72:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 01423ms SCall: 1 DCall: 00061 [66.225.202.72:4569] -- IAX2/NuFone/1 is making progress passing it to SIP/3044-5300 Oct 3 12:38:19 WARNING[49584]: app_dial.c:372 wait_for_answer: Unable to forwar d frame -- Hungup 'IAX2/NuFone/1' == Spawn extension (macro-outvoip-2, s, 7) exited non-zero on 'SIP/3044-5300' in macro 'outvoip-2' == Spawn extension (crystal-sip, 8800759, 1) exited non-zero on 'SIP/3044- 5300' --- Tom Vile [EMAIL PROTECTED] wrote: how many digits is your callerid passing to the trunk? I am seeing 11 8663xx3 is that correct? I had an issue last week with passing to many digits to my provider and the call would hang up immediately. You could also turn debugging on for this so we can get a better log. iax2 debug peer nufone On 10/3/05, Crystal Stream, Incorporated [EMAIL PROTECTED] wrote: After -- IAX2/NuFone/3 is making progress passing it to SIP/3044-bcd0 I'm getting a Busy tone and it's not even connecting the call. -- Executing Macro(SIP/3044-bcd0, outvoip-2|1800759) in new stack -- Executing SetCIDName(SIP/3044-bcd0, X X X|a) in new stack -- Executing SetCIDNum(SIP/3044-bcd0, 8663xx3|a) in new stack -- Executing Authenticate(SIP/3044-bcd0, xx) in new stack -- Playing 'agent-pass' (language 'en') -- Playing 'auth-thankyou' (language 'en') -- Executing Monitor(SIP/3044-bcd0, wav) in new stack -- Executing Ringing(SIP/3044-bcd0, ) in new stack -- Executing Wait(SIP/3044-bcd0, 2) in new stack -- Executing Dial(SIP/3044-bcd0, IAX2/[EMAIL PROTECTED]/1800759) in new stack -- Called [EMAIL PROTECTED]/1800759 -- Call accepted by 66.225.202.72 http://66.225.202.72 (format ulaw) -- Format for call is ulaw -- IAX2/NuFone/3 is making progress passing it to SIP/3044-bcd0 -- Hungup 'IAX2/NuFone/3' == Spawn extension (macro-outvoip-2, s, 7) exited non-zero on 'SIP/3044-bcd0
[Asterisk-Users] AgentRecord In and Out streams
How do I combined these in and out wav files on the fly through asterisk to where I hear the whole conversation and only have one wav-file (i.e. : agent-1001-asterisk-478-1127389080-17-in_out.wav) agent-1001-asterisk-478-1127389080-17-in.wav agent-1001-asterisk-478-1127389080-17-out.wav __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Problem with Queues
I am getting this on the console once people call in -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1001 -- Executing Macro(Local/[EMAIL PROTECTED],2, sipline|3044) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/3044|20|t) in new stack -- Called 3044 -- SIP/3044-6a6e is ringing -- Agent/1001 is ringing -- SIP/3044-6a6e answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge -- Started music on hold, class 'default', on Local/[EMAIL PROTECTED],2 -- Stopped music on hold on Local/[EMAIL PROTECTED],2 Now, I answer, press # on my phone to acknowledge the call and it goes back again.Am I doing something wrong? __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Addendum to Problem with Queues question
Here is the full transaction -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1001 -- Executing Macro(Local/[EMAIL PROTECTED],2, sipline|3044) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/3044|20|t) in new stack -- Called 3044 -- SIP/3044-ea92 is ringing -- Agent/1001 is ringing -- SIP/3044-ea92 answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge -- Started music on hold, class 'default', on Local/[EMAIL PROTECTED],2 -- Unable to find extension '' in context 'crystal-sip' -- Playing 'pbx-invalid' (language 'en') Sep 21 10:30:30 WARNING[52987]: file.c:550 ast_readaudio_callback: Failed to write frame -- Stopped music on hold on Local/[EMAIL PROTECTED],2 Sep 21 10:30:30 WARNING[52987]: res_features.c:450 ast_bridge_call: Bridge failed on channels Local/[EMAIL PROTECTED],2 and SIP/3044-ea92 == Spawn extension (macro-sipline, s, 1) exited non-zero Why doesn't ast_bridge_call do it's thing __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Problem with Queues
features.conf is devoid of # the queue doesn't have h in it. only have tT --- Kevin Bockman [EMAIL PROTECTED] wrote: Crystal Stream, Incorporated wrote: I am getting this on the console once people call in -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1001 -- Executing Macro(Local/[EMAIL PROTECTED],2, sipline|3044) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/3044|20|t) in new stack -- Called 3044 -- SIP/3044-6a6e is ringing -- Agent/1001 is ringing -- SIP/3044-6a6e answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge -- Started music on hold, class 'default', on Local/[EMAIL PROTECTED],2 -- Stopped music on hold on Local/[EMAIL PROTECTED],2 Now, I answer, press # on my phone to acknowledge the call and it goes back again.Am I doing something wrong? You mean, it hangs up and calls you back again? Sounds like you have option h on Queue and have # set to hangup in features.conf. Kevin ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] re: Problems with Queues
Here is my extensions.conf file for debugging __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com extensions.conf Description: 3949034846-extensions.conf ___ --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] DIDx
Hey anyone know once you've bought a DID from DIDx under the Purchased DIDs tab you can click a link for one of your numbers that says '0 (SIP)' and when you click that like there's SIP and a space or the URL and AIX and a space for the URL What do I put here? How do I setup Asterisk to accept these Joshua __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Unable to allocate channel structure
I'm getting these messages from this Asterisk version: Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h Sep 19 07:11:18 WARNING[99391]: chan_sip.c:2084 sip_new: Unable to allocate channel structure Sep 19 07:11:18 NOTICE[99391]: chan_sip.c:7523 handle_request: Unable to create/find channel When I kill the asterisk process and start back up Asterisk it works fine. This happens about 30 minutes after Asterisk has been up. Joshua __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Error with VOIP-INFO
when I searched for _**6XX the site returned:
An error occured in a database query!
