[asterisk-users] How to count ongoing calls from the dialplan
Hi, When I need to route calls depending on the number of (incoming and outgoing) calls a SIP device is currently handling, I mostly use function SIPPEER and its curcalls option. I can read and there references to function GROUP for the same usage, but I intuitively thought that though this method also applies to non-SIP devices and a large range of asterisk versions, it would require more work from me to tune my dialplan. Which method would you recommend ? Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which port should be open for asterisk communication
Hi List, Please tell me which ports should be required open for communication with asterisk. like 5060 for sip calls, 4569 for IAX, 10,000 to 20,000.. Apart from these ports what else is required ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which port should be open for asterisk communication
Hi, That depends on what else your asterisk is doing i.e if an AMI-based code is running then AMI port needs to be open as well. It also depends what other appliactions are running on asterisk-box which require port opening i.e apache or mysql etc. Regards, Sammy On Mon, Dec 12, 2011 at 3:21 PM, virendra bhati virbh...@gmail.com wrote: Hi List, Please tell me which ports should be required open for communication with asterisk. like 5060 for sip calls, 4569 for IAX, 10,000 to 20,000.. Apart from these ports what else is required ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Atxfer for the calling party
Nothing?
On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
Hi There,
I'm still having this problem, Does somebody know what can be
happening?
Regards.
On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
Hello,
The exten is the parameter passed to the macro, which contains
the sip device name. I'll change the name to another less confusing.
* Alexandre, também sou brasileiro hehe, notei que você já escreveu
um livro sobre asterisk, será que você poderia me ajudar com esse
problema? Já tem alguns dias que estou na luta aqui hehe.
On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller wrote:
You're using ${exten} inside your macro, you should use ${EXTEN}.
--
Atenciosamente,
ALEXANDRE KELLER
http://twitter.com/alexandrekeller
http://www.facebook.com/alexandre.keller.BR
Dinheiro é a consequência de um trabalho bem feito e não o motivo
para se fazer um bom trabalho.
P Antes de imprimir pense em seu compromisso com o Meio Ambiente.
On 11/11/2011, at 08:38, Antonio Modesto wrote:
On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas wrote:
It can have to do with either the telephones dial plan or the
context in the Asterisk dial plan combined with your
features.conf settings.
I noticed that my problem occurs when i use a macro to dial sip
devices, my dialplan is like this:
- Each sip device has its own context
- This context includes the outgoing call contexts that this
extension can use for making calls and includes a context called
ramais, which has the dial plan to call another extensions, it
uses a macro to do this.
Here is the configuration for my extension modesto :
# sip.conf
[modesto](default_extension)
username=modesto
context=modesto
callerid=modesto 106
callgroup=4
pickupgroup=4
# Default extension template
type=friend
dtmfmode=auto
host=dynamic
disallow=all
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
canreinvite=yes
qualify=no
callcounter=yes
# context for SIP/modesto
context modesto {
includes {
vivo;
tim;
oi;
claro;
vivoddd;
timddd;
oiddd;
claroddd;
embratel;
embratel2;
};
includes {
ramais;
};
};
# Although the problem is occurring also for others contexts
included, i'll show only the ramais context, which is used to
call local extensions:
context ramais {
101 = dial_sip(suporte1);
102 = dial_sip(suporte2);
103 = dial_sip(suporte3);
105 = dial_sip(suporte05);
106 = dial_sip(modesto);
107 = dial_sip(gustavo);
108 = dial_sip(pauloh);
109 = dial_sip(fernanda);
111 = dial_sip(marcos);
112 = dial_sip(thiago);
115 = dial_sip(helder);
116 = dial_sip(atendimento01);
117 = dial_sip(atendimento03);
118 = dial_sip(atendimento02);
119 = dial_sip(marlon);
120 = dial_sip(suporteemp);
122 = dial_sip(telemais);
123 = dial_sip(casagustavo);
127 = dial_sip(manutencao);
128 = dial_sip(guilherme);
129 = dial_sip(marcelo);
130 = dial_sip(rafael);
132 = dial_sip(netita2);
133 = dial_sip(unotel);
};
If I use the Dial() application instead of this macro, it works
well. I noticed that when I use the macro and try to transfer a
call (The problem occurs only for the calling party, the called
party can do transfers with no problems), asterisk tries to find
the extension in the macro-name context and of course, there
is no dialplan to call the extensions there.
