Re: [asterisk-users] Register Attacks End of ENUM ?
On Sun, Jul 25, 2010 at 3:11 AM, Norbert Zawodsky norb...@zawodsky.at wrote: Hello again! after it being relatively quiet her for the last weeks, my Astrerisk server was the target of 3 of that nasty REGISTER attacks during the last days. While I can see not much danger coming from these attacks (I use very long, complicated random generated passwords), they are still very annoying, because they always lead to my server crashing. (I think it's some out of memory condition because its a very tiny server. Slow CPU, not much memory...) Now, as a quick-fix I had the idea to use iptables' --scr-range rule to close the whole adress-range from 0.0.0.0 to 255.255.255.255 EXCEPT that small range of my VOIP provider. This should keep out all attacks. (At least, I think so). But I'm not a iptables-guru at all !! But the side-effect would be that ENUM wouldn't work any more. I still think that the best, clean solution would be, if some mechanism was built into asterisk (maybe sip.conf was the right place ???) where you could configure from which source (ip-range, ethernet-port or whatever...) asterisk will accept or ignore REGISTER requests. For example, in my small installation, valid REGISTERs can only originate from the internal LAN, never from the outside world. So I could restrict the range for valid REGISTERs to 192.168.1.0/24. AFAIK incoming calls would start the conversation with INVITE and those still may come from the outside (=any IP adress). Another thought makes me feel nervous: What if some sick brain gets the idea of sending INVITEs instead of those REGISTERs... Norbert If all you need is block the SIP traffic from external sources, you may do the following: # iptables -A INPUT -s 192.168.1.0/24 -p udp --dport 5060 -j ACCEPT # iptables -A INPUT -p udp --dport 5060 -j DROP # iptables-save /etc/iptables.up.rules and somewhere in init scripts (depending on your lsb release): # iptables-restore /etc/iptables.up.rules fail2ban is more suitable if you have external environment (plus it's more complicated than just these 2 rules). -- _ -- 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] Not urgent [was: urgent:how to transfer a call using asterisk FAGI]
Lately there have been a number of messages about urgent issues. This is intended as a reply to them all, not just to this specific one. On Tue, Jul 27, 2010 at 11:08:24AM +0530, Janu Mukherjee wrote: Hi, [Snip description of a problem] How can i achieve this???Please help me in this regard as this is very urgent. If it's so urgent, pay someone. This won't escalate the support issue to second-level support. In fact, it only serves to make the support personnel more annoyed of the support issue you have opened. As you may recall, this is not a payed support service. This is a community mailing list. We answer here mainly for fun and not for profit. This answer would not be complete without the obvious reference: http://catb.org/~esr/faqs/smart-questions.html#urgent That whole document is worth reading, BTW. -- 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 -- 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 Crashes - Segmentation Fault
Hi Dan, I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) --Manmohan Singh. On Tue, Jul 27, 2010 at 3:03 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan Singh Jandu wrote: OK, now i added the column members in the table booking manually. and disabled selinux to have this working. Now i am struggling with recording option in webmeetme. Not sure on how to enable it, though m checking the checkbox while creating the conference. But where does this save and how to retrieve it? The location of the recordings is set in lib/defines.php as RECORDING_PATH, which defaults to /var/lib/asterisk/sounds/conf-recordings/ You can listen to the recordings after the conferences scheduled stop time by looking at the Past conferences page and clicking on the speaker icon next to the conference number. A couple of items to note- 1. You may have to check the path to ensure it exists and that the asterisk process can write to it. 2. Your web service accounts needs read permissions for that path 3. The speaker icon only displays if a recording exists. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] 1 second Audio Lag
Hi All (reposting after 24 hours). I will do a test call from a soft phone to my mobile. I can speak into my headset and the audio is heard instantly. But if I speak into my mobile there is a 1-2 second delay in the Audio. I am using SIP. I am only finding it in the Zoiper Softphones that we are using. We are able to make a call without lag on the X-lite softphone no problem. Sadly the customer is Quite attached to the Zoiper. I have set QOS = CS5 for both SIP and RTP packets. Altering these settings has no effect to the lag issue. We have three 24 port Gigabit switches, with the top switch connecting in the Asterisk Box. Even the stations plugged into the TOP switch have this delay and to the same extent as the other switches. No routers on the loop I have tried switching the stations to IAX. No effect. I have tried using GSM instead of G711 (alaw). No effect. I have about 30 stations. No change under heavy or light load. I have done a Wireshark trace on the stations and no issues detected when I go analyse on the RTP packets. All sequencing is correct. Is Zoiper any good? Anyone else had these problems? -- _ -- 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] Register Attacks End of ENUM ?
Blocking SIP traffic is still going to break ENUM. The problem with your suggestion Norbert is that Asterisk still would have to process the requests at an application layer, providing no real advantage to users of boxes with no grunt. You could potentially write something to do inspection on the packets, there are a handful of L7 Linux switch projects around. Of course - still relatively resource intensive. Fail2Ban is probably the best solution. What someone needs to offer is an ENUM gateway service :-) Nick. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motiejus Jakštys Sent: Tuesday, 27 July 2010 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Register Attacks End of ENUM ? On Sun, Jul 25, 2010 at 3:11 AM, Norbert Zawodsky norb...@zawodsky.at wrote: Hello again! after it being relatively quiet her for the last weeks, my Astrerisk server was the target of 3 of that nasty REGISTER attacks during the last days. While I can see not much danger coming from these attacks (I use very long, complicated random generated passwords), they are still very annoying, because they always lead to my server crashing. (I think it's some out of memory condition because its a very tiny server. Slow CPU, not much memory...) Now, as a quick-fix I had the idea to use iptables' --scr-range rule to close the whole adress-range from 0.0.0.0 to 255.255.255.255 EXCEPT that small range of my VOIP provider. This should keep out all attacks. (At least, I think so). But I'm not a iptables-guru at all !! But the side-effect would be that ENUM wouldn't work any more. I still think that the best, clean solution would be, if some mechanism was built into asterisk (maybe sip.conf was the right place ???) where you could configure from which source (ip-range, ethernet-port or whatever...) asterisk will accept or ignore REGISTER requests. For example, in my small installation, valid REGISTERs can only originate from the internal LAN, never from the outside world. So I could restrict the range for valid REGISTERs to 192.168.1.0/24. AFAIK incoming calls would start the conversation with INVITE and those still may come from the outside (=any IP adress). Another thought makes me feel nervous: What if some sick brain gets the idea of sending INVITEs instead of those REGISTERs... Norbert If all you need is block the SIP traffic from external sources, you may do the following: # iptables -A INPUT -s 192.168.1.0/24 -p udp --dport 5060 -j ACCEPT # iptables -A INPUT -p udp --dport 5060 -j DROP # iptables-save /etc/iptables.up.rules and somewhere in init scripts (depending on your lsb release): # iptables-restore /etc/iptables.up.rules fail2ban is more suitable if you have external environment (plus it's more complicated than just these 2 rules). -- _ -- 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] Meetme Question
Hi Danny, Thanks a lot for the reply , I tried the dial plan you have provided , when pressing '0' it gets connected with the extension defined in [meetme-oper] context , but after disconnecting the operator call user exits from the meetme room , can we avoid that ? I am trying to achieve the following callflow, 1.User calls meetme bridge number 2.User enters the bridge with PIN 3.During the session if he needs any assistance he presses '0' and talks with the operator 4.After talking with the operator user gets back to the conference again Thanks in advance. Regards Shiju V.Joseph From: Danny Nicholas da...@debsinc.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: 07/21/2010 06:36 PM Subject: Re: [asterisk-users] Meetme Question Sent by: asterisk-users-boun...@lists.digium.com Hi , I am trying to add an operator assistance feature to meetme , when the user dials '0' ,support / help desk personnel should be added to the live conference for live support / troubleshooting. How can i do this ? Can I edit the meetme * menu and add a new menu item ' Press '0' for support' .I think I will have to edit the meetme.c source to do this , hard way :( or is it possible to write an AGI script which detects when a user dials '0' and calls the helpdesk number (preconfigured number) or generally is it possible to collect the DTMF response from a user during a meetme conf call and trigger some action / script , I searched a lot in forums / mailing list , most of the threads are pretty old and confusing. Any help / hints will be greatly appreciated. Thanks Shiju V.Joseph Just add ?X? to the meetme string and define 0 action; something like this Exten = 1234,1,Goto(meetme-oper|s|1) [meetme-oper] Exten = s,1,meetme(1234,X) Exten = s,n,hangup Exten = 0,1,dial(SIP/100,30,m) When you dial 1234, you are put into conference 1234 If you press 0 while in the conference, you are transferred to extension 100. _ -- 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 -- The information contained in this communication is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. It may contain confidential or legally privileged information. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking any action in reliance on the contents of this information is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by responding to this email and then delete it from your system. Ernst Young is neither liable for the proper and complete transmission of the information contained in this communication nor for any delay in its receipt. -- _ -- 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] 1 second Audio Lag
