Re: [asterisk-users] getting results messages from CLI commands via -rx
On Fri, Sep 19, 2008 at 12:54:58PM -0700, George Williams wrote: Hi, I am issuing CLI commands via script, using the asterisk -rx method. Its working great. Now, I need to get the results of the command to look for error messages, etc. I've tried setting several -v flags as well, but I only get the Asterisk startup text (version, license info, etc), not the results of the command itself. Don't use v-s. They are global. It seems you didn't have the '-x'. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] getting results messages from CLI commands via -rx
Hi, I am issuing CLI commands via script, using the asterisk -rx method. Its working great. Now, I need to get the results of the command to look for error messages, etc. I've tried setting several -v flags as well, but I only get the Asterisk startup text (version, license info, etc), not the results of the command itself. Is this even possible? Thanx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting results messages from CLI commands via -rx
On Friday 19 September 2008 14:54:58 George Williams wrote: I am issuing CLI commands via script, using the asterisk -rx method. Its working great. Now, I need to get the results of the command to look for error messages, etc. I've tried setting several -v flags as well, but I only get the Asterisk startup text (version, license info, etc), not the results of the command itself. If you're doing activities via a script, the recommended method is to do this via the Asterisk Manager Interface. There is a sample script within the contrib/scripts directory called astcli, which shows how to do this in Perl. Note that the script is only available in trunk, though it should work with other versions of Asterisk. http://svn.digium.com/view/asterisk/trunk/contrib/scripts/astcli?view=log -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd results from fxotune?
I recently ran fxotune against our incoming PSTN lines to try and help with some echo problems. It produced the following fxotune.conf file: 2=8,253,2,244,255,10,244,3,253 3=4,0,0,0,0,0,0,0,0 4=4,0,0,0,0,0,0,0,0 I'm a bit surprised by all of the '0's for channels 34, esp. given that it's populated values for channel 2. Is this considered 'normal' behaviour, or is something amiss? Thanks, j ___ --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] Dialstatus results
Hi all, i just have a question: could i Known the state of a SIP phone without make it a Dial ? Thanks Giordano ___ --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] Dreadful results from zttest with TE210P and Dell 2850?
Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. Thanks! ___ --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] Dreadful results from zttest with TE210P and Dell2850?
Using an SMP kernel will fix the interrupt sharing, you could also disable hyperthreading and set runlevel 3. FWIW I almost exclusively use Poweredge 850 for my * servers with a third party sata raid controller if raid is required. Never had any problems. Craig - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Monday, April 24, 2006 6:38 PM Subject: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850? Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. Thanks! ___ --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] Dreadful results from zttest with TE210P and Dell2850?
Thanks for the hints and tips. While you are familiar with the 2850, I am using the PERC raid controller but guess this shouldn't make any real difference. I used the middle PCI slot for the TE210P, do you use any particular slot. I will disable HyperThreading and the box was already running an SMP kernel (there were no irq conflicts shown by lspci -v) in runlevel 3. Are you using the onboard e1000 ethernet controllers? The wiki is advising not to. Thanks for your input! Remco On Mon, 24 Apr 2006, Craig Guy wrote: Using an SMP kernel will fix the interrupt sharing, you could also disable hyperthreading and set runlevel 3. FWIW I almost exclusively use Poweredge 850 for my * servers with a third party sata raid controller if raid is required. Never had any problems. Craig - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Monday, April 24, 2006 6:38 PM Subject: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850? Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. Thanks! ___ --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 ___ --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] Unexpected results with While and EndWhile applications
At 11:39 PM -0400 on 9/5/05, C F wrote:
On 9/5/05, John Todd [EMAIL PROTECTED] wrote:
I seem to be having a conceptual problem with the While and
EndWhile applications. It seems that on the first cycle, even if
the result of the While is false that the enclosed applications
will get run. Is this expected? It seems to be counter-intuitive,
but I don't know what the intent of the While routines is. I could
of course put a GotoIf before the While loop to check to ensure
that the first expression is true before entry into the While loop,
but that seems redundant and ugly since the while point of While and
EndWhile is to avoid the inelegance of GotoIf, I thought.