Here is the dial_sip macro:
macro dial_sip(exten) {
Verbose(2,== Chamando a MACRO dial_sip - ponto 1
macros.ael ==);
Verbose(4, Macro dial_sip iniciada.);
ChanIsAvail(SIP/${exten});
Verbose(2,== ${AVAILORIGCHAN});
if (${AVAILORIGCHAN} != )
{
Verbose(4, SIP/${exten} parece estar
disponivel, vou disca-lo agora.);
Set(FromExt=${CALLERID(num)});
System(/bin/sh /var/spool/asterisk/calllog/log.sh SIP/${FromExt}
SIP/${exten} SIP-TO-SIP);
Verbose(4, System status:
${SYSTEMSTATUS});
Dial(SIP/${exten},${SIP_DIAL_TIMEOUT},Ttr);
Hangup();
}
else
[asterisk-users] Asterisks Statistics (Albert)
Hi Albert, we currently use QueueMetrics to monitor and report on call center statistics... regards Anthony -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which port should be open for asterisk communication
Hi Sammy, Thanks for fastest reply. I to know just for calling time which port's should asterisk need to be open only On Mon, Dec 12, 2011 at 4:03 PM, Sammy Govind govoi...@gmail.com wrote: Hi, That depends on what else your asterisk is doing i.e if an AMI-based code is running then AMI port needs to be open as well. It also depends what other appliactions are running on asterisk-box which require port opening i.e apache or mysql etc. Regards, Sammy On Mon, Dec 12, 2011 at 3:21 PM, virendra bhati virbh...@gmail.comwrote: Hi List, Please tell me which ports should be required open for communication with asterisk. like 5060 for sip calls, 4569 for IAX, 10,000 to 20,000.. Apart from these ports what else is required ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Populate CDR issues
Danny,
Why would you think this is a circumvent? I'm using a nice feature of 1.8
where I can create any CDR field I like and populate it by using the
CDR(fieldname) function. While all other fields that I created are populated
properly (however before the 'dial' commences) it seems like at this point of
the dial plan the CDR is closed for editing even though I configured
endbeforehexten=no in my cdr.conf.
It might be related to issue ASTERISK-18875 as suggested by Daniel (Vol. 89
issue 8, topic 9).
I'll be happy to know if someone has a different knowledge on the subject,
otherwise I'll simply follow ASTERISK-18875. My problem with this issue is that
it is defined as low importance which means that it will probably take long to
handle if at all...
Harel
**
Message: 4
Date: Tue, 6 Dec 2011 07:29:54 -0600
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Populate CDR issues
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: 009801ccb41b$24417cd0$6cc47670$@debsinc.com
Content-Type: text/plain; charset=us-ascii
IMO you are trying to circumvent basic Asterisk functionality. It's your CDR
so you can do what you want with it - I think the answer to this is to populate
another DB with the live call data, then update the CDR from that after the
call has ended (perhaps a daemon).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harel Cohen
Sent: Tuesday, December 06, 2011 3:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Populate CDR issues
Hello Everyone,
I didn't get a reply to my problem below so I'm posting again just in case
someone who might be able to help missed my previous post.
Thank You.
*
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel. I've tried using the option
'U' with my dial command (execute subroutine for called channel after called
channel answered but before the call is bridged). While this throws the correct
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC
and the table therein contains the relevant fields.
This is the console with 'very-verbose' output for the 'Dial' application where
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP
192.168.20.226. My comments added prefixed by ** and on separate line:
** channel here is source channel: SIP/office_Admin2-0015
[Dec 1 12:14:31] -- Executing [316@InternalDP:5]
Dial(SIP/office_Admin2-0015,
SIP/office_ServerRoom,,FgU(jump2SetVar)) in new stack
[Dec 1 12:14:31] == Using UDPTL CoS mark 5
[Dec 1 12:14:31] == Using SIP RTP CoS mark 5
[Dec 1 12:14:31] -- Called SIP/office_ServerRoom
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:33] -- SIP/office_ServerRoom-0016 answered
SIP/office_Admin2-0015
** from here the channel is the destination channel:
SIP/office_ServerRoom-0016
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:1]
Gosub(SIP/office_ServerRoom-0016, SetVar,postdial,1) in new stack
** This is how I obtain channel information:
** exten =
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peer
name)},port)}) ; resulting format: a.b.c.d:port
** same = n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same = n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial@SetVar:1]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutsigip)=192.168.20.226:5065) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:2]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutmediaip)=192.168.20.226:23008) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:3]
Set(SIP/office_ServerRoom-0016, CDR(chanoutcodec)=g729) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:4]
Goto(SIP/office_ServerRoom-0016, endsub,1) in new stack
[Dec 1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec 1 12:14:33] -- Executing [endsub@SetVar:1]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:2]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1]
NoOp(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Auto fallthrough, channel
'SIP/office_ServerRoom-0016'
[asterisk-users] How to see initiall dialled extension in CDR records ?