Context:
File/tiki-searchresults.php
Url
/tiki-searchresults.php?words=_**6XXwhere=pagessearch=go
Query:
SELECT COUNT(*) FROM tiki_comments c, tiki_pages p
WHERE c.objectType = wiki page AND
p.pageName=c.object AND (UPPER(c.title) REGEXP
'.*_**6XX.*' OR UPPER(c.data) REGEXP '.*_**6XX.*')
Values:
array(5) {
[0]=
array(6) {
[file]=
string(46)
/var/www/html/tikiwiki-1.8.5/lib/tikidblib.php
[line]=
int(109)
[function]=
string(9) sql_error
[class]=
string(9) searchlib
[type]=
string(2) -
[args]=
array(3) {
[0]=
string(185) SELECT COUNT(*) FROM tiki_comments
c, tiki_pages p WHERE c.objectType = wiki page AND
p.pageName=c.object AND (UPPER(c.title) REGEXP
'.*_**6XX.*' OR UPPER(c.data) REGEXP '.*_**6XX.*')
[1]=
NULL
[2]=
bool(false)
}
}
[1]=
array(6) {
[file]=
string(46)
/var/www/html/tikiwiki-1.8.5/lib/searchlib.php
[line]=
int(106)
[function]=
string(6) getone
[class]=
string(9) searchlib
[type]=
string(2) -
[args]=
array(1) {
[0]=
string(185) SELECT COUNT(*) FROM tiki_comments
c, tiki_pages p WHERE c.objectType = wiki page AND
p.pageName=c.object AND (UPPER(c.title) REGEXP
'.*_**6XX.*' OR UPPER(c.data) REGEXP '.*_**6XX.*')
}
}
[2]=
array(6) {
[file]=
string(46)
/var/www/html/tikiwiki-1.8.5/lib/searchlib.php
[line]=
int(110)
[function]=
string(5) _find
[class]=
string(9) searchlib
[type]=
string(2) -
[args]=
array(5) {
[0]=
array(10) {
[from]=
string(29) tiki_comments c, tiki_pages p
[name]=
string(7) c.title
[data]=
string(6) c.data
[hits]=
string(6) p.hits
[lastModif]=
string(13) c.commentDate
[href]=
string(31) tiki-index.php?page=%s#comments
[id]=
array(2) {
[0]=
string(10) p.pageName
[1]=
string(10) c.threadId
}
[pageName]=
string(33) CONCAT(p.pageName, : , c.title)
[search]=
array(2) {
[0]=
string(7) c.title
[1]=
string(6) c.data
}
[filter]=
string(50) c.objectType = wiki page AND
p.pageName=c.object
}
[1]=
string(6) _**6XX
[2]=
#8747;(0)
[3]=
string(2) 10
[4]=
bool(false)
}
}
[3]=
array(6) {
[file]=
string(46)
/var/www/html/tikiwiki-1.8.5/lib/searchlib.php
[line]=
int(156)
[function]=
string(5) _find
[class]=
string(9) searchlib
[type]=
string(2) -
[args]=
array(5) {
[0]=
array(10) {
[from]=
string(29) tiki_comments c, tiki_pages p
[name]=
string(7) c.title
[data]=
string(6) c.data
[hits]=
string(6) p.hits
[lastModif]=
string(13) c.commentDate
[href]=
string(31) tiki-index.php?page=%s#comments
[id]=
array(2) {
[0]=
string(10) p.pageName
[1]=
string(10) c.threadId
}
[pageName]=
string(33) CONCAT(p.pageName, : , c.title)
[search]=
array(2) {
[0]=
string(7) c.title
[1]=
string(6) c.data
}
[filter]=
string(50) c.objectType = wiki page AND
p.pageName=c.object
}
[1]=
string(6) _**6XX
[2]=
#8747;(0)
[3]=
string(2) 10
[4]=
bool(true)
}
}
[4]=
array(6) {
[file]=
string(51)
/var/www/html/tikiwiki-1.8.5/tiki-searchresults.php
[line]=
int(165)
[function]=
string(10) find_wikis
[class]=
string(9) searchlib
[type]=
string(2) -
[args]=
array(4) {
[0]=
string(6) _**6XX
[1]=
#8747;(0)
[2]=
string(2) 10
[3]=
bool(true)
}
}
}
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