[asterisk-users] AMD with analog lines - DIALSTATUS empty
Hello,
I would like to use AMD on outgoing calls using analog line. I tested
with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1
Other end is analog number behind another cisco/asterisk, also tested
calling a mobile number with the same result.
What I did: dial is done like exten = s,n,Dial(SIP/IP gw/dialed
number,,M(myMacro)), which tell Asterisk to execute myMacro when the
call is answered by calling party.
[myMacro]
exten = s,1,NoOP(Executed when call is answered)
same = n,AMD()
same = n,NoOp(Dial status=${DIALSTATUS})
same = n,NoOp(AMD status=${AMDSTATUS} cause=${AMDCAUSE})
same = n,MacroExit()
Problem is that [myMacro] is executed as soon as the call is going out
from the gw (cisco or linksys) and before called party answered.
DIALSTATUS is empty (should be ANSWER), AMDSTATUS=NOTSURE and
AMDCAUSE=TOOLONG-5000 which seems OK as DIALSTATUS isn't reliable.
The same dialplan using a SIP trunk is working as expected.
So question is, why, when using analog line, I dont get the right behavior.
Thanks for any hint
--
Daniel
--
_
-- 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 CLI Banner
Hello Experts, I want to know if there is any way to modify welcome banner on asterisk console when I connect using asterisk -r Thanks,Haider -- _ -- 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] Verbose only one context
You can do this sip set debug ip x.x.x.x On Wed, Mar 26, 2014 at 11:28 AM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi It's possible in Asterisk 1.8 enable verbose only in one context or extension? thanks Att, *Rafael dos Santos Saraiva* http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 -- _ -- 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] Asterisk CLI Banner
Am 28.03.2014 10:32, schrieb Haider Khalil: Hello Experts, I want to know if there is any way to modify welcome banner on asterisk console when I connect using asterisk -r Hi, did you compile asterisk from source? Take a look at main/asterisk.c (line 174 in asterisk v 11.5.1). I think you have to change it there manually and recompile it. -Thorsten- -- _ -- 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] Asterisk CLI Banner
Thank you so much everyone, I think I understand all I need for this. Haider To: asterisk-users@lists.digium.com From: kevin.lar...@pioneerballoon.com Date: Fri, 28 Mar 2014 11:18:16 -0500 Subject: Re: [asterisk-users] Asterisk CLI Banner asterisk-users-boun...@lists.digium.com wrote on 03/28/2014 10:51:13 AM: From: Haider Khalil haiderkha...@hotmail.com Thank you Thorsten Göllner. Matthew, What does violating license of Asterisk means ? Does it means I won't be able to use any commercial modules or asterisk commercially ? I thought it was open and anyone can change the code ? Haider I am neither a lawyer or a licensing expert, but the basics are that if you make such a change for your own internal use, you are probably fine. Example: You have 10 sites with Asterisk in them and at each site you have someone in your company who has to log into the CLI and do stuff. You change the header to pass them a message. This is probably (not going to guarantee this) going to be fine as it is not something you are releasing out into the wild nor are you selling it and making a profit from it. However, let's say you make a commercial project that uses Asterisk under the hood and you change the header to hide the fact that it uses Asterisk and that Digium has any ownership of the code. That would be not be okay in most, if not all cases. Basically, the code is open source, but it is still owned by Digium and they have specific rights that you have to be careful of in regards to licensing. If someone outside of your organization will ever be running the code you change, there are specific rules that have to be followed, including those that relate to releasing your changes to the code and to giving credit back to those who wrote the code your code is based on. Basically, Richard Kenner is spot on. If you are unclear, best to consult an attorney who specializes in this, especially if you are redistributing the altered code. -- _ -- 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] AMD with analog lines - DIALSTATUS empty
Le 28/03/2014 15:40, Richard Mudgett a écrit :
On Fri, Mar 28, 2014 at 4:01 AM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
Hello,
I would like to use AMD on outgoing calls using analog line. I
tested with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as
well as 11.8.1 Other end is analog number behind another
cisco/asterisk, also tested calling a mobile number with the same
result.
What I did: dial is done like exten = s,n,Dial(SIP/IP
gw/dialed number,,M(myMacro)), which tell Asterisk to execute
myMacro when the call is answered by calling party.