Hi guys,
I have following problem. For statistical reasons I need to know what
was initiall number dialled by customer. I have 2 premium numbers, for
which customers are billed differently per minute. But in my CDR table i
can see only last dialled extension from voice menu. In this example it
shows me that customer (077XXX) was billed 293 seconds but i am not
seeing which number he dialed. Only what i can see is last destination
he choosed from voice menu which was '1'.
What shall I do to see which number was dialled initially?
ps. Original numer was X-ed.
calldate |clid|src | dst |
dcontext |channel | dstchannel | lastapp
| lastdata | duration | billsec | disposition
| amaflags | accountcode | uniqueid| userfield
-+++-++++++--+-+-+--+-+---+---
2011-12-12 15:11:26 | 077XXX | 077XXX | 1 |
stories-en-options | DAHDI/i1/077XXX-13 || Playback |
custom/l_stories/en/en_bible | 293 | 293 | ANSWERED|
3 | | 1323691886.91 |
Below is part on my extensioans file and voice menu structure
[incoming-calls-from-e1-span1]
exten = 902000111,1,Verbose(Call is comming ${EXTEN})
exten = 902000111,n,Goto(voiceservices_menu,2666,1)
exten = 902000222,1,Verbose(Call is comming ${EXTEN})
exten = 902000222,n,GotoIfTime(08:00-22:59,mon-sun,*,*?supportmenu,2555,1)
exten = 902000222,n,Playback(custom/l_line/callcenter-closed)
exten = 902000222,n,Goto(voiceservices_menu,2666,1)
[voiceservices_menu]
exten = 2666,1,Set(EXT=${EXTEN})
exten = 2666,n,Verbose(EXT - ${EXT})
exten = 2666,n,Goto(voiceservices_options,s,1)
[voiceservices_options]
exten = s,1,Background(custom/l_stories/swelcome)
exten = s,n,WaitExten(15) ;wait 10 sec
exten = 1,1,Verbose(Customer pressed key 1)
exten = 1,n,Goto(storiesmenu-lu,${EXT},1) ;if customer choosed 1 jump
to LU lang
exten = 2,1,Verbose(Customer pressed key 2)
exten = 2,n,Goto(storiesmenu-en,${EXT},1) ;if customer choosed 1 jump
to EN lang
exten = i,1,Playback(invalid)
exten = i,n,Goto(voiceservices_menu,${EXT},1)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Goto(voiceservices_menu,${EXT},1)
[storiesmenu-lu]
exten = 2666,1,Verbose(Entering Luganda menu of stories)
exten = 2666,n,Verbose(EXT = ${EXT})
exten = 2666,n,Goto(storiesmenu-lu-options,s,1)
[storiesmenu-lu-options]
exten = s,1,Background(custom/l_stories/lu_service_menu)
exten = s,n,Verbose(EXT = ${EXT})
exten = s,n,WaitExten(10) ;wait 10 sec
; if customer pressed '1'
exten = 1,1,Goto(stories-lu,${EXT},1)
; if customer pressed '2'
exten = 2,1,Playback(custom/l_stories/lu/lu_bible1)
exten = 2,n,Wait(2)
exten = 2,n,Playback(custom/l_stories/lu/lu_bible2)
exten = 2,n,Wait(2)
exten = 2,n,Playback(custom/l_stories/lu/lu_bible3)
exten = 2,n,Wait(2)
exten = 2,n,Playback(custom/l_stories/lu/lu_bible4)
exten = 2,n,Wait(2)
exten = 2,n,Playback(custom/l_stories/lu/lu_bible5)
exten = 2,n,Wait(2)
exten = 2,n,Goto(storiesmenu-lu,${EXT},1)
Thanks for your feedback!
Regards,
Albert
--
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Re: [asterisk-users] What version to upgrade to...?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 12/11/2011 10:59 PM, Mike Diehl wrote: Should I go to 1.8.x? Or all the way up to 10.x? This is a production system and I can't afford to be testing code. The 1.8 series is the current LTS release. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFO5gQWCFu3bIiwtTARArU9AJ9/ZWb5uyjqjBFKqyjZa4X1+2fC+wCfVHP1 KY1D7w1siMJtCd1Ktxffwy4= =PbAV -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySql Custom CDR issues
hello , I have been working hard to solve the issue of custom CDR in the Asterik with Mysql but in vain. I searched google for complete 2 hours but in vain. What i want to achieve is CDR(customcolumn)=anyvaluealthough we can achieve it through other ways like making a script that runs when a call ends and modify the cdr and insert in custom value BUT is there any way to make this work ? Thank you in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version to upgrade to...?