If anyone can't come up with a better explanation, I'll open a ticket
on this but I'd like to first make sure that this behavior is not
expected.
exten = 2231,1,Set(staticnumber=0)
exten = 2231,n,Set(counter=1)
exten = 2231,n,While($[${counter}${staticnumber}])
Put A space around the operator, like this
exten = 2231,n,While($[${counter} ${staticnumber}])
This should help it.
That's no longer required in CVS-HEAD, if I recall correctly.
In any case, this does not make a difference, and even looking
logically at the example shows that it is not behaving correctly. (It
parses correctly on the second instance, but not on the first.)
This is looking more like a bug the longer I think about it.
JT
exten = 2231,n,NoOp(This part of the code should never run!)
exten = 2231,n,Set(counter=$[${counter}+1])
exten = 2231,n,EndWhile
exten = 2231,n,NoOp(This part of the code should be the only thing
that gets run!)
Console output from dialing 2231:
-- Executing Set(SIP/2203-c134, staticnumber=0) in new stack
-- Executing Set(SIP/2203-c134, counter=1) in new stack
-- Executing While(SIP/2203-c134, 0) in new stack
-- Executing NoOp(SIP/2203-c134, This part of the code should
never run!) in new stack
-- Executing Set(SIP/2203-c134, counter=2) in new stack
-- Executing EndWhile(SIP/2203-c134, ) in new stack
-- Executing NoOp(SIP/2203-c134, This part of the code should
be the only thing that gets run!) in new stack
*CLI show version
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux
on 2005-09-03 23:27:34 UTC
JT
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[Asterisk-Users] Unexpected results with While and EndWhile applications
I seem to be having a conceptual problem with the While and
EndWhile applications. It seems that on the first cycle, even if
the result of the While is false that the enclosed applications
will get run. Is this expected? It seems to be counter-intuitive,
but I don't know what the intent of the While routines is. I could
of course put a GotoIf before the While loop to check to ensure
that the first expression is true before entry into the While loop,
but that seems redundant and ugly since the while point of While and
EndWhile is to avoid the inelegance of GotoIf, I thought.
If anyone can't come up with a better explanation, I'll open a ticket
on this but I'd like to first make sure that this behavior is not
expected.
exten = 2231,1,Set(staticnumber=0)
exten = 2231,n,Set(counter=1)
exten = 2231,n,While($[${counter}${staticnumber}])
exten = 2231,n,NoOp(This part of the code should never run!)
exten = 2231,n,Set(counter=$[${counter}+1])
exten = 2231,n,EndWhile
exten = 2231,n,NoOp(This part of the code should be the only thing
that gets run!)
Console output from dialing 2231:
-- Executing Set(SIP/2203-c134, staticnumber=0) in new stack
-- Executing Set(SIP/2203-c134, counter=1) in new stack
-- Executing While(SIP/2203-c134, 0) in new stack
-- Executing NoOp(SIP/2203-c134, This part of the code should
never run!) in new stack
-- Executing Set(SIP/2203-c134, counter=2) in new stack
-- Executing EndWhile(SIP/2203-c134, ) in new stack
-- Executing NoOp(SIP/2203-c134, This part of the code should
be the only thing that gets run!) in new stack
*CLI show version
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux
on 2005-09-03 23:27:34 UTC
JT
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Re: [Asterisk-Users] Unexpected results with While and EndWhile applications
On 9/5/05, John Todd [EMAIL PROTECTED] wrote:
I seem to be having a conceptual problem with the While and
EndWhile applications. It seems that on the first cycle, even if
the result of the While is false that the enclosed applications
will get run. Is this expected? It seems to be counter-intuitive,
but I don't know what the intent of the While routines is. I could
of course put a GotoIf before the While loop to check to ensure
that the first expression is true before entry into the While loop,
but that seems redundant and ugly since the while point of While and
EndWhile is to avoid the inelegance of GotoIf, I thought.
If anyone can't come up with a better explanation, I'll open a ticket
on this but I'd like to first make sure that this behavior is not
expected.
exten = 2231,1,Set(staticnumber=0)
exten = 2231,n,Set(counter=1)
exten = 2231,n,While($[${counter}${staticnumber}])
Put A space around the operator, like this
exten = 2231,n,While($[${counter} ${staticnumber}])
This should help it.
exten = 2231,n,NoOp(This part of the code should never run!)
exten = 2231,n,Set(counter=$[${counter}+1])
exten = 2231,n,EndWhile
exten = 2231,n,NoOp(This part of the code should be the only thing
that gets run!)