[myMacro]
exten = s,1,NoOP(Executed when call is answered)
same = n,AMD()
same = n,NoOp(Dial status=${DIALSTATUS})
same = n,NoOp(AMD status=${AMDSTATUS} cause=${AMDCAUSE})
same = n,MacroExit()
Problem is that [myMacro] is executed as soon as the call is going
out from the gw (cisco or linksys) and before called party
answered. DIALSTATUS is empty (should be ANSWER),
AMDSTATUS=NOTSURE and AMDCAUSE=TOOLONG-5000 which seems OK as
DIALSTATUS isn't reliable.
The same dialplan using a SIP trunk is working as expected.
So question is, why, when using analog line, I dont get the right
behavior.
Thanks for any hint
Analog lines don't have a reliable way to know when the far end
actually answers. Polarity
reversals could signal when the far end actually answers, but it isn't
normally available or
standardized. Thus, the line is considered answered when dialing is
complete.
OK, so it's a no way.
Thanks for your answer
--
Daniel
--
_
-- 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] Asterisk CLI Banner
On Friday 28 Mar 2014, Haider Khalil wrote: Thank you Thorsten Göllner. Matthew, What does violating license of Asterisk means ? Does it means I won't be able to use any commercial modules or asterisk commercially ? I thought it was open and anyone can change the code ? Haider Nothing, *unless* you plan to distribute it. Modifying a program you have legitimately acquired is Fair Dealing. The Law of the Land gives you the right to do that, even if the vendor restricts your exercise of that right in practice by withholding the Source Code. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Asterisk CLI Banner
Modifying a program you have legitimately acquired is Fair Dealing. The Law of the Land gives you the right to do that, even if the vendor restricts your exercise of that right in practice by withholding the Source Code. That is false. Modifying a program is creating a derivative work. As purchaser of a copyrighted item, you normally *do not* have that right. And this certainly may vary from jurisdiction to jurisdiction. For a (quite dated at this point) discussion of this issue from a US perspective, see http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157 The author is a recognized expert in software IP law. -- _ -- 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] Asterisk CLI Banner
Thank you Thorsten Göllner. Matthew, What does violating license of Asterisk means ? Does it means I won't be able to use any commercial modules or asterisk commercially ? I thought it was open and anyone can change the code ? Haider Date: Fri, 28 Mar 2014 10:00:42 -0500 From: mjor...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk CLI Banner On Fri, Mar 28, 2014 at 9:10 AM, Thorsten Göllner t...@ovm-group.com wrote: Am 28.03.2014 10:32, schrieb Haider Khalil: Hello Experts, I want to know if there is any way to modify welcome banner on asterisk console when I connect using asterisk -r Hi, did you compile asterisk from source? Take a look at main/asterisk.c (line 174 in asterisk v 11.5.1). I think you have to change it there manually and recompile it. Please note that modifying the banner in main/asterisk.c may cause you to violate the licensing of Asterisk, specifically Section 1 of the GPL (if you distribute the modified source in any fashion) and/or Section 2c. Unless you really know what you're doing with regards to software licensing, I would highly suggest not modifying the welcome message. -- Matthew Jordan Digium, Inc. | Engineering Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USACheck 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] AMD with analog lines - DIALSTATUS empty
On Fri, Mar 28, 2014 at 4:01 AM, Administrator TOOTAI ad...@tootai.netwrote:
Hello,
I would like to use AMD on outgoing calls using analog line. I tested with
SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1 Other
end is analog number behind another cisco/asterisk, also tested calling a
mobile number with the same result.
What I did: dial is done like exten = s,n,Dial(SIP/IP gw/dialed
number,,M(myMacro)), which tell Asterisk to execute myMacro when the call
is answered by calling party.
[myMacro]
exten = s,1,NoOP(Executed when call is answered)
same = n,AMD()
same = n,NoOp(Dial status=${DIALSTATUS})
same = n,NoOp(AMD status=${AMDSTATUS} cause=${AMDCAUSE})
same = n,MacroExit()
Problem is that [myMacro] is executed as soon as the call is going out
from the gw (cisco or linksys) and before called party answered. DIALSTATUS
is empty (should be ANSWER), AMDSTATUS=NOTSURE and AMDCAUSE=TOOLONG-5000
which seems OK as DIALSTATUS isn't reliable.