Do you see the issue when calling between two softphones? Do you see the issue if you call from your mobile into an echo test? Setting TOS flags on packets will make no difference unless the gear in between is configured to treat them differently. Not that I envision this is the issue at all. Nick. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of colin mcdermott Sent: Tuesday, 27 July 2010 5:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1 second Audio Lag Hi All (reposting after 24 hours). I will do a test call from a soft phone to my mobile. I can speak into my headset and the audio is heard instantly. But if I speak into my mobile there is a 1-2 second delay in the Audio. I am using SIP. I am only finding it in the Zoiper Softphones that we are using. We are able to make a call without lag on the X-lite softphone no problem. Sadly the customer is Quite attached to the Zoiper. I have set QOS = CS5 for both SIP and RTP packets. Altering these settings has no effect to the lag issue. We have three 24 port Gigabit switches, with the top switch connecting in the Asterisk Box. Even the stations plugged into the TOP switch have this delay and to the same extent as the other switches. No routers on the loop I have tried switching the stations to IAX. No effect. I have tried using GSM instead of G711 (alaw). No effect. I have about 30 stations. No change under heavy or light load. I have done a Wireshark trace on the stations and no issues detected when I go analyse on the RTP packets. All sequencing is correct. Is Zoiper any good? Anyone else had these problems? -- _ -- 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] Configuring X-lite for a remote user
Hi. Don't know if you have static public IP, but guess not, so you will have to configure one dynamic dns service. There's some services available like www.no-ip.com or www.dyndns.com. I know that no-ip.com got a linux client, that you can install on your Asterisk server. Then you have to forward SIP and RTP ports to your Asterisk machine, in your router and configure the remote x-lite to connect to your dns name. With this you friend will always connect to the same dns name, that will be point to your dynamic ISP address, updated by the no-ip, or whatever, client. Regards! On 27/07/10 02:14, Adolphe Cher-aime wrote: To have your asterisk box reachable from internet you must configure static nat on your router to get sip traffic to the public Ip redirected to your internal ip. Make sure that sip and rtp traffic are not bloked by firewall. And configure xlite to connect to your public ip address. Adolphe Cher-aime From my Iphone On Jul 26, 2010, at 7:48 PM, ayodele abejide ayodeleabej...@hotmail.com mailto:ayodeleabej...@hotmail.com wrote: I have asterisk running at home, a friend would be traveling out of the country and I want him to be able to put a call through from his remote location, I am wondering how I would configure the X-lite client on his pc so he would be able to call through assuming my public address is A.B.C.D and the static address the asterisk machine is on is 192.168.0.3. Thanks in anticipation Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 -- _ -- 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 -- Hugo Serrano hugo.serr...@javali.pt Javali Administração e Desenvolvimento de Sistemas Informáticos, Lda. Madan ParqueEdifício VI Campus da FCT/UNL Quinta da Torre 2829-516 Caparica Portugal Phone: +351 212949666 Fax: +351 212948313 www.javali.pt i...@javali.pt -- _ -- 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] URgent - capturing 'answered'
Thats great,
However I need to find a solution to this very problem, not able to code
something from scratch.
Even this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
'$agi.get_variable(EXTEN)', '$variable1', '$variable2'
end
$my.query(UPDATE call_log SET endtime = NOW() WHERE
id = #{call_log_id})
- query gets executed, but log.txt wasnt created.
Not to mention that I still didnt manage to catch who answered the call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, July 26, 2010 8:10 PM
To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On Mon, 26 Jul 2010, Andres wrote:
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like: $message=/bin/echo my variables are '$loc',
'$variable1', '$variable2', etc /tmp/variables.txt;
system($message);
I prefer syslog().
) You don't litter your system with little files.
) You get nicely timestamped messages you can centralize across servers.
) You can control how much verbosity you want by setting the logging
priority.
) You can vary the logging priority at run time.
) You can leave the logging code in place in production.
I code all of my AGIs to recognize (via getopt_long()) --debug and
--verbose command line options. When something weird starts to happen, I
can enable debugging in the dialplan and debug the code that is running in
production.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] URgent - capturing 'answered'
No, neither that didnt work.
Even this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
'$agi.get_variable(EXTEN)', '$variable1', '$variable2'
end
$my.query(UPDATE call_log SET endtime = NOW() WHERE
id = #{call_log_id})
- query gets executed, but log.txt wasnt created.
Not to mention that I still didnt manage to catch who answered the call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, July 26, 2010 8:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On Mon, 26 Jul 2010, Zarko Zivanovic wrote:
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
I'm just a c weenie, but that syntax would execute a command named
$message, not the value of the variable $message.
Would
system($message);
do what you want?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] URgent - capturing 'answered'
I tried this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
'$agi.get_variable(EXTEN)', '$variable1', '$variable2'
end
$my.query(UPDATE call_log SET endtime = NOW() WHERE
id = #{call_log_id})
- query gets executed, but log.txt wasnt created.
Not to mention that I still didnt manage to catch who answered the call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 8:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 1:40 PM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
This is what I do with Perl AGI scripts and it works fine. You need to
figure out how to output to a text file with Ruby. I don't think the
'system' command would work with Ruby. Start with a basic AGI script
and test wether you can write to a file or not. That is the best way to
troubleshoot.
Andres
http://www.neuroredes.com
permissions seem to be fine, echo is in place.
I posted the whole script that i am using in the main thread - if you can
please loook at it.
Zarko.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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Re: [asterisk-users] URgent - capturing 'answered'
Great, but how exactly do i find that channel - that is my question - which
command.
I am using ruby instead of agi - and i am looking for a command to capture
it in ruby.
I tried this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
'$agi.get_variable(EXTEN)', '$variable1', '$variable2'
end
$my.query(UPDATE call_log SET endtime = NOW() WHERE
id = #{call_log_id})
- query gets executed, but log.txt wasnt created.
Not to mention that I still didnt manage to catch who answered the call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Monday, July 26, 2010 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
If all you need to do is the the channel name of the channel that answered
the phone why are you doing so much work? Version 1.4 allows for an agi to
be called when the dial command is answered. Version 1.6+ allows an agi as
well as a macro to be called. You can find the channel that answered a multi
channel dial command. Is this not what you wanted to know?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 10:40 AM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
permissions seem to be fine, echo is in place.
I posted the whole script that i am using in the main thread - if you can
please loook at it.
Zarko.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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New to Asterisk?
Re: [asterisk-users] URgent - capturing 'answered'
Hi! Great, but how exactly do i find that channel - that is my question - which command. For the third time: Use the M option to Dial() and create a Macro. In that macro use the SIPCHANINFO() or CHANNEL() function to get what you want to get. No AGI (and AGI is a protocol while Ruby is a language). If all you need to do is the the channel name of the channel that answered the phone why are you doing so much work? Version 1.4 allows for an agi to be called when the dial command is answered. Version 1.6+ allows an agi as well as a macro to be called. You can find the channel that answered a multi channel dial command. Is this not what you wanted to know? Philipp -- _ -- 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] Register Attacks End of ENUM ?