Console output from dialing 2231:
-- Executing Set(SIP/2203-c134, staticnumber=0) in new stack
-- Executing Set(SIP/2203-c134, counter=1) in new stack
-- Executing While(SIP/2203-c134, 0) in new stack
-- Executing NoOp(SIP/2203-c134, This part of the code should
never run!) in new stack
-- Executing Set(SIP/2203-c134, counter=2) in new stack
-- Executing EndWhile(SIP/2203-c134, ) in new stack
-- Executing NoOp(SIP/2203-c134, This part of the code should
be the only thing that gets run!) in new stack
*CLI show version
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux
on 2005-09-03 23:27:34 UTC
JT
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Re: [Asterisk-Users] goto() results in invalid extension
Michael Rowley wrote: Hello, Trying to rewrite my dialplan, and it is a little complex. But my extensions.conf redirection works, but the referred to contexts result in invalid extension Please help... I have the extension set to 's' currently, but originally it was 6044. The change didn't make any difference. Still receive the invalid extension message. Michael [main] ; 6044 main office line. exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1) exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1) exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1) exten = 6044,4,Goto(afterhours,1) Only when you forget to put the correct parameters on Goto. The least line above says go to the extension named afterhours with priority 1 ___ 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] goto() results in invalid extension
Yeah, yeah, rtfm, I know... :) This is after several editions. The original was trying to refer to the incoming DID of 6044, so was programatically correct at Goto(afterhours,6044,1) Plus, the GotoifTime's should work. Actually, the afterhours should work, if I have an extension of 'afterhours' in the current context. I didn't notice this until you pointed it out... but it was correct in previous revisions... and I was tyring to test it during the week, when the previous Goto's should have taken precidence... and they _all_ failed. Any reason why the rest should give me 'invalid extension'? Michael On Nov 1, 2004, at 9:34 AM, Eric Wieling wrote: Michael Rowley wrote: Hello, Trying to rewrite my dialplan, and it is a little complex. But my extensions.conf redirection works, but the referred to contexts result in invalid extension Please help... I have the extension set to 's' currently, but originally it was 6044. The change didn't make any difference. Still receive the invalid extension message. Michael [main] ; 6044 main office line. exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1) exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1) exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1) exten = 6044,4,Goto(afterhours,1) Only when you forget to put the correct parameters on Goto. The least line above says go to the extension named afterhours with priority 1 ___ 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
Re: [Asterisk-Users] goto() results in invalid extension
Michael Rowley wrote: Yeah, yeah, rtfm, I know... :) This is after several editions. The original was trying to refer to the incoming DID of 6044, so was programatically correct at Goto(afterhours,6044,1) Plus, the GotoifTime's should work. Actually, the afterhours should work, if I have an extension of 'afterhours' in the current context. I didn't notice this until you pointed it out... but it was correct in previous revisions... and I was tyring to test it during the week, when the previous Goto's should have taken precidence... and they _all_ failed. Any reason why the rest should give me 'invalid extension'? exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1) exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1) exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1) exten = 6044,4,Goto(afterhours,1) It shouldn't make a difference, but you are altering terminators half way through the line...i.e. try exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours|s|1) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] goto() results in invalid extension
Hello,
Trying to rewrite my dialplan, and it is a little complex. But my
extensions.conf redirection works, but the referred to contexts result
in invalid extension Please help... I have the extension set to 's'
currently, but originally it was 6044. The change didn't make any
difference. Still receive the invalid extension message.