The same dialplan using a SIP trunk is working as expected.
So question is, why, when using analog line, I dont get the right behavior.
Thanks for any hint
Analog lines don't have a reliable way to know when the far end actually
answers. Polarity
reversals could signal when the far end actually answers, but it isn't
normally available or
standardized. Thus, the line is considered answered when dialing is
complete.
Richard
--
_
-- 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] Debugging stuck inbound call
1) put a maximum number of loops in your IVR to terminate calls which are dead or gone. 2) put maximum message length in voicemail.conf (ever tried to delete a 4 day long voicemail?) 3) Call sometimes get stuck. This is life. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Friday, March 28, 2014 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Debugging stuck inbound call Asterisk 11.1.0 running on Ubuntu 12.04 64 bit Dahdi Digium T1 card Occasionally, I will find an inbound call that just seems to be stuck, usually in our after-hours menu portion of the dial plan. This morning I had this one core show channels concise DAHDI/i1/5184097869-1baf2!MainMenu!s!20!Up!BackGround!custom/aa-night-hellocustom/hours_8:0-17:0_0:0-0:0_0:0-0:0custom/aa-night-instructions!5184097869!!!3!9393!(None)!sip1-1396004671.285644 which had been there for about 2.5 hours (time from core show channels verbose). The inbound channel here is to our toll free number on the T1. When we've researched these in the past, we've not found a correspondingly long call on the phone bill, leading me to wonder if the call is actually being disconnected, but Asterisk just doesn't find out. How can I go about debugging this? Are the dahdi commands that can show me the connection status from the hardware perspective? -- Mitch -- _ -- 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] Asterisk CLI Banner
asterisk-users-boun...@lists.digium.com wrote on 03/28/2014 10:51:13 AM: From: Haider Khalil haiderkha...@hotmail.com Thank you Thorsten Göllner. Matthew, What does violating license of Asterisk means ? Does it means I won't be able to use any commercial modules or asterisk commercially ? I thought it was open and anyone can change the code ? Haider I am neither a lawyer or a licensing expert, but the basics are that if you make such a change for your own internal use, you are probably fine. Example: You have 10 sites with Asterisk in them and at each site you have someone in your company who has to log into the CLI and do stuff. You change the header to pass them a message. This is probably (not going to guarantee this) going to be fine as it is not something you are releasing out into the wild nor are you selling it and making a profit from it. However, let's say you make a commercial project that uses Asterisk under the hood and you change the header to hide the fact that it uses Asterisk and that Digium has any ownership of the code. That would be not be okay in most, if not all cases. Basically, the code is open source, but it is still owned by Digium and they have specific rights that you have to be careful of in regards to licensing. If someone outside of your organization will ever be running the code you change, there are specific rules that have to be followed, including those that relate to releasing your changes to the code and to giving credit back to those who wrote the code your code is based on. Basically, Richard Kenner is spot on. If you are unclear, best to consult an attorney who specializes in this, especially if you are redistributing the altered code.-- _ -- 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] Asterisk CLI Banner
Richard Kenner wrote: Modifying a program you have legitimately acquired is Fair Dealing. The Law of the Land gives you the right to do that, even if the vendor restricts your exercise of that right in practice by withholding the Source Code. That is false. Modifying a program is creating a derivative work. As purchaser of a copyrighted item, you normally *do not* have that right. And this certainly may vary from jurisdiction to jurisdiction. For a (quite dated at this point) discussion of this issue from a US perspective, see http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157 The author is a recognized expert in software IP law. Of course, any good attorney will never commit to anything. They will never say it is alright to do X, unless X is do nothing Patent copyright attorneys seem especially non committal, at least in the US. probably because if any case ever goes to court, the decision and possible punishment is up to the whims of the judge and/or jury, and every law is up to interpretation, which can vary from moment to moment. Law is not physics! John Novack -- Dog is my Co-pilot -- _ -- 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] Asterisk CLI Banner