2011/12/12, Mike Diehl mdi...@diehlnet.com: Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd also like to get to a new enough version that my patches would be useful to the community. Should I go to 1.8.x? Or all the way up to 10.x? This is a production system and I can't afford to be testing code. -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm roughly wondering the same thing. If I may add, I read few weeks ago, that Asterisk's SNMP features required asterisk to run as root. If any of asterisk 1.8 or 10 version could solve this limitation, that would convince to dive in that one. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySql Custom CDR issues
Are you using FreePBX or another packaged Asterisk? Sent from my iPhone 4S On Dec 12, 2011, at 9:23 AM, silent sayz silent.s...@gmail.com wrote: hello , I have been working hard to solve the issue of custom CDR in the Asterik with Mysql but in vain. I searched google for complete 2 hours but in vain. What i want to achieve is CDR(customcolumn)=anyvaluealthough we can achieve it through other ways like making a script that runs when a call ends and modify the cdr and insert in custom value BUT is there any way to make this work ? Thank you in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help needed for chan_ss7 for Digium device
Hi All, I have installed centos 5.6 32 bit on xeon server and i have also installed latest version of asterisk 1.6 and dahdi as well. I want to install chan_ss7 for this server and I want to know about the following device. Digium TE420B I dont know much about the configuration files for Digium TE420B. Can anybody provide me required ss7.conf file and also provide dahdi configuration which is needed for this device. Thanks you so much in advance!! Thanks, Max Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySql Custom CDR issues
Hi! I am using Asterisk 1.6.2.20 with elastix -- Forwarded message -- From: Robert-IPhone rhuddles...@gmail.com Date: Mon, Dec 12, 2011 at 5:45 PM Subject: Re: [asterisk-users] MySql Custom CDR issues To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Are you using FreePBX or another packaged Asterisk? Sent from my iPhone 4S On Dec 12, 2011, at 9:23 AM, silent sayz silent.s...@gmail.com wrote: hello , I have been working hard to solve the issue of custom CDR in the Asterik with Mysql but in vain. I searched google for complete 2 hours but in vain. What i want to achieve is CDR(customcolumn)=anyvaluealthough we can achieve it through other ways like making a script that runs when a call ends and modify the cdr and insert in custom value BUT is there any way to make this work ? Thank you in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySql Custom CDR issues
Hello, I installed asterisk addons package and it is solved. Thank you. On Mon, Dec 12, 2011 at 5:55 PM, silent sayz silent.s...@gmail.com wrote: Hi! I am using Asterisk 1.6.2.20 with elastix -- Forwarded message -- From: Robert-IPhone rhuddles...@gmail.com Date: Mon, Dec 12, 2011 at 5:45 PM Subject: Re: [asterisk-users] MySql Custom CDR issues To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Are you using FreePBX or another packaged Asterisk? Sent from my iPhone 4S On Dec 12, 2011, at 9:23 AM, silent sayz silent.s...@gmail.com wrote: hello , I have been working hard to solve the issue of custom CDR in the Asterik with Mysql but in vain. I searched google for complete 2 hours but in vain. What i want to achieve is CDR(customcolumn)=anyvaluealthough we can achieve it through other ways like making a script that runs when a call ends and modify the cdr and insert in custom value BUT is there any way to make this work ? Thank you in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version to upgrade to...?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, December 12, 2011 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What version to upgrade to...? 2011/12/12, Mike Diehl mdi...@diehlnet.com: Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd also like to get to a new enough version that my patches would be useful to the community. Should I go to 1.8.x? Or all the way up to 10.x? This is a production system and I can't afford to be testing code. -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm roughly wondering the same thing. If I may add, I read few weeks ago, that Asterisk's SNMP features required asterisk to run as root. If any of asterisk 1.8 or 10 version could solve this limitation, that would convince to dive in that one. I'm wondering if the bind 161 as root statement is a mis-statement or if not, maybe somebody like Tzafir can explain why since none of the other Asterisk binds require root access (this message is still in 10.0-rc3). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version to upgrade to...?