Michael
[main]
; 6044 main office line.
exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1)
exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1)
exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1)
exten = 6044,4,Goto(afterhours,1)
[officehours]
exten =s,2,Dial(${RECEPTION},15,r)
exten =s,3,Dial(${STAFF},10,r)
exten =s,4,Answer
exten =s,5,NoOp,${CALLERID}
exten =s,10,ResponseTimeout(5)
exten =s,16,Background(thankyouwmfm)
exten =s,17,Background(911)
exten =s,18,Background(mdorhospital)
exten =s,19,Background(nooneavail2answer)
exten =s,20,Background(appointmentdesk)
exten =s,21,Background(press1)
exten =s,22,Background(nursemessage)
exten =s,23,Background(press2)
exten =s,24,Goto(s,10)
include = menu
[lunch]
exten =s,1,Answer
exten =s,2,ResponseTimeout(5)
exten =s,6,Background(thankyouwmfm)
exten =s,7,Background(911)
exten =s,8,Background(mdorhospital)
exten =s,9,Background(closed4lunch)
exten =s,10,Background(reopenatoneoclk)
exten =s,11,Background(pleasecallbackatthattime)
exten =s,12,Goto(s,2)
include = menu-after-hours
[afterhours]
exten =s,3,Answer
exten =s,4,NoOp,${CALLERID}
exten =s,5,ResponseTimeout(5)
exten =s,6,Background(thankyouwmfm)
exten =s,7,Background(911)
exten =s,9,Background(nowclosed)
exten =s,8,Background(mdorhospital)
exten =s,10,Background(patientoptions)
exten =s,11,Background(appointmentdesk)
exten =s,12,Background(press1)
exten =s,13,Background(nursemessage)
exten =s,14,Background(press2)
exten =s,15,Background(4hoursOfop)
exten =s,16,Background(press3)
exten =s,17,Background(physicianoncall)
exten =s,18,Background(press4)
exten =s,20,Goto(s,5)
include = menu-after-hours
[on-call]
exten =s,1,ResponseTimeout(5)
exten =s,2,Playback(oncallmdline)
exten =s,3,Playback(nonurgentmatters)
exten =s,4,Playback(mdfee10)
exten =s,5,Playback(feewaived)
exten =s,6,Playback(voicemailphysoncall)
exten =s,7,Background(speakoncallmd)
exten =s,8,Background(press9)
exten =s,9,Background(otherwise)
exten =s,10,Background(press3)
exten =s,11,Background(return2nurse)
exten =s,12,Goto(s,1)
include = menu
;---
; Menu System.
;---
[menu] ; menu used when people are supposed to be here.
exten =1,1,Macro(sipexten,100,10)
exten =1,2,Voicemail(u100)
exten =1,3,Hangup
exten =2,1,Macro(sipexten,110,10)
exten =2,2,Voicemail(u110)
exten =2,3,Hangup
exten =3,1,Playback(hoursofop)
exten =3,2,Goto(main,s,1)
exten =4,1,Goto(on-call,s,1)
exten =9,1,Playback(pbx-transfer)
exten =9,2,Dial(${ONCALL})
exten =9,3,Hangup
include = invalid
[menu-after-hours] ; when the office is likely empty.
;exten =1,1,Macro(sipexten,100,10)
exten =1,2,Voicemail(u100)
exten =1,3,Hangup
;exten =2,1,Macro(sipexten,110,10)
exten =2,2,Voicemail(u110)
exten =2,3,Hangup
exten =3,1,Playback(hoursofop)
exten =3,2,Goto(main,1)
exten =4,1,Goto(on-call,s,1)
exten =9,1,Playback(pbx-transfer)
exten =9,2,Dial(${ONCALL})
exten =9,3,Hangup
include = invalid
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RE: [Asterisk-Users] goto() results in invalid extension
[EMAIL PROTECTED] wrote:
[main]
; 6044 main office line.
exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1)
exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1)
exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1)
exten = 6044,4,Goto(afterhours,1)
Your numbering sequence is incorrect, spot the difference:
exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1)
exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1)
exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1)
exten = 6044,4,Goto(afterhours,1)
snip
[afterhours]
exten =s,3,Answer
exten =s,4,NoOp,${CALLERID}
exten =s,5,ResponseTimeout(5)
exten =s,6,Background(thankyouwmfm)
There's nowhere to go with (afterhours,1). I'd try to Goto(afterhours,s,3)
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11 3011 VH Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
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[Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this script
has only first result although i have many in my database. this is the part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into the db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this script
has only first result although i have many in my database. this is the part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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winmail.dat
Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..
i really don't know what to put in keyword and data... but i have something
like 4 datas in my sip table
1234,account,sip1,0
1235,account,sip2,0
1236,user,sip3,0
1236,peer,sip3,0
what do u mean by db schema???