On Fri, Mar 28, 2014 at 9:10 AM, Thorsten Göllner t...@ovm-group.com wrote: Am 28.03.2014 10:32, schrieb Haider Khalil: Hello Experts, I want to know if there is any way to modify welcome banner on asterisk console when I connect using asterisk -r Hi, did you compile asterisk from source? Take a look at main/asterisk.c (line 174 in asterisk v 11.5.1). I think you have to change it there manually and recompile it. Please note that modifying the banner in main/asterisk.c may cause you to violate the licensing of Asterisk, specifically Section 1 of the GPL (if you distribute the modified source in any fashion) and/or Section 2c. Unless you really know what you're doing with regards to software licensing, I would highly suggest not modifying the welcome message. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk CLI Banner
What does violating license of Asterisk means? Does it means I won't be able to use any commercial modules or asterisk commercially? I thought it was open and anyone can change the code? Anyone *can* change the code. But it's licensed software, just like most other software. The difference is that the GPL gives you rights that you don't have for other non-open software. However, in both cases, you have to be sure that you don't violate the terms of the license. If you're unclear as to whether what you propose to do will violate the license, I'd suggest consulting an attorney: nobody on this list (or any other) should be providing you legal advice. -- _ -- 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] Debugging stuck inbound call
Asterisk 11.1.0 running on Ubuntu 12.04 64 bit Dahdi Digium T1 card Occasionally, I will find an inbound call that just seems to be stuck, usually in our after-hours menu portion of the dial plan. This morning I had this one core show channels concise DAHDI/i1/5184097869-1baf2!MainMenu!s!20!Up!BackGround!custom/aa-night-hellocustom/hours_8:0-17:0_0:0-0:0_0:0-0:0custom/aa-night-instructions!5184097869!!!3!9393!(None)!sip1-1396004671.285644 which had been there for about 2.5 hours (time from core show channels verbose). The inbound channel here is to our toll free number on the T1. When we've researched these in the past, we've not found a correspondingly long call on the phone bill, leading me to wonder if the call is actually being disconnected, but Asterisk just doesn't find out. How can I go about debugging this? Are the dahdi commands that can show me the connection status from the hardware perspective? -- Mitch -- _ -- 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] Asterisk CLI Banner
Of course, any good attorney will never commit to anything. They will never say it is alright to do X, unless X is do nothing No, but a good attorney can give guidance as to likely expectations. As you say, nobody can be sure of something even if it's previously been established law, but a good attorney can point out potential pitfalls on the one hand and identify, on the other, things that are much less likely to be an issue. It's not a guarantee, but you can often get a recommendation about whether or not it's a good idea (not necessarily alright) to do something. Attorneys often have to a take a stand on these matters. If a company needs to use software that performs a specific thing and, say, only three companies provide such, but under different licensing terms, it's the job of that company's legal department to say which, if any, they can be used. Doing nothing will have a cost and risk here too because this example is talking about something that the company needs done. -- _ -- 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] Asterisk CLI Banner
On Fri, 28 Mar 2014, Richard Kenner wrote: And this certainly may vary from jurisdiction to jurisdiction. For a (quite dated at this point) discussion of this issue from a US perspective, see http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157 The publication (43 pages) is dated 1988. The DMCA (1998) and subsequent legislation may have changed the landscape. My (ignorant) opinion -- just don't. Is it worth the effort to research? Is it worth paying a lawyer to research it and give an opinion that may be worth nothing until it is examined in court? If you want to display something custom, how about a 'wrapper' script that displays a file using 'curl' before handing off to Asterisk -- easier to implement, easier to maintain, no legal BS to consider. Or can you express your creativity by fiddling with ASTERISK_PROMPT? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Need some PHP/AMI guidance please
Hello all,
I've got some PHP code that opens an AMI socket and does a ConfBridgeList
for a specific bridge ().
This all works just fine but I need to filter the information displayed to
only CallerIDName so I can see a complete list of names of participants.
After days of googling and playing with it, I'm no closer than I was when I
started.
I'm not at all married to a table. A simple list of names is fine...
Any help is much appreciated!