Am 27.07.2010 08:42, schrieb Motiejus Jakštys: If all you need is block the SIP traffic from external sources, you may do the following: # iptables -A INPUT -s 192.168.1.0/24 -p udp --dport 5060 -j ACCEPT # iptables -A INPUT -p udp --dport 5060 -j DROP # iptables-save /etc/iptables.up.rules and somewhere in init scripts (depending on your lsb release): # iptables-restore /etc/iptables.up.rules fail2ban is more suitable if you have external environment (plus it's more complicated than just these 2 rules). Hello Motiejus, Hello Nick! thanks for your answers. My OP was definitely not meant as a request for help. I just wanted to start some small discussion. The point is that a) I don't know fail2ban, and b) I think that small box which runs my asterisk wouldn't take another additional application (like fail2ban) @Motiejus: Thanks for your rules! Since it seems that you are an iptables expert, may I ask you: I want to restrict SIP traffic to my internal network AND to a special adress-range (adresses of my voip provider) from external network. iptables -A INPUT -s 192.168.1.0/24 -p udp --dport 5060 -j ACCEPT iptables -A INPUT -m iprange --src-range [FROM_IP]-[TO_IP] -j ACCEPT iptables -A INPUT -p udp --dport 5060 -j DROP Would that do the trick ? But that would keep out any calls via ENUM mechanism too. Am I right? Norbert -- _ -- 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] URgent - capturing 'answered'
Which version of Asterisk are you running?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 2:19 AM, Zarko Zivanovic wrote:
Great, but how exactly do i find that channel - that is my question - which
command.
I am using ruby instead of agi - and i am looking for a command to capture
it in ruby.
I tried this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
'$agi.get_variable(EXTEN)', '$variable1', '$variable2'
end
$my.query(UPDATE call_log SET endtime = NOW() WHERE
id = #{call_log_id})
- query gets executed, but log.txt wasnt created.
Not to mention that I still didnt manage to catch who answered the call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Monday, July 26, 2010 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
If all you need to do is the the channel name of the channel that answered
the phone why are you doing so much work? Version 1.4 allows for an agi to
be called when the dial command is answered. Version 1.6+ allows an agi as
well as a macro to be called. You can find the channel that answered a multi
channel dial command. Is this not what you wanted to know?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 10:40 AM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
permissions seem to be fine, echo is in place.
I posted the whole script that i am using in the main thread - if you can
please loook at it.
Zarko.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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Re: [asterisk-users] Register Attacks End of ENUM ?
Hello Motiejus, Hello Nick! thanks for your answers. My OP was definitely not meant as a request for help. I just wanted to start some small discussion. The point is that a) I don't know fail2ban, and It's really easy. I just installed it on my company asterisk box - it took ~5 minutes to install and configure. Thanks all. Moreover, it's scanning for sshd brute-force attacks out of the box. b) I think that small box which runs my asterisk wouldn't take another additional application (like fail2ban) It has a _very_ small footprint :-) I observe 0% cpu (in top) and 2MB system ram usage. @Motiejus: Thanks for your rules! Since it seems that you are an iptables expert, :-) may I ask you: I want to restrict SIP traffic to my internal network AND to a special adress-range (adresses of my voip provider) from external network. iptables -A INPUT -s 192.168.1.0/24 -p udp --dport 5060 -j ACCEPT iptables -A INPUT -m iprange --src-range [FROM_IP]-[TO_IP] -j ACCEPT iptables -A INPUT -p udp --dport 5060 -j DROP Would that do the trick ? Yes, syntax looks correct, it should. Try :-) But that would keep out any calls via ENUM mechanism too. Am I right? The above rule will block all UDP port 5060 (SIP) traffic from external ips to your asterisk machine. I do not know how ENUM works, so cannot answer, but probably Nick is right. If your asterisk is ENUM server listening on UDP 5060 and remote hosts query your machine with ENUM - then yes, it will not work. Any other configuration - it will. Regards Motiejus Jakštys -- _ -- 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] 1 second Audio Lag
Colin, I'm working for Zoiper, you can contact us directly on supp...@zoiper.com Zoa Nick Brown wrote: Do you see the issue when calling between two softphones? Do you see the issue if you call from your mobile into an echo test? Setting TOS flags on packets will make no difference unless the gear in between is configured to treat them differently. Not that I envision this is the issue at all. Nick. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of colin mcdermott Sent: Tuesday, 27 July 2010 5:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1 second Audio Lag Hi All (reposting after 24 hours). I will do a test call from a soft phone to my mobile. I can speak into my headset and the audio is heard instantly. But if I speak into my mobile there is a 1-2 second delay in the Audio. I am using SIP. I am only finding it in the Zoiper Softphones that we are using. We are able to make a call without lag on the X-lite softphone no problem. Sadly the customer is Quite attached to the Zoiper. I have set QOS = CS5 for both SIP and RTP packets. Altering these settings has no effect to the lag issue. We have three 24 port Gigabit switches, with the top switch connecting in the Asterisk Box. Even the stations plugged into the TOP switch have this delay and to the same extent as the other switches. No routers on the loop I have tried switching the stations to IAX. No effect. I have tried using GSM instead of G711 (alaw). No effect. I have about 30 stations. No change under heavy or light load. I have done a Wireshark trace on the stations and no issues detected when I go analyse on the RTP packets. All sequencing is correct. Is Zoiper any good? Anyone else had these problems? -- _ -- 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] CallerID disappear from CDR on transfer
Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/ ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound trunk) 7) if john doe want to speak with caller assistant bridge the two lines using the transfer function of GXP2000 phone (REFER). After the transfer in the CDR i can't see the callerid of the caller, only data of the bridged call is reported. Any idea on what i can do to keep it ? thanks lechuck -- _ -- 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] IAX bandwidth optimisation
Hello, I want to reduce the bandwidth taken by an IAX trunk when used with a small number of voice channels. When only one call is passed through an IAX trunk, the IP overhead is indecent. I would like to increase the IAX voice packet emission interval (20ms - i'm using speex) to something much larger, in order for the packets to transport as much data as possible. I've been trying to fuddle with the trunkfreq parameter and push it to 80ms, but that did not change the voice packet rate at all. How can I change that voice packet rate ? Thanks ! -- _ -- 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 crashes to start if compiled with pbx_lua on latest updated CentOS
On Tue, Jul 27, 2010 at 12:45 AM, Faisal Hanif fai...@vopium.com wrote: Did any one got it solved? If yes how? Yes, read doc/backtrace.txt. It will explain how to generate an unoptimized backtrace, then uploaded it to the mailing list. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] MeetMe
1) Ok, I'm using now self/peer on the feature map
2) there's no space in features.conf.
toca_macaco = 1,self/caller,Playback,tt-monkeys
But it's not working yet.
On Mon, Jul 26, 2010 at 11:33 PM, Tilghman Lesher tles...@digium.comwrote:
On Monday 26 July 2010 15:20:26 Felipe Figueiredo wrote:
Hi guys,
i'm trying to use the featuremap of features.conf inside the app
meetme,
but it's no working.
like:
_5XXX = {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hangup();
};
in features.conf:
toca_macaco = 123, peer, Playback,tt-monkeys
1) There is no peer when you invoke MeetMe. There is only a single call
leg.
You therefore want self or caller.
2) Kill the spaces on this line. All of them. Note that self,
caller,
or peer do not match anything and will thus signal Invalid 'ActivateOn'
specification for feature... at boot or reload. Similarly, there is a
dialplan application named Playback, but there is no dialplan application
named Playback.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] URgent - capturing 'answered'
://www.eset.com -- _ -- 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 __ Information from ESET NOD32 Antivirus, version of virus signature database 5315 (20100726) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __ Information from ESET NOD32 Antivirus, version of virus signature database 5315 (20100726) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -- _ -- 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 __ Information from ESET NOD32 Antivirus, version of virus signature database 5316 (20100727) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __ Information from ESET NOD32 Antivirus, version of virus signature database 5316 (20100727) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -- _ -- 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 transfer a call to operator using FAGI asterisk
Hi, I have xlite client registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==1500,1,AGI(localhost//hello.agi) So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard Thanks Regards, Jahnavi. -- _ -- 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] 'dirty' upgrade of 1.4
Thanks to everyone who replied. This is great news ;). I'll get the thing upgraded tonight (when it's quiet). Thanks again. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: 26 July 2010 16:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 When you run make, it compiles the binaries in the src directory. Once it is done compiling stop asterisk. Running make install will copy the compiled binaries into their respective folders on your system. Then just start asterisk. If you need to revert, stop asterisk, run make install in the old src directory, then start asterisk. Ryan On Mon, Jul 26, 2010 at 9:45 AM, Andrew Thomas a...@datavox.co.uk wrote: Hi Danny, I understand (and welcome) the separate src directories. This would allow me to 'revert' should I feel the need (assuming I can just re-compile over each one). I just need to know if I can re-compile over the existing first. Thanks for your reply. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 26 July 2010 14:15 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- Subject: [asterisk-users] 'dirty' upgrade of 1.4 Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Question 1 - unless you are un-tarring to a specific directory, you would have /usr/local/src/asterisk-1.4.24.1 and /usr/local/src/asterisk-1.4.34 segregated source trees. Question 2 - you don't have to stop asterisk, but you should (best practice?) since installing a new release usually involves removing/replacing the .so files in /usr/lib/asterisk/modules. -- _ -- 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 -- _ -- 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] Fail2ban - SuSEfirewall
The problem sounds like fail2ban is failing to write the new rules to a permanent file, which would otherwise allow the rules to persist after a reboot. Tilghman, That is exactly right. I'm thinking I need to revise the SuSEfirewall init scripts to follow up with restarting fail2ban, but then I think fail2ban will need to have a persistent jail after restarting, which I did find online. I am a big fan of centralized management, so I prefer to do that rather than have static IP addresses on the network (except of course where absolutely essential). For the OP: maybe a workaround is to assign a fixed IP address from your DHCP server and use a very long lease time? John, Agreed re management. The lease would have to be real long, like a year or so. That would do the trick. -- _ -- 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 transfer a call to operator using FAGI asterisk
On Tue, Jul 27, 2010 at 05:42:01PM +0530, Janu Mukherjee wrote: Hi, I have xlite client registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==1500,1,AGI(localhost//hello.agi) Obviously this is not the dialplan you have, as this would fail to work. So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. The operator is? Context? Extension? -- 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 -- 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] urgent:how to transfer a call using asterisk FAGI
Use dial application along with (agi command) exec for more see http://www.voip-info.org/wiki/view/exec On Tue, Jul 27, 2010 at 10:38 AM, Janu Mukherjee janu.mu...@gmail.comwrote: Hi, I have xlite registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==1500,1,AGI(localhost//hello.agi So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard as this is very urgent. Thanks Regards, Jahnavi. -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- 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 Zap-Sip calls.