On Mon, Dec 12, 2011 at 12:26 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, December 12, 2011 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What version to upgrade to...? 2011/12/12, Mike Diehl mdi...@diehlnet.com: Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd also like to get to a new enough version that my patches would be useful to the community. Should I go to 1.8.x? Or all the way up to 10.x? This is a production system and I can't afford to be testing code. -- Take care and have fun, Mike Diehl. I'm roughly wondering the same thing. If I may add, I read few weeks ago, that Asterisk's SNMP features required asterisk to run as root. If any of asterisk 1.8 or 10 version could solve this limitation, that would convince to dive in that one. I'm wondering if the bind 161 as root statement is a mis-statement or if not, maybe somebody like Tzafir can explain why since none of the other Asterisk binds require root access (this message is still in 10.0-rc3). Any port under 1024 is a reserved system port and normally can only be opened by root. 161 is under 1024, thus root. You can run snmp on other ports if you really want to. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version to upgrade to...?
Asterisk uses libcap to do root-like things when running as non-root. Setting the DSCP/QoS value of packets requires root access, but Asterisk seems to manage just fine using libcap (not libpcap, that is different). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Monday, December 12, 2011 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What version to upgrade to...? On Mon, Dec 12, 2011 at 12:26 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, December 12, 2011 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What version to upgrade to...? 2011/12/12, Mike Diehl mdi...@diehlnet.com: Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd also like to get to a new enough version that my patches would be useful to the community. Should I go to 1.8.x? Or all the way up to 10.x? This is a production system and I can't afford to be testing code. -- Take care and have fun, Mike Diehl. I'm roughly wondering the same thing. If I may add, I read few weeks ago, that Asterisk's SNMP features required asterisk to run as root. If any of asterisk 1.8 or 10 version could solve this limitation, that would convince to dive in that one. I'm wondering if the bind 161 as root statement is a mis-statement or if not, maybe somebody like Tzafir can explain why since none of the other Asterisk binds require root access (this message is still in 10.0-rc3). Any port under 1024 is a reserved system port and normally can only be opened by root. 161 is under 1024, thus root. You can run snmp on other ports if you really want to. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version to upgrade to...?
Leads to the next question - has anybody tested SNMP using non-root Asterisk? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, December 12, 2011 9:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What version to upgrade to...? Asterisk uses libcap to do root-like things when running as non-root. Setting the DSCP/QoS value of packets requires root access, but Asterisk seems to manage just fine using libcap (not libpcap, that is different). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Monday, December 12, 2011 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What version to upgrade to...? On Mon, Dec 12, 2011 at 12:26 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, December 12, 2011 8:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What version to upgrade to...? 2011/12/12, Mike Diehl mdi...@diehlnet.com: Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd also like to get to a new enough version that my patches would be useful to the community. Should I go to 1.8.x? Or all the way up to 10.x? This is a production system and I can't afford to be testing code. -- Take care and have fun, Mike Diehl. I'm roughly wondering the same thing. If I may add, I read few weeks ago, that Asterisk's SNMP features required asterisk to run as root. If any of asterisk 1.8 or 10 version could solve this limitation, that would convince to dive in that one. I'm wondering if the bind 161 as root statement is a mis-statement or if not, maybe somebody like Tzafir can explain why since none of the other Asterisk binds require root access (this message is still in 10.0-rc3). Any port under 1024 is a reserved system port and normally can only be opened by root. 161 is under 1024, thus root. You can run snmp on other ports if you really want to. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version to upgrade to...?
On 12/12/2011 09:26 AM, Danny Nicholas wrote: I'm wondering if the bind 161 as root statement is a mis-statement or if not, maybe somebody like Tzafir can explain why since none of the other Asterisk binds require root access (this message is still in 10.0-rc3). This is accurate. Non-root users cannot bind ports =1024. There are ways around it, however. See setcap/CAP_NET_BIND_SERVICE at http://www.kernel.org/doc/man-pages/online/pages/man7/capabilities.7.html I haven't looked at the Asterisk code, but there may be changes necessary to disable that check, if this is enabled. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail and IMAP
Hello all, I have recently upgraded to version 1.8.7.2 and have started to see the following errors in the logs: [ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: SELECT failed [ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: must be in SELECTED state They are not having a detrimental effect on the storing of VMs in IMAP just filling up the logs quickly :) What do they mean please ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail and IMAP
On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote: Hello all, I have recently upgraded to version 1.8.7.2 and have started to see the following errors in the logs: From what version? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to see initiall dialled extension in CDR records ?
Please start a new thread for new conversations.