Pertinent code:
?php
$ami = fsockopen(127.0.0.1, 5038, $errno, $errstr);
if (!$ami) {
echo ERROR: $errno - $errstrbr /\n;
} else {
fwrite($ami, Action: Login\r\nUsername: someuser\r\nSecret:
somesecret\r\nEvents: off\r\n\r\n);
fwrite($ami, Action: ConfbridgeList\r\nConference: \r\n\r\n);
sleep(1);
$record = fread($ami,1024);
$record = explode(\r\n, $record);
echo META HTTP-EQUIV=Refresh CONTENT=\20\;
echo table border=\1\ style='color: black;';
foreach($record as $value){
if(!strlen(stristr($value,'Asterisk'))0
!strlen(stristr($value,'Response'))0
!strlen(stristr($value,'Message'))0
!strlen(stristr($value,'Event'))0
strlen(strpos($value,' '))0)
php_table($value);;
}
echo /table;
fclose($ami);
}
function php_table($value){
$row1 = true;
$value = explode( , $value);
foreach($value as $field){
if($row1){
echo trtd.$field./td;
$row1 = false;
}
else{
echo td.$field./td/tr;
$row1 = true;
}
}
}
?
I think the explode is where I should be looking but I'm very new to PHP
Thank you!
Pat...
--
_
-- 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] DAHDI-Linux and DAHDI-Tools 2.9.1 Now Available
The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1 DAHDI-Tools-v2.9.1 dahdi-linux-complete-2.9.1+2.9.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release includes a firmware update for the wcaxx, wcte13xp, and wcte43x series of cards. This includes the A4B, A8B, TE133, TE131, WCTE235 and WCTE435. This firmware update resolves a rare case with certain chipsets that would cause transmitted audio to be muted for a short time on analog cards or spans to go down for a short time on digital cards. Shortlog of dahdi-linux changes since v2.9.0: Oron Peled (2): xpp: PRI stability fixes xpp: fix PANIC for old dahdi_registration Shaun Ruffell (9): firmware: Refactor by using build_tools/install_firmware. build_tools/install_firwmare: Try to extract the .bin file from .tar.gz wcxb: Print running version when recommending power cycle. wcxb: Reset TDM engine on IO errors. wcxb: Add diagnostic message if DMA retries are increasing when DEBUG is defined. wcte13xp: Update firmware for TE133/TE131 to 780019 wcaxx: Update firmware for A8B/A4B to 1d0019/b0019. wcte43x: Update firmware for TE435 / TE235 to e0019. wcxb: Disable presence detect reporting on upstream port during PCIe hard reset. Tzafrir Cohen (1): dahdi_get_auto_assigned_spans Shortlog of dahdi-tools changes since v2.9.0.1: Aslan Laoz (1): waitfor_xpds: handle the case of a failing AB Oron Peled (2): hotplug: call handle_device.d/ actions for remove registration-order: Added dahdi_auto_assign_compat Shaun Ruffell (2): hotplug: Check for auto_assign_spans only when ACTION is add. dahdi_cfg: error()-perror() when sem_open fails. Tzafrir Cohen (1): auto_assign_spans may be true even if not '1' The diffstat from the dahdi-linux v2.9.0 release: build_tools/install_firmware| 23 + drivers/dahdi/dahdi-base.c | 12 ++- drivers/dahdi/firmware/Makefile | 191 +--- drivers/dahdi/wcaxx-base.c | 4 +- drivers/dahdi/wcte13xp-base.c | 13 ++- drivers/dahdi/wcte43x-base.c| 2 +- drivers/dahdi/wcxb.c| 97 drivers/dahdi/wcxb.h| 9 ++ drivers/dahdi/xpp/card_pri.c| 14 ++- drivers/dahdi/xpp/xbus-core.c | 14 +-- drivers/dahdi/xpp/xbus-sysfs.c | 30 +-- drivers/dahdi/xpp/xpp_dahdi.c | 53 +++ drivers/dahdi/xpp/xpp_dahdi.h | 4 +- include/dahdi/kernel.h | 3 +- 14 files changed, 221 insertions(+), 248 deletions(-) The diffstat from the dahdi-tools v2.9.0.1 release: Makefile| 1 + dahdi.init | 8 +++- dahdi_cfg.c | 6 +++--- hotplug/dahdi_auto_assign_compat| 25 + hotplug/dahdi_handle_device | 24 +--- hotplug/dahdi_span_config | 25 ++--- hotplug/handle_device.d/10-span-types | 7 +++ hotplug/handle_device.d/20-span-assignments | 13 ++--- xpp/dahdi_registration | 2 +- xpp/waitfor_xpds| 4 10 files changed, 81 insertions(+), 34 deletions(-) For a full list of changes in these releases, please see the shortlog at: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.9.1 http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.9.1 Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira [1] https://issues.asterisk.org/jira/browse/DAHLIN [2] https://issues.asterisk.org/jira/browse/DAHTOOL Thank you for your continued support of Asterisk! -- _ -- 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] Best zwave controller for MH