We had attempted adding the 'r' to the dial command parameter and that didn't seem to have an effect. We played around with the progressinband a bit and tried to see if we could find a solution and only ended up with same results no matter if it were set to yes, no, or never. We set everything back to where it was in the beginning and it seems to be working now somehow. It has been running just fine and ringing since midday yesterday. Thanks for the help Philipp and Faisal! On 7/26/2010 11:37 PM, Faisal Hanif wrote: You may need to add r as option perameter to dial command. Regards, Faisal Hanif On 7/26/2010 9:39 PM, Chris Ramirez wrote: The problem we are having with Asterisk is when we initiate a call via a Zap line and it goes out on a Sip line. When it goes out via Sip we hear no sound until the party we are calling answers the line. If the call were to go out Sip-Sip or Zap-Zap it works perfectly fine. It is only with the Zap-Sip calls. If anyone knows anything that could possibly help it would be greatly appreciated. I have checked many different things already and tried comparing Zap-Zap and Zap-Sip call logs. Thanks! -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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 transfer a call to operator using FAGI asterisk
El 27/07/10 07:12, Janu Mukherjee escribió: Hi, I have xlite client registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==1500,1,AGI(localhost//hello.agi) So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard Thanks Regards, Jahnavi. Ok so you read Tzafrir's link hehe... not so urgent huh? This is easy, if you want to do this inside the AGI, use the AGI commands set context, set extension and set priority to set where you want the call to continue when the AGI finishes. If you use a library to handle the AGI communication like Asterisk-Java or PHPAGI, there are one line commands to achieve this. Finally, if you don't have control of your AGI and you need to make the transfer outside the AGI, simply do a Goto after the AGI to transfer the call where you need. Even asterisk itself gives you help: *CLI agi show set context *CLI agi show set extension *CLI agi show set priority Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] Urgent help = RUBY AGI
Here's something that should be easy for RUBY pro's.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
# $agi.set_variable(' WHOANSWERED ',...)
retry if r.message.include?('BUSY')
end
when it's executed it shows this in the console:
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx EXEC DIAL SIP/voipuserZap/32Zap/33Zap/34Zap/35
-- AGI Script Executing Application: (DIAL) Options:
(SIP/voipuserZap/32Zap/33Zap/34Zap/35)
-- Called voipuser
-- Called 32
-- Called 33
-- Called 34
-- Called 35
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/voipuser-e989 is ringing
-- SIP/ voipuser-e989 answered Zap/1-1
What we need is to be able to populate the variable WHOANSWERED with info
SIP/ voipuser
In this case, or whoever answers next time.
Thanks in advance!
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Re: [asterisk-users] Urgent help = RUBY AGI
- Zarko Zivanovic outlaw...@gmail.com wrote: Here’s something that should be easy for RUBY pro’s. Here's something that would be infinitely easier: You could understand that this list isn't your personal technical support resource where you can delegate how urgent or not your issue is. If you're really having 'urgent' issues, find someone who knows what they're doing and pay them. Don't spam the list with 'Urgent' on every new post you make. Frankly, I'm surprised you've received such a response already. I had some thoughts and ideas on your previous issues but simply chose to ignore you since I found your 'Urgency' distasteful. Others apparently have not... Tzafrir has already mentioned this to you. Maybe you missed that message? Here it is so you may review it: http://lists.digium.com/pipermail/asterisk-users/2010-July/251677.html --Tim -- _ -- 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] Urgent help = RUBY AGI
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Subject: Re: [asterisk-users] Urgent help = RUBY AGI - Zarko Zivanovic outlaw...@gmail.com wrote: Here's something that should be easy for RUBY pro's. My .02 (perhaps irrelevant); Out of the thousands/millions of Asterisk users (and the hundreds/thousands that read and reply to this list), I would wager that a very small percentage of us (royal we) are Ruby pro's. From what I read, the language proficiencies of most users are either PHP, Perl or C. Once you stop making things urgent and become as Asterisk Pro, perhaps you will also be this Ruby Pro that you speak of :-) -- _ -- 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] Urgent help = RUBY AGI
On 07/27/2010 09:38 AM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tim Nelson *Subject:* Re: [asterisk-users] Urgent help = RUBY AGI - Zarko Zivanovic outlaw...@gmail.com wrote: Here’s something that should be easy for RUBY pro’s. My .02 (perhaps irrelevant); Out of the thousands/millions of Asterisk users (and the hundreds/thousands that read and reply to this list), I would wager that a very small percentage of us (royal we) are “Ruby pro’s”. From what I read, the language proficiencies of most users are either PHP, Perl or C. Once you stop making things urgent and become as “Asterisk Pro”, perhaps you will also be this “Ruby Pro” that you speak of J ... and since you are using Ruby already, you could switch to using the Adhearsion framework, which makes interaction with Asterisk trivially easy, and handles all the AGI/AMI stuff 'under the covers' for you. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] Urgent help = RUBY AGI
I am sorry, I wasn’t aware that there is such a problem with urgency, as well as I never found it when it was directed to me on the other boards and lists, but that is the whole other issue. I always try to help others with what I know (In this area I must say that is not much), and I admit that once a while when I bump into an area that I do not know much of, as this one, I get frustrated with being unable to do some things that look pretty basic. And yes, so far I got much more replies where people were directing me to other solutions not related to issue I have and that is why i decided to start this thread and mention that I look to a solution to this very specific problem and I am not able to install other packages, use other languages etc. It should be that simple. Again I am apologizing for urgency, as much as I didnt mean to cause stress to any of users, neither to cause any harm. But for my 2 cents, let me say that it really surprised me that so far I didnt get a single precise answer to the question even though I posted almost a full script that we are using atm. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Tuesday, July 27, 2010 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Urgent help = RUBY AGI - Zarko Zivanovic outlaw...@gmail.com wrote: Here’s something that should be easy for RUBY pro’s. Here's something that would be infinitely easier: You could understand that this list isn't your personal technical support resource where you can delegate how urgent or not your issue is. If you're really having 'urgent' issues, find someone who knows what they're doing and pay them. Don't spam the list with 'Urgent' on every new post you make. Frankly, I'm surprised you've received such a response already. I had some thoughts and ideas on your previous issues but simply chose to ignore you since I found your 'Urgency' distasteful. Others apparently have not... Tzafrir has already mentioned this to you. Maybe you missed that message? Here it is so you may review it: http://lists.digium.com/pipermail/asterisk-users/2010-July/251677.html --Tim __ Information from ESET NOD32 Antivirus, version of virus signature database 5317 (20100727) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -- _ -- 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] Urgent help = RUBY AGI
Thanks Danny, I have no intentions to become either Ruby pro or Asterisk pro, and I believe that there are many people here who understand asterisk much better than I will ever do. That is why I am here and looking for this specific fix. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, July 27, 2010 4:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Urgent help = RUBY AGI From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Subject: Re: [asterisk-users] Urgent help = RUBY AGI - Zarko Zivanovic outlaw...@gmail.com wrote: Here's something that should be easy for RUBY pro's. My .02 (perhaps irrelevant); Out of the thousands/millions of Asterisk users (and the hundreds/thousands that read and reply to this list), I would wager that a very small percentage of us (royal we) are Ruby pro's. From what I read, the language proficiencies of most users are either PHP, Perl or C. Once you stop making things urgent and become as Asterisk Pro, perhaps you will also be this Ruby Pro that you speak of J __ Information from ESET NOD32 Antivirus, version of virus signature database 5317 (20100727) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -- _ -- 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] Urgent help = RUBY AGI