On Monday 12 Dec 2011, Albert wrote:
I have following problem. For statistical reasons I need to know what
was initiall number dialled by customer. I have 2 premium numbers,
for which customers are billed differently per minute. But in my CDR
table i can see only last dialled extension from voice menu. In this
example it shows me that customer (077XXX) was billed 293
seconds but i am not seeing which number he dialed. Only what i can
see is last destination he choosed from voice menu which was '1'.
What shall I do to see which number was dialled initially?
ps. Original numer was X-ed.
calldate |clid|src | dst |
dcontext |channel | dstchannel | lastapp
| lastdata | duration | billsec |
| disposition
|
| amaflags | accountcode | uniqueid| userfield
-+++-+---
-++++
+--+-+-+-
-+-+---+---
2011-12-12 15:11:26 | 077XXX | 077XXX | 1 |
stories-en-options | DAHDI/i1/077XXX-13 || Playback
| custom/l_stories/en/en_bible | 293 | 293 | ANSWERED|
3 | | 1323691886.91 |
Below is part on my extensioans file and voice menu structure
[incoming-calls-from-e1-span1]
exten = 902000111,1,Verbose(Call is comming ${EXTEN})
exten = 902000111,n,Goto(voiceservices_menu,2666,1)
exten = 902000222,1,Verbose(Call is comming ${EXTEN})
exten =
902000222,n,GotoIfTime(08:00-22:59,mon-sun,*,*?supportmenu,2555,1)
exten = 902000222,n,Playback(custom/l_line/callcenter-closed) exten
= 902000222,n,Goto(voiceservices_menu,2666,1)
If I understand your problem correctly, you have a few options:
1. Answer() the call in the 902000XXX extension. That will cause the
CDR to be written with that extension as the dst.
2. Save the called number in the CDR userfield in the 902000XXX
extension itself using something like:
exten = 902000111,n,Set(CDR(userfield)=${EXTEN})
There're probably many other (better) ways of achieving this too.
Regards,
-- Raj
--
Raj Mathur || r...@kandalaya.org || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves || http://schizoid.in || D17F
--
_
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Re: [asterisk-users] VoiceMail and IMAP
1.8.7.0 ... am using Zimbra as the backend IMAP storage. -- Thanks, Phil - Original Message - On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote: Hello all, I have recently upgraded to version 1.8.7.2 and have started to see the following errors in the logs: From what version? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rsrvd state and off hook dahdi issue
Hello again, Still with the same issue of dahdi off hock state. I changed from: dahdi - 2.2.0.2 to 2.6.0. astersik - 1.4.26.2 to 1.4.29. If I restart Asterisk, the problem persists. If I restart dahdi, and after start asterisk, the issue disappears for a while. Thus the problem seams to be from dahdi driver. The off hock situations dropped significantly when I connected the telephony server to an UPS. From 1 per month to 1 per year. Is there something on dahdi driver options that an can change to reduce this issue?? thanks, Alex 2009/12/21 Tzafrir Cohen tzafrir.co...@xorcom.com: On Mon, Dec 21, 2009 at 04:19:46PM +, Alexandre Rodrigues wrote: Hello all, I am still studing the problem and I'm now focusing on the Disconnect supervision asterisk issue. I which to obtain some feedback of the ideas I had to resolve this problem. I will set busydetect to yes and chage FXS from Kewl Start to Loop Start. Appreciate your Feedback. :) Asterisk can provide disconnect supervision by Kewl-start (power denial at the end of a call). Generally your side is the FXS. It does not decide if the line is open. The remote side does. Changing from loopstart to kewlstart means you don't give that hint to the remote party. Busydetect is something to be done on the FXO side. But if they support detecting a power denial, it would be better (faster and more reliable). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail and IMAP
Hmmm, just tried leaving a voicemail on a new mailbox where the imapfolder contains a space in the name and it errors; so that could be the cause of it all. Is is valid to have a space in an IMAP folder name ? -- Thanks, Phil - Original Message - 1.8.7.0 ... am using Zimbra as the backend IMAP storage. -- Thanks, Phil - Original Message - On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote: Hello all, I have recently upgraded to version 1.8.7.2 and have started to see the following errors in the logs: From what version? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail and IMAP
Generally speaking, no. if you need the space, use quotes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: Monday, December 12, 2011 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoiceMail and IMAP Hmmm, just tried leaving a voicemail on a new mailbox where the imapfolder contains a space in the name and it errors; so that could be the cause of it all. Is is valid to have a space in an IMAP folder name ? -- Thanks, Phil - Original Message - 1.8.7.0 ... am using Zimbra as the backend IMAP storage. -- Thanks, Phil - Original Message - On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote: Hello all, I have recently upgraded to version 1.8.7.2 and have started to see the following errors in the logs: From what version? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail and IMAP