I (canadian) store has a deal on for the vera lite controller: http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=8930107sku=VEP-STARTER1 but this looks different than the vera lite green white: http://www.amazon.com/Mi-Casa-Verde-VeraLite-Controller/dp/B007005364/ref=cm_cr_pr_product_top? and maybe there are more zwave controllers. Can someone (who is actually running Zwave+MH) comment on what the best model of controller to buy is? -- _ -- 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] Best zwave controller for MH
?oops...wrong list :) From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis mdup...@ocg.ca Sent: Friday, March 28, 2014 5:43 PM To: Asterisk Users List Subject: [asterisk-users] Best zwave controller for MH I (canadian) store has a deal on for the vera lite controller: http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=8930107sku=VEP-STARTER1 but this looks different than the vera lite green white: http://www.amazon.com/Mi-Casa-Verde-VeraLite-Controller/dp/B007005364/ref=cm_cr_pr_product_top? and maybe there are more zwave controllers. Can someone (who is actually running Zwave+MH) comment on what the best model of controller to buy is? -- _ -- 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] Security log format / content
Why does the failed authentication place the number dialed, instead of the username used, in the account field? Any way to distinguish a failed dial attempt from a failed register attempt using just the security log? (I couldn't see how looking at the log) From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Michael L. Young myo...@acsacc.com Sent: Thursday, March 27, 2014 2:42 PM To: Asterisk Users List Subject: Re: [asterisk-users] Security log format / content - Original Message - From: Michelle Dupuis mdup...@ocg.ca To: Asterisk Users List asterisk-users@lists.digium.com Sent: Thursday, March 27, 2014 12:55:21 AM Subject: [asterisk-users] Security log format / content I've noticed that the Asterisk (v11) security log captures attempts do dial without first authenticating, and places the number dialed into the accountid field. I'm trying to distinguish between failed attempts to register and attempts to dial without registering, but the security log treats them identically (using the accountid field for either the username or number dialed). I have noticed that the eventversion field is set to 2 for failed dial attempts, and 1 otherwise. Is this coincidence? Or can I rely on the eventversion=2 in the future to distinguish these two event types? (I've looked here: https://wiki.asterisk.org/wiki/display/AST/Security+Log+File+Format but it doesn't really help) The eventversion field is just a way to distinguish different versions of the same event. Between Asterisk 10 and 11, that particular event's logging output changed requiring a bump up in the version. It should not be used to distinguish different events. What do you mean by eventversion field is set to 2 for failed dial attempts, and 1 otherwise? What is the event? I have a feeling those are two different events. You are correct about the events looking identical whether it is a failed registration or a failed dial attempt. From the standpoint of Asterisk, an attempt was made to either register or place a call but the credentials failed. Therefore, an InvalidPassword event is logged. When an authorized device successfully places a call, you will only have a ChallengeSent entry in your log. If an attempt to place a call is made and it does not respond back with the right credentials to the challenge sent to Asterisk, then you will have a ChallengeSent entry with a subsequent InvalidPassword. You should be able to connect the two events based on the fields in those events. If a successful attempt to register is made, you will have a ChallengeSent with a subsequent SuccessfulAuth. If it is not successful, then you will have a ChallengeSent with a subsequent InvalidPassword. Again, there should be enough information present with the other fields to help connect the events together. The security events in Asterisk are designed to present the events. It does not determine anything else for you. You have to create a consumer of those events that can attempt to connect the dots for you. Hopefully we are providing enough information for the consumer to do whatever you would like the consumer to do with the information. I hope that helps. Michael -- _ -- 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] CLI command to see if SRTP is active?
Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. Thanks, Patrick -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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