Hello Kevin. Thanks for your suggestion, these ruby scripts are something that we currently cant change due to many reasons. We are currently only looking for that specific fix that no one seemed to be able to sort out so far. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, July 27, 2010 4:42 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Urgent help = RUBY AGI On 07/27/2010 09:38 AM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tim Nelson *Subject:* Re: [asterisk-users] Urgent help = RUBY AGI - Zarko Zivanovic outlaw...@gmail.com wrote: Here's something that should be easy for RUBY pro's. My .02 (perhaps irrelevant); Out of the thousands/millions of Asterisk users (and the hundreds/thousands that read and reply to this list), I would wager that a very small percentage of us (royal we) are Ruby pro's. From what I read, the language proficiencies of most users are either PHP, Perl or C. Once you stop making things urgent and become as Asterisk Pro, perhaps you will also be this Ruby Pro that you speak of J ... and since you are using Ruby already, you could switch to using the Adhearsion framework, which makes interaction with Asterisk trivially easy, and handles all the AGI/AMI stuff 'under the covers' for you. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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 __ Information from ESET NOD32 Antivirus, version of virus signature database 5317 (20100727) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __ Information from ESET NOD32 Antivirus, version of virus signature database 5317 (20100727) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -- _ -- 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] Urgent help = RUBY AGI
Hi, I missed the beginning of this thread but you or anyone else looking for help with Ruby + Asterisk should contact Jason Goecke (@jsgoecke on Twitter or if you don't do Twitter you can look for contact info there http://twitter.com/gsgoecke). Jason probably knows as much about that world as anyone and he con help you find someone. Kevin's advice is good (as usual): check out Adhearsion. Jason is a part of that, too. /r -- _ -- 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] Urgent help = RUBY AGI
I have never used 1.2.9.1 or anything in the 1.2.x range so I can not give you
an exact solution but I can tell you that the script that you are using will
not work. In the dial command you need to add the M option which will call a
macro when the call is connected. In that macro you can then find the channel
that answered the call and do what you want from there. You can call another
AGI or set variables or whatever. If agi.exec works like a dialplan step then
the dial step will hang if the call is answered and the agi.get_variable
statement will not execute unless the call was not answered.
Try
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,M(testing))
And then have something like this in extensions.conf
[macro-testing]
exten = s,1,DumpChan()
You will see that this macro runs when the call is answered and you will see on
the CLI all the variables that are available to you. ${CHANNEL} will have SIP/
voipuser-e989 in your example below.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 7:21 AM, Zarko Zivanovic wrote:
Here’s something that should be easy for RUBY pro’s.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
# $agi.set_variable(' WHOANSWERED ',...)
retry if r.message.include?('BUSY')
end
when it’s executed it shows this in the console:
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx EXEC DIAL SIP/voipuserZap/32Zap/33Zap/34Zap/35
-- AGI Script Executing Application: (DIAL) Options:
(SIP/voipuserZap/32Zap/33Zap/34Zap/35)
-- Called voipuser
-- Called 32
-- Called 33
-- Called 34
-- Called 35
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/voipuser-e989 is ringing
-- SIP/ voipuser-e989 answered Zap/1-1
What we need is to be able to populate the variable WHOANSWERED with info
SIP/ voipuser
In this case, or whoever answers next time.
Thanks in advance!
__ Information from ESET NOD32 Antivirus, version of virus signature
database 5317 (20100727) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
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Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce bruceb...@gmail.com wrote: I seem to not be able to find any good open source Asterisk Queue Analyzer and Asterisk Log Analyzer on the web. google 'freepbx' It does some of what you want. For the rest of what you want, strongly consider paying a professional consultant. -- _ -- 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 1.8.0-beta2 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list. Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page. http://www.asterisk.org/asterisk-versions This release contains fixes since the last beta release as reported by the community. Some of the changes include: * Remove duplicate -c flag when using $(INSTALL) (Closes issue #17695. Reported, patched by pabelanger) * Don't re-register CDR module on reload. (Closes issue #17304. Reported, tested by jnemeth. Patched by tilghman) * Don't assume qlog is open. (Closes issue #17704. Reported, tested by vrban. Patched by pabelanger) * Expand the correct value within AST_OPTION_ONLY. (Closes issue #17703. Reported by stuarth. Patched by seanbright) * Allow for systems without locale support to be usable. (Closes issue #17697. Reported, patched by pprindeville. Tested by mmichelson) * Fixes for sounds/Makefile to install on systems using older GNU make. (Closes issue #17716. Reported by farisraouf. Patched by tilghman, qwell, seanbright) * Update logger.conf.sample to include documentation about new 'fax' logger level. (Closes issue #17715. Reported, tested by vrban. Patched by pabelanger) Asterisk 1.8 contains many new features over previous releases of Asterisk. A short list of included features includes: * Secure RTP * IPv6 Support * Connected Party Identification Support * Calendaring Integration * A new call logging system, Channel Event Logging (CEL) * Distributed Device State using Jabber/XMPP PubSub * Call Completion Supplementary Services support * Advice of Charge support * Much, much more! A full list of new features can be found in the CHANGES file. http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta2 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
Re: [asterisk-users] Urgent help = RUBY AGI
Jim thanks.
I will test this first thing in the morning as I am out of the office now.
As a matter of fact I cant wait to test this, as it has been the first
reasonable thing that looks like it could work.
In the meantime , do you happen to know if there is a way to call both macro
(M) and music on hold (m) in that $agi.exec line?
or is the right thing to do to place moh command in macro?
As I said, I cant wait to try it first thing in the morning and tell you
(and others) how it went. I am sure this will be the good reference to other
people looking for the same thing online as I have found quite a bunch of
similar open threads.
Zarko
On Tue, Jul 27, 2010 at 5:31 PM, Jim Dickenson dicken...@cfmc.com wrote:
I have never used 1.2.9.1 or anything in the 1.2.x range so I can not give
you an exact solution but I can tell you that the script that you are using
will not work. In the dial command you need to add the M option which will
call a macro when the call is connected. In that macro you can then find the
channel that answered the call and do what you want from there. You can call
another AGI or set variables or whatever. If agi.exec works like a dialplan
step then the dial step will hang if the call is answered and the
agi.get_variable statement will not execute unless the call was not
answered.
Try
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,M(testing))
And then have something like this in extensions.conf
[macro-testing]
exten = s,1,DumpChan()
You will see that this macro runs when the call is answered and you will
see on the CLI all the variables that are available to you. ${CHANNEL} will
have SIP/ voipuser-e989 in your example below.
--
Jim Dickenson
mailto:dicken...@cfmc.com dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 7:21 AM, Zarko Zivanovic wrote:
Here’s something that should be easy for RUBY pro’s.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
# $agi.set_variable(' WHOANSWERED ',...)
retry if r.message.include?('BUSY')
end
when it’s executed it shows this in the console:
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx EXEC DIAL SIP/voipuserZap/32Zap/33Zap/34Zap/35
-- AGI Script Executing Application: (DIAL) Options:
(SIP/voipuserZap/32Zap/33Zap/34Zap/35)
-- Called voipuser
-- Called 32
-- Called 33
-- Called 34
-- Called 35
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/voipuser-e989 is ringing
-- SIP/ voipuser-e989 answered Zap/1-1
What we need is to be able to populate the variable WHOANSWERED with info
*SIP/ voipuser*
In this case, or whoever answers next time.
Thanks in advance!