Okay, though removing the space and reloading the module still throws the same error messages. -- Thanks, Phil - Original Message - Generally speaking, no. if you need the space, use quotes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: Monday, December 12, 2011 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoiceMail and IMAP Hmmm, just tried leaving a voicemail on a new mailbox where the imapfolder contains a space in the name and it errors; so that could be the cause of it all. Is is valid to have a space in an IMAP folder name ? -- Thanks, Phil - Original Message - 1.8.7.0 ... am using Zimbra as the backend IMAP storage. -- Thanks, Phil - Original Message - On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote: Hello all, I have recently upgraded to version 1.8.7.2 and have started to see the following errors in the logs: From what version? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to query Microsoft SQL server for caller-id source
Any suggestions from people who have done this before? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.comhttp://www.impalanetworks.com/ P: (505) 327-7300 F: (505) 327-7545 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to query Microsoft SQL server for caller-id source
Build Asterisk with ODBC support and then use the ODBC functions to do the database dips. On Dec 12, 2011, at 13:44, Douglas Mortensen d...@impalanetworks.com wrote: Any suggestions from people who have done this before? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO PCI Master Abort (was Re: Help! Logs filling up with errors!)
Well, I was wrong. The messages went away for a day, then came back. I am now rebuilding the server using an older motherboard. Hopefully that will solve the problem. On 12/9/2011 4:09 PM, Brent Davidson wrote: For the sake of posterity, I'm posting this solution: When I checked the server, the PnP OS option in the BIOS was set to No. Changing the option to Yes and rebooting has solved the problem. On 12/8/2011 10:58 AM, Brent Davidson wrote: I am still having issues with the error message Dec 7 14:25:06 servername kernel: FXO PCI Master abort filling up my log files. I've temporarily managed a work around by having the message log emptied every 10 minutes, but this is not a permanent solution. I expanded my google search to simple kernel pci master abort and came across a couple of sites recommending that the BIOS option PnP OS be set to No to solve these problems. Does anyone have any experience with this and think this might actually help? (The problem server is in a remote office and I don't want to make the 2 hour drive until I'm sure I have a solution.) Thanks, Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for partners to develop Asterisk Call Centre Applications - A call to investors and programmers
Hi everyone, We are looking to develop our own call centre application (HTML5, real-time, shiny GUI, easy access, etc...) on top of Asterisk. We are tired of using the proprietary packages that currently exist due to no proper support, expensive licensing costs, ugly GUIs, and closed nature of the applications. If there are any developers out there or those who want to partner with the project with us by investing please contact me off-list. Due it's complex nature, we have came to the conclusion that it's best to share costs and feedback to come up with an amazing call centre product. We do have an interest to release this as open-source and that is why I am posting to group. Thanks Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA with TCP/TLS support?
Hi List, Has anyone heard of an ATA device that supports TCP TLS? Not much comes up in searching, thought to check here for some device suggestions. TIA, Skyler -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA with TCP/TLS support?
The OBI110 ($45 USD) supports both of these. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler Sent: Monday, December 12, 2011 1:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ATA with TCP/TLS support? Hi List, Has anyone heard of an ATA device that supports TCP TLS? Not much comes up in searching, thought to check here for some device suggestions. TIA, Skyler -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Populate CDR issues
On Monday 12 December 2011 4:28:17 am Harel Cohen wrote: Danny, Why would you think this is a circumvent? I'm using a nice feature of 1.8 where I can create any CDR field I like and populate it by using the CDR(fieldname) function. While all other fields that I created are populated properly (however before the 'dial' commences) it seems like at this point of the dial plan the CDR is closed for editing even though I configured endbeforehexten=no in my cdr.conf. I agree, this is a perfectly valid use of the CDR. I do the same thing, btw. I think what you are seeing is that when your call starts, Asterisk creates a record, either in memory, or in a db transaction. When the call is torn down, the record is updated and committed to the db. The down-shot is that any changes you make to the db record get clobbered by this last update. I ended up making some of my updates in the hang-up phase via the h extension. See if that will do what you need. -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version to upgrade to...?