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 9:57 AM
Subject: RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into the
db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i
have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this
script
has only first result although i have many in my database. this is the
part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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RE: [Asterisk-Users] no results.
the database schema is the table and it's associated columns. did you use the create
table script in the header of the pl file? basically, for each of your sip entries,
they would be broken down per line. so if your sip.conf entry looks like this:
[1234]
type=friend
username=1234
secret=blah
nat=yes
host=dynamic
canreinvite=no
qualify=200
defaultip=192.168.0.4
your entries in the mysql database would be like this:
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234', 'type', 'friend',
'0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234', 'username',
'1234', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234', 'secret', 'blah',
'0');
and so on. the 'flags' column allows you to disable an entry without deleting
the entry completely. Hope this helps!
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 11:31 PM
To: [EMAIL PROTECTED]
Cc:
Subject: Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..
i really don't know what to put in keyword and data... but i have something
like 4 datas in my sip table
1234,account,sip1,0
1235,account,sip2,0
1236,user,sip3,0
1236,peer,sip3,0
what do u mean by db schema???
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 9:57 AM
Subject: RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into the
db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i
have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this
script
has only first result although i have many in my database. this is the
part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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winmail.dat
Re: [Asterisk-Users] no results.
ok i guess,, we also have to put
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'account', '1234', '0');
at the beginning.. its working now
thankx
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 10:27 AM
Subject: RE: [Asterisk-Users] no results.
the database schema is the table and it's associated columns. did you use
the create table script in the header of the pl file? basically, for each
of your sip entries, they would be broken down per line. so if your
sip.conf entry looks like this:
[1234]
type=friend
username=1234
secret=blah
nat=yes
host=dynamic
canreinvite=no
qualify=200
defaultip=192.168.0.4
your entries in the mysql database would be like this:
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'type', 'friend', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'username', '1234', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'secret', 'blah', '0');
and so on. the 'flags' column allows you to disable an entry
without deleting the entry completely. Hope this helps!
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 11:31 PM
To: [EMAIL PROTECTED]
Cc:
Subject: Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..
i really don't know what to put in keyword and data... but i have
something
like 4 datas in my sip table
1234,account,sip1,0
1235,account,sip2,0
1236,user,sip3,0
1236,peer,sip3,0
what do u mean by db schema???
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 9:57 AM
Subject: RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into
the
db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i
have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this
script
has only first result although i have many in my database. this is the
part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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RE: [Asterisk-Users] no results.
Title: Message
oops! I forgot on important one! you have to have at the
minimum this entry:
INSERT INTO `sip` (`id`, `keyword`,
`data`, `flags`) VALUES ('1234',account', '1234', '0');
you'll
notice that the 'id' column is an integer. so you have to keep it
numeric. some people like to have their sip extensions be alphanumeric, so
in order to accommodate that there is the id field. this is independent of
anything in your normal sip.conf file. but all entries for each sip.conf
entry must have the same 'id' set. so account is actually the name of the
sip entry. I hope I didn't just make that very
unclear!
Sean
-Original Message-From: Sean Cheesman
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
CheesmanSent: Tuesday, January 06, 2004 11:43 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] no
results.
the database schema is the table and it's associated columns. did
you use the create table script in the header of the pl file? basically,
for each of your sip entries, they would be broken down per line. so if
your sip.conf entry looks like this:
[1234]type=friendusername=1234secret=blahnat=yeshost=dynamiccanreinvite=noqualify=200defaultip=192.168.0.4
your entries in the mysql database would be like this:
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'type', 'friend', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'username', '1234', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'secret', 'blah', '0');
and so on. the 'flags' column allows you to "disable" an entry
without deleting the entry completely. Hope this helps!