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Re: [asterisk-users] Urgent help = RUBY AGI
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zarko
Zivanovic
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
snip
In the meantime , do you happen to know if there is a way to call both macro
(M) and music on hold (m) in that $agi.exec line?
or is the right thing to do to place moh command in macro?
This should work:
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,mM(testing))
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Re: [asterisk-users] Urgent help = RUBY AGI
You can put multiple options in the dial command if that is what you are asking.
And by the way several emails, including a previous one of mine, told you to
use the M option and a macro.
In this email I gave you more detailed information but if you had done core
show application dial on CLI you should have been able to ask more directed
questions.
--
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mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 9:28 AM, Zarko Zivanovic wrote:
Jim thanks.
I will test this first thing in the morning as I am out of the office now. As
a matter of fact I cant wait to test this, as it has been the first
reasonable thing that looks like it could work.
In the meantime , do you happen to know if there is a way to call both macro
(M) and music on hold (m) in that $agi.exec line?
or is the right thing to do to place moh command in macro?
As I said, I cant wait to try it first thing in the morning and tell you (and
others) how it went. I am sure this will be the good reference to other
people looking for the same thing online as I have found quite a bunch of
similar open threads.
Zarko
On Tue, Jul 27, 2010 at 5:31 PM, Jim Dickenson dicken...@cfmc.com wrote:
I have never used 1.2.9.1 or anything in the 1.2.x range so I can not give
you an exact solution but I can tell you that the script that you are using
will not work. In the dial command you need to add the M option which will
call a macro when the call is connected. In that macro you can then find the
channel that answered the call and do what you want from there. You can call
another AGI or set variables or whatever. If agi.exec works like a dialplan
step then the dial step will hang if the call is answered and the
agi.get_variable statement will not execute unless the call was not answered.
Try
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,M(testing))
And then have something like this in extensions.conf
[macro-testing]
exten = s,1,DumpChan()
You will see that this macro runs when the call is answered and you will see
on the CLI all the variables that are available to you. ${CHANNEL} will have
SIP/ voipuser-e989 in your example below.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 7:21 AM, Zarko Zivanovic wrote:
Here’s something that should be easy for RUBY pro’s.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
# $agi.set_variable(' WHOANSWERED ',...)
retry if r.message.include?('BUSY')
end
when it’s executed it shows this in the console:
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx EXEC DIAL SIP/voipuserZap/32Zap/33Zap/34Zap/35
-- AGI Script Executing Application: (DIAL) Options:
(SIP/voipuserZap/32Zap/33Zap/34Zap/35)
-- Called voipuser
-- Called 32
-- Called 33
-- Called 34
-- Called 35
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/voipuser-e989 is ringing
-- SIP/ voipuser-e989 answered Zap/1-1
What we need is to be able to populate the variable WHOANSWERED with info
SIP/ voipuser
In this case, or whoever answers next time.
Thanks in advance!
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Re: [asterisk-users] Configuring X-lite for a remote user
thanks, i would try all the options out. I am very grateful _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- 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 app_fax, T.30, weird received faxes
On Monday, July 26, 2010 09:55:38 am Tzafrir Cohen wrote: I suppose I should make a list of known good packages, and put it on that FAQ page. GIMP is useless for FAX. Not only does it get the shape of the images wrong, it can only display the first page of a FAX. I am not familiar with gqview or feh. The package I usually use to display FAXes on Linux/BSD machines is okular. That seems to behave very well, unless you have a really old version. convert and the rest of imagemagick should handle multi-page tiff (e.g. convert it to PDF). libTIFF's tiff2pdf works well also. -- 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] CallerID disappear from CDR on transfer
Hi! 7) if john doe want to speak with caller assistant bridge the two lines using the transfer function of GXP2000 phone (REFER). After the transfer in the CDR i can't see the callerid of the caller, only data of the bridged call is reported. Any idea on what i can do to keep it ? Either store the Caller ID somewhere using your PHP script and then later add it to the CDR data of the other call, or look at at ImportVAR() or the SHARED() function (of which Tilghman has made a 1.4 backport available). In either case you will need to do some matching magic to logically link the two calls, maybe with the help of BRIDGEPEER. Possibly there are some transfer specific channel variables that can help you with that (like BLINDTRANSFER or TRANSFERSTATUS). This depends on your Asterisk version, though. Anyway, since you already determined where to transfer the call to that matching should not really be an issue. Philipp -- _ -- 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] Urgent help = RUBY AGI
On Tue, 27 Jul 2010, Jim Dickenson wrote: In this email I gave you more detailed information but if you had done core show application dial on CLI you should have been able to ask more directed questions. Maybe RTFHT* should have been the first response :) *) Read The Frick'n Help Text -- 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
Re: [asterisk-users] Urgent help = RUBY AGI
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Subject: Re: [asterisk-users] Urgent help = RUBY AGI Maybe RTFHT* should have been the first response :) *) Read The Frick'n Help Text Don't know about the Ruby part, but don't you have that book on 1.2 :)? -- _ -- 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] Peculiar Polycom IP6000 behavior
Here's a strange thing. I'm deploying Asterisk 1.6.2.9 with a pile of Cisco 79xx phones. For conference rooms we're using Polycom IP6000's. We bought two of them brand new. When I configure one phone with a username(SPIDR-3758)/password , it works fine. The other phone won't register with it's user(SPIDR-3749)/pass pair. When I try to use the first phone with the second user/pass pair, it won't work with that pair either. So, you'd think something was wrong with my sip.conf. I deleted the second entry and re-did it with new text. Still no joy. [SPIDR-3758](caryspider) mailbox=3...@default The above entry works, but: [SPIDR-3749](caryspider) mailbox=3...@default This one doesn't. [caryspider] looks like this: [caryspider](!) type=friend context=users host=dynamic secret=xx Any ideas? I'm stumped. Cassius Smith -- _ -- 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] IAX bandwidth optimisation
On 10-07-27 07:56 AM, G star wrote: How can I change that voice packet rate ? I think you want to read doc/rtp-packetization.txt in your Asterisk source. Leif. -- _ -- 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] CallerID disappear from CDR on transfer
Try to use local channel, and the pass the callerid of the caller to the local channel, an the later put this in CDR using h extention. -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com lechuck wrote: Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/ ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound trunk) 7) if john doe want to speak with caller assistant bridge the two lines using the transfer function of GXP2000 phone (REFER). After the transfer in the CDR i can't see the callerid of the caller, only data of the bridged call is reported. Any idea on what i can do to keep it ? thanks lechuck -- _ -- 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] sip peer becomes unreachable in Asterisk 1.6
Hello, I recently upgraded from asterisk 1.4 to 1.6. I am using the same SIP settings in sip.conf in this version also. I am facing a problem when a SIP client makes a call. When a SIP client registers to asterisk its status shows 'OK' and it is able to receive incoming calls. But as soon as this client make a call, its status becomes 'UNREACHABLE' and it cannot receive any incoming calls. Its status remains 'UNREACHABLE' until it re-registers again. I have faced this problem on various versions of 1.6 (1.6.2.0, 1.6.2.7, 1.6.1.1), but this never happened in the 1.4 (1.4.24, 1.4.26) versions. I have kept the SIP re-register time in the clients to a very small value to avoid becoming 'UNREACHABLE' for a long time, but his doesn't seem to be the solution. Is there any specific SIP settings which needs to be made in 1.6 to avoid this problem? I am using realtime sip. Some of my sip settings are rtcachefriends=yes rtupdate=no qualify=yes canreinvite=yes nat=yes Thanks -- _ -- 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 and Amazon Web Services
Anyone tried installing Asterisk in a AWS server? \\||/ Rod -- -- _ -- 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] MeetMe
Guys,
I put the option X and used the MEETME_EXIT_CONTEXT and it's working
thanks for the help!!!
=)
On Tue, Jul 27, 2010 at 9:07 AM, Felipe Figueiredo
felipe.figueired...@gmail.com wrote:
1) Ok, I'm using now self/peer on the feature map
2) there's no space in features.conf.
toca_macaco = 1,self/caller,Playback,tt-monkeys
But it's not working yet.
On Mon, Jul 26, 2010 at 11:33 PM, Tilghman Lesher tles...@digium.comwrote:
On Monday 26 July 2010 15:20:26 Felipe Figueiredo wrote:
Hi guys,
i'm trying to use the featuremap of features.conf inside the app
meetme,
but it's no working.
like:
_5XXX = {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hangup();
};
in features.conf:
toca_macaco = 123, peer, Playback,tt-monkeys
1) There is no peer when you invoke MeetMe. There is only a single call
leg.