On Monday 12 December 2011 6:39:34 am Barry L. Kline wrote: On 12/11/2011 10:59 PM, Mike Diehl wrote: Should I go to 1.8.x? Or all the way up to 10.x? This is a production system and I can't afford to be testing code. The 1.8 series is the current LTS release. Barry Well, that clinches it for me. I'll get the latest 1.8 and get going. Thank you for your time. -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Configuration GUI Question
Hi All, There are a lot of existing projects for configuring Asterisk via GUI, so instead of trudging through them all, I'm hopeing to get some guidance. My architecture is ITSP based, we supply hosted PBX's to business customers. A few systems are dedicated PBX's but the majority are virtualized instances. We have been very successful managing the systems for our customers, not a lot of request for user portals or anything like that, so our PBX management consist of command line editing of Asterisk flat files and minimal sql database routines. We have built a few custom user portals for some of our customers using LAMP and have deployed a couple of other user web utilities, CDR search, Operator panel, Queue stats. I would like to implement a more standardized user portal for basic functions like call forward, voicemail password reset, user info change, queue member add/remove, ect I know there are many projects that could do just that, but most of what I'm finding are GUI's that take over the system and have conventions for many more configurable elements than I really need. Most are overkill for what I'm looking for. Because the majority of my PBX's are hosted virtual systems, overhead must be light. I would like to have a centralized management portal that pushes configs out to the PBX's but I'm not apposed to running a GUI on each PBX instance as long as it is light. I would like to be able to customize the interface, brand with my business logos, add or remove configuration elements. I kind of like the Digium Asterisk GUI but I'm just not real familiar with it, just test driving it a bit. What I do like about it is the flat file manipulation, no database needed. Any guidance is much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TLS bug in asterisk?
Hi folks. I've got a problem dialing with my new Snom M9 via TLS on asterisk 1.8.7.1 . Registration works like a charm - the phone becomes 'AVAILABLE'. An INVITE is responded by a 401 to be expected, but then asterisk closes the TLS connection upon the Snom's ACK. The authenticated INVITE the Snom sends immediately after the ACK meets a closed socket and merely triggers a TCP RST packet on asterisk's behalf. There's no ERROR or WARNING put out on the asterisk CLI. The only hint I get is asterisk complaining about not finding the CSeq anymore it used a second ago for the beginning of the dialog. I couldn't really figure a reason for asterisk to close the connection when it should wait for an authenticated INVITE, so I posted the problem details in the bug tracker under https://issues.asterisk.org/jira/browse/ASTERISK-19003?focusedCommentId=186012#comment-186012 I'd be very happy though, if someone could show me that this is not a bug, or how to work around it (I've got the Snom for about one more week, and then I'll have to decide whether to return it ;) ). Cheers, Gregor. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP MESSAGE outside calls - state of the art?
Hiya, SIP Messaging is implemented in asterisk-10... The only documentation I can find talks about a patch and is pretty old:http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging Like anything on voip-info.org it's horrible outdated. I think there's a documentation for the message-routing in docs Regards Jay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP MESSAGE outside calls - state of the art?
I think it only works with certain soft phones. I tried Aastra and it doesn't work. But EyeBeam soft phone receives messages. -Bruce On Mon, Dec 12, 2011 at 6:40 PM, Jay R. Worthington jayrworthing...@gmail.com wrote: Hiya, SIP Messaging is implemented in asterisk-10... The only documentation I can find talks about a patch and is pretty old:http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging Like anything on voip-info.org it's horrible outdated. I think there's a documentation for the message-routing in docs Regards Jay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help needed for chan_ss7 for Digium device
hello: you can refer this link: http://mirror.su.lt/voip-info/wiki/view/Asterisk+ss7+channels.html Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Mon, 12 Dec 2011 20:21:36 +0530 From: max.aster...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Help needed for chan_ss7 for Digium device Hi All, I have installed centos 5.6 32 bit on xeon server and i have also installed latest version of asterisk 1.6 and dahdi as well. I want to install chan_ss7 for this server and I want to know about the following device. Digium TE420B I dont know much about the configuration files for Digium TE420B. Can anybody provide me required ss7.conf file and also provide dahdi configuration which is needed for this device. Thanks you so much in advance!! Thanks, Max Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to query Microsoft SQL server for caller-id source
Is there a need to do it within the dialplan? If not you will find it easier to do it within AGI. Either connecting directly to the DB or in our case our developer build a web service which I make SOAP calls to. Nick. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Tuesday, 13 December 2011 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to query Microsoft SQL server for caller-id source Any suggestions from people who have done this before? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.comhttp://www.impalanetworks.com/ P: (505) 327-7300 F: (505) 327-7545 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