Sean
-Original Message- From: Chandra
[mailto:[EMAIL PROTECTED] Sent: Tue 1/6/2004 11:31 PM
To: [EMAIL PROTECTED] Cc:
Subject: Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..i
really don't know what to put in keyword and data... but i have
somethinglike 4 datas in my sip
table1234,account,sip1,01235,account,sip2,01236,user,sip3,01236,peer,sip3,0what
do u mean by db schema???- Original Message -From: "Sean
Cheesman" [EMAIL PROTECTED]To:
[EMAIL PROTECTED]Sent: Wednesday, January 07, 2004
9:57 AMSubject: RE: [Asterisk-Users] no results. have
you set up the db schema? and have you entered any sip data into
thedb? Sean -Original
Message- From: Chandra [mailto:[EMAIL PROTECTED]]
Sent: Tue 1/6/2004 10:57 PM To:
[EMAIL PROTECTED] Cc: Subject:
[Asterisk-Users] no results. i have been
working with the retrieve_sip_conf_from_mysql.pl file and ihave
set everything as required. but when i run this script i am
continuously getting the "no results" in my screen and
the file written by thisscript has only first result although i
have many in my database. this is thepart of this
script. my @resSet =
@{$result}; print
$#resSet; if (
$#resSet == -1 )
{
print "no
results\n";
exit;
} can any one tell me what is happening? and get rid of this
error? for those who have no clue.. this file is in the
/usr/src/asterisk directory... (asterisk source
diretory.) thanks, chandra
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Re: [Asterisk-Users] no results.
ok thats done as u said. i am getting
No sip accounts defined in sip
error now.
??
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 10:27 AM
Subject: RE: [Asterisk-Users] no results.
the database schema is the table and it's associated columns. did you use
the create table script in the header of the pl file? basically, for each
of your sip entries, they would be broken down per line. so if your
sip.conf entry looks like this:
[1234]
type=friend
username=1234
secret=blah
nat=yes
host=dynamic
canreinvite=no
qualify=200
defaultip=192.168.0.4
your entries in the mysql database would be like this:
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'type', 'friend', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'username', '1234', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'secret', 'blah', '0');
and so on. the 'flags' column allows you to disable an entry
without deleting the entry completely. Hope this helps!
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 11:31 PM
To: [EMAIL PROTECTED]
Cc:
Subject: Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..
i really don't know what to put in keyword and data... but i have
something
like 4 datas in my sip table
1234,account,sip1,0
1235,account,sip2,0
1236,user,sip3,0
1236,peer,sip3,0
what do u mean by db schema???
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 9:57 AM
Subject: RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into
the
db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i
have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this
script
has only first result although i have many in my database. this is the
part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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[Asterisk-Users] [HS] results testing asterisk with ISDN BRI look for help tounderstand configuring SIP with asterisk
configuration ISDN BRI card : ISDN Olitec PCI 128 (hisax gazel) internet connection by ISDN 64kb/s dynamic IP nom de domaine registered to : dyndns.org avec ddclient to register IP par ddclient asterisk (on internet gateway) configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf) logical telephone SIP SJPHONE on 2 local stations windows (i don't succeed to use telephon SIP X-lite with asterisk) testing résults with asterisk SJPHONE local - IVR asterisk : OK extern telephon (analogic) - SJPhone : OK SJphone - extern telephon : OK extern telephon - SJPHONE : OK local network SJPhone -local network SJPhone (with asterisk) OK configuration sjphone : Use Local OuntBound Proxy (selected) Proxy IP address 192,168,0,1 port 5060 caller ID : SIP station@domain.dyndns.org (stations défined dans /etc/asterisk/sip.conf) I don't understand what i have to make and set to communicate with external telephons SIP (Sjphone, X-lite, MS messenger ...) Must i have a SIP provider subscription, how to integrate this subscription with asterisk The purpose i have is to keep control with asterisk to tape, redirect, establish conference ... with communicates I am swimming with (english) documentation anglaise and i understand very badly asterisk system, my knowledge in system software an linux is too low But with your patient help, i am sure i'll reach thanks to help me ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [HS] results testing asterisk with ISDN BRI look for help to understand configuring SIP with asterisk
Hi, I don't understand what i have to make and set to communicate with external telephons SIP (Sjphone, X-lite, MS messenger ...) Must i have a SIP provider subscription, how to integrate this subscription with asterisk Do you mean internally i.e. Sjphone, X-lite, MS messenger phones on your pc's or other people - out there on the net? You could take a look at my guide - it may help explain things (then again it may not) http://www.automated.it/guidetoasterisk.htm I recently had to move hosting co's, just noticed the one I moved was old!! I've updated it... I am swimming with (english) documentation anglaise You're lucky, I'm English and I have trouble speaking it! HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