You therefore want self or caller.
2) Kill the spaces on this line. All of them. Note that self,
caller,
or peer do not match anything and will thus signal Invalid
'ActivateOn'
specification for feature... at boot or reload. Similarly, there is a
dialplan application named Playback, but there is no dialplan
application
named Playback.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] Asterisk and Amazon Web Services
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson raand...@cyber-office.net wrote: Anyone tried installing Asterisk in a AWS server? \\||/ Rod -- -- _ -- 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 It probably works as well as it does virtualized other ways. I've seen peoples opinions on how virtualizing asterisk is a bad idea and might have trouble related to timing and hosting conferences. -- _ -- 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 and Amazon Web Services
Kyle Kienapfel wrote: On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson raand...@cyber-office.net wrote: Anyone tried installing Asterisk in a AWS server? \\||/ Rod -- -- _ -- 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 It probably works as well as it does virtualized other ways. I've seen peoples opinions on how virtualizing asterisk is a bad idea and might have trouble related to timing and hosting conferences. Thanks. I was pretty sure there would be conference call issues especially if they are recorded. Thanks, Rod -- -- _ -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
:-) I knew someone would bring up FreePBX. I have FreePBX installed and it's not good for Queues at all. It's using the reporting tool from Areski and Areski has recently released an upgrade to it which again is not what I want. There are few other programs that do this but really none that are neat in interface or useful in features. I guess no one else has any thoughts on this? Maybe there is none available? Thanks, Bruce On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg dbackeb...@gmail.comwrote: On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce bruceb...@gmail.com wrote: I seem to not be able to find any good open source Asterisk Queue Analyzer and Asterisk Log Analyzer on the web. google 'freepbx' It does some of what you want. For the rest of what you want, strongly consider paying a professional consultant. -- _ -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
There is none for free. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-27 6:12 PM, bruce bruce bruceb...@gmail.com wrote: :-) I knew someone would bring up FreePBX. I have FreePBX installed and it's not good for Queues at all. It's using the reporting tool from Areski and Areski has recently released an upgrade to it which again is not what I want. There are few other programs that do this but really none that are neat in interface or useful in features. I guess no one else has any thoughts on this? Maybe there is none available? Thanks, Bruce On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg dbackeb...@gmail.com wrote: On Mon, Jul 26... -- _ -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Try with: http://queuemetrics.com/download/qloaderd-1.17.tar.gz http://www.areski.net/asterisk-stat-v2/about.php http://www.micpc.com/qloganalyzer/ Regards, This is not the first time on this issue. I post in the past an in house solution. On Tue, Jul 27, 2010 at 5:38 PM, bruce bruce bruceb...@gmail.com wrote: :-) I knew someone would bring up FreePBX. I have FreePBX installed and it's not good for Queues at all. It's using the reporting tool from Areski and Areski has recently released an upgrade to it which again is not what I want. There are few other programs that do this but really none that are neat in interface or useful in features. I guess no one else has any thoughts on this? Maybe there is none available? Thanks, Bruce On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg dbackeb...@gmail.com wrote: On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce bruceb...@gmail.com wrote: I seem to not be able to find any good open source Asterisk Queue Analyzer and Asterisk Log Analyzer on the web. google 'freepbx' It does some of what you want. For the rest of what you want, strongly consider paying a professional consultant. -- _ -- 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 -- - Luis Morales Consultor de Tecnologia Cel: +(58)412-2352745 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- 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 and Amazon Web Services
Kyle Kienapfel wrote: On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson raand...@cyber-office.net wrote: Anyone tried installing Asterisk in a AWS server? I'd think twice about trying this, taking into account the recent spate of attacks to so many of us coming from Amazon EC2 and particularly their answer to complaints, which was something like Deal with it. /r -- _ -- 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] Peculiar Polycom IP6000 behavior
Hello Le 27/07/2010 20:57, Cassius Smith a écrit : Here's a strange thing. I'm deploying Asterisk 1.6.2.9 with a pile of Cisco 79xx phones. For conference rooms we're using Polycom IP6000's. We bought two of them brand new. [...] Any ideas? I'm stumped. If tour register server is outside your local network, you will have a problem as the IP [5|6|7]000 are registering using port 5060 on public IP (symetric nat) which will allow only one device. -- 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] Grab voicemail WAV file when done
I need to grab the voicemail WAV file once the voicemail command is done. Is there a hook to be notified that voicemail is done, and get the name of the recorded file? Thanks MD -- _ -- 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] Recording interface (pause/PLAY/RERECORD)
Is there a prebuild module/dialplan which gives me a nice interface to recording messages? Assuming I can't use the voicemail command, I need to offer users a way to record, playback, erase, rerecord, etc. I can probably do it through dialplan but it feels like I'm reinventing the wheel. Thanks, MD -- _ -- 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] Recording interface (pause/PLAY/RERECORD)
There's an app_record, and I believe app_dictate On 7/27/2010 7:39 PM, Michelle Dupuis wrote: Is there a prebuild module/dialplan which gives me a nice interface to recording messages? Assuming I can't use the voicemail command, I need to offer users a way to record, playback, erase, rerecord, etc. I can probably do it through dialplan but it feels like I'm reinventing the wheel. Thanks, MD -- _ -- 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 and Amazon Web Services
On Tue, Jul 27, 2010 at 4:16 PM, Randy R randulo2...@gmail.com wrote: Kyle Kienapfel wrote: On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson raand...@cyber-office.net wrote: Anyone tried installing Asterisk in a AWS server? I'd think twice about trying this, taking into account the recent spate of attacks to so many of us coming from Amazon EC2 and particularly their answer to complaints, which was something like Deal with it. /r -- _ -- 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 just set up fail2ban, I didn't even think to check where the attacks were coming from '109.170.0.33' r...@yzz some website '173.213.105.2' Cpanel. Duncanwierman.com DUNCANWIERMANCOM (NET-173-213-105-0-1) 173.213.105.0 - 173.213.105.31 '211.99.208.45' China '222.123.98.236' Maxnet in Bangkok '67.212.176.82' Redhotservers.com, cpanel '85.214.123.204' Germany, Parallels Plesk Panel '94.102.1.212' Turkey, cpanel '98.234.67.70' Comcast Looks like nobody is using ec2 to attack my wimpy little asterisk box in a corner, more shared hosting that i thought though. -- _ -- 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] Random DTMF Tones Only on heard on ATA
I have a couple of Linksys PAP2T-NA Grandstream HT-502 extensions that are receiving random DTMF tones on their side, but that are not heard by the outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have always had this issue. I am only using SIP on the Asterisk server and all extensions and trunks are set to rfc2833; outside of this issue DTMF operation works fine. I've included the Asterisk DTMF debug lines during an occurrence of this issue for extension 5211. The debug lines seem similar to a normal DTMF event with exception of the ...but want minimum 80... message. Any ideas? [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin passthrough '1' on SIP/5211-0078 [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF end '1' received on SIP/5211-0078, duration 39 ms [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF end accepted with begin '1' on SIP/5211-0078 [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF end '1' has duration 39 but want minimum 80, emulating on SIP/5211-0078 [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF end emulation of '1' queued on SIP/5211-0078 -- _ -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
On 10-07-27 06:08 PM, bruce bruce wrote: :-) I knew someone would bring up FreePBX. I have FreePBX installed and it's not good for Queues at all. It's using the reporting tool from Areski and Areski has recently released an upgrade to it which again is not what I want. There are few other programs that do this but really none that are neat in interface or useful in features. I guess no one else has any thoughts on this? Maybe there is none available? I have a client using QueueMetrics and they seem to be fairly pleased with it. Their response times on issues has been pretty good from what I can tell (I had the client communicate with them directly where necessary). Unless you build it yourself, I'm not sure there is any good + free queue metrics program. Queue's typically are a money generating adventure and as such makes sense for this type of application to be a pay-for system. Leif. -- _ -- 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] Grab voicemail WAV file when done
On 10-07-27 08:38 PM, Michelle Dupuis wrote: I need to grab the voicemail WAV file once the voicemail command is done. Is there a hook to be notified that voicemail is done, and get the name of the recorded file? Look at the 'externnotify' option to voicemail.conf. Leif Madsen. -- _ -- 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] Recording interface (pause/PLAY/RERECORD)
On 10-07-27 08:39 PM, Michelle Dupuis wrote:
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase, rerecord, etc.
I can probably do it through dialplan but it feels like I'm reinventing the
wheel.
Ya I haven't really written one generally yet, but here is something I'm
whipping together without testing :) I'll probably test it tomorrow during our
daily documentation session.
[globals]
CUSTOM_RECORDINGS=/var/lib/asterisk/sounds/en/custom
[subRecordPrompt]
exten = _[A-Za-z0-9].,1,NoOp()
; Safely handle extension name -- this will be our filename
same = n,Set(RecordedFilename=${FILTER(A-Za-z0-9,${EXTEN})})
same = n,Set(RandomNumber=${RAND()})
same = n,Answer()
; Record the prompt
same = n(record),Playback(please-enter-yourvm-messageafter-the-tone)
same = n,Wait(1)
same =
n,Record(${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav)
; Ask how we want to handle the recording
same = n(handle_recording),Read(ActionItem,vm-review,1)
; Verify we got values we expect
same = n,GotoIf($['${ActionItem}' = '1' | '${ActionItem}' = '2' |
${ActionItem}' = 3]?valid_action)
same = n,Playback(wrong-try-again-smarty)
same = n,Goto(handle_recording)
same = n(valid_action),NoOp()
; Handle the recording
; 1 accept
; 2 review
; 3 re-record
same = n,GotoIf($['${ActionItem}' = '1']?accept,1) ; keep this recording
same = n,GotoIf($['${ActionItem}' = '3']?record) ; re-record it
; If we get here they pressed 2
same =
n,Playback(${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber})
same = n,Goto(handle_recording)
exten = accept,1,Verbose(2,Recording accepted!)
same = n,System(mv
${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav
${GLOBAL(CUSTOM_RECORDINGS)}/${RecordedFilename}.wav)
exten = h,1,Verbose(2,Cleanup the file)
same = n,System(rm -f
${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav)
...or something like that.
Leif Madsen.
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Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
That's along the lines of what I was thinking, but how do you trap the DTMF
during record and cause that to end recording? I thought record kept going
until hangup?
MD
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:49 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
On 10-07-27 08:39 PM, Michelle Dupuis wrote:
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase, rerecord, etc.
I can probably do it through dialplan but it feels like I'm reinventing the
wheel.
Ya I haven't really written one generally yet, but here is something I'm
whipping together without testing :) I'll probably test it tomorrow during our
daily documentation session.
[globals]
CUSTOM_RECORDINGS=/var/lib/asterisk/sounds/en/custom
[subRecordPrompt]
exten = _[A-Za-z0-9].,1,NoOp()
; Safely handle extension name -- this will be our filename
same = n,Set(RecordedFilename=${FILTER(A-Za-z0-9,${EXTEN})})
same = n,Set(RandomNumber=${RAND()})
same = n,Answer()
; Record the prompt
same = n(record),Playback(please-enter-yourvm-messageafter-the-tone)
same = n,Wait(1)
same =
n,Record(${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav)
; Ask how we want to handle the recording
same = n(handle_recording),Read(ActionItem,vm-review,1)
; Verify we got values we expect
same = n,GotoIf($['${ActionItem}' = '1' | '${ActionItem}' = '2' |
${ActionItem}' = 3]?valid_action)
same = n,Playback(wrong-try-again-smarty)
same = n,Goto(handle_recording)
same = n(valid_action),NoOp()
; Handle the recording
; 1 accept
; 2 review
; 3 re-record
same = n,GotoIf($['${ActionItem}' = '1']?accept,1) ; keep this recording
same = n,GotoIf($['${ActionItem}' = '3']?record) ; re-record it
; If we get here they pressed 2
same =
n,Playback(${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber})
same = n,Goto(handle_recording)
exten = accept,1,Verbose(2,Recording accepted!)
same = n,System(mv
${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav
${GLOBAL(CUSTOM_RECORDINGS)}/${RecordedFilename}.wav)
exten = h,1,Verbose(2,Cleanup the file)
same = n,System(rm -f
${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav)
...or something like that.
Leif Madsen.
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Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
How can I found more info about them? The voip-info wiki seems to have a one line description only Hopefully I don't have to read the source code to figure out the features;( Thanks, MD From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan [sherwood.mcgo...@gmail.com] Sent: Tuesday, July 27, 2010 8:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD) There's an app_record, and I believe app_dictate On 7/27/2010 7:39 PM, Michelle Dupuis wrote: Is there a prebuild module/dialplan which gives me a nice interface to recording messages? Assuming I can't use the voicemail command, I need to offer users a way to record, playback, erase, rerecord, etc. I can probably do it through dialplan but it feels like I'm reinventing the wheel. Thanks, MD -- _ -- 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] Grab voicemail WAV file when done
The problem is that I need to catch the filename in the dialplan, since I will be recording several other files and concatenating them with SOX. MD From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen [leif.mad...@asteriskdocs.org] Sent: Tuesday, July 27, 2010 9:22 PM To: Asterisk Users List Subject: Re: [asterisk-users] Grab voicemail WAV file when done On 10-07-27 08:38 PM, Michelle Dupuis wrote: I need to grab the voicemail WAV file once the voicemail command is done. Is there a hook to be notified that voicemail is done, and get the name of the recorded file? Look at the 'externnotify' option to voicemail.conf. Leif Madsen. -- _ -- 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] Recording interface (pause/PLAY/RERECORD)
On Tue, Jul 27, 2010 at 10:05 PM, Michelle Dupuis mdup...@ocg.ca wrote: Hopefully I don't have to read the source code to figure out the features;( *CLI core show application Record -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Recording interface (pause/PLAY/RERECORD)
Terrific! I tried the same for app_dictate but got a very brief usage description (but not really a description of the features or what it does). Is there any other documentation on app_dictate out there? Thanks MD From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger [paul.belan...@polybeacon.com] Sent: Tuesday, July 27, 2010 10:10 PM To: Asterisk Users List Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD) On Tue, Jul 27, 2010 at 10:05 PM, Michelle Dupuis mdup...@ocg.ca wrote: Hopefully I don't have to read the source code to figure out the features;( *CLI core show application Record -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Recording interface (pause/PLAY/RERECORD)
Record does not continue until the end of the call, it records until the
# is pressed or the max duration is reached:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Record
Enjoy
On 7/27/2010 9:00 PM, Michelle Dupuis wrote:
That's along the lines of what I was thinking, but how do you trap the DTMF
during record and cause that to end recording? I thought record kept going
until hangup?
MD
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:49 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)
On 10-07-27 08:39 PM, Michelle Dupuis wrote:
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase, rerecord, etc.
I can probably do it through dialplan but it feels like I'm reinventing the
wheel.
Ya I haven't really written one generally yet, but here is something I'm
whipping together without testing :) I'll probably test it tomorrow during our
daily documentation session.
[globals]
CUSTOM_RECORDINGS=/var/lib/asterisk/sounds/en/custom
[subRecordPrompt]
exten = _[A-Za-z0-9].,1,NoOp()
; Safely handle extension name -- this will be our filename
same = n,Set(RecordedFilename=${FILTER(A-Za-z0-9,${EXTEN})})
same = n,Set(RandomNumber=${RAND()})
same = n,Answer()
; Record the prompt
same = n(record),Playback(please-enter-yourvm-messageafter-the-tone)
same = n,Wait(1)
same =
n,Record(${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav)
; Ask how we want to handle the recording
same = n(handle_recording),Read(ActionItem,vm-review,1)
; Verify we got values we expect
same = n,GotoIf($['${ActionItem}' = '1' | '${ActionItem}' = '2' |
${ActionItem}' = 3]?valid_action)
same = n,Playback(wrong-try-again-smarty)
same = n,Goto(handle_recording)
same = n(valid_action),NoOp()
; Handle the recording
; 1 accept
; 2 review
; 3 re-record
same = n,GotoIf($['${ActionItem}' = '1']?accept,1) ; keep this recording
same = n,GotoIf($['${ActionItem}' = '3']?record) ; re-record it
; If we get here they pressed 2
same =
n,Playback(${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber})
same = n,Goto(handle_recording)
exten = accept,1,Verbose(2,Recording accepted!)
same = n,System(mv
${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav
${GLOBAL(CUSTOM_RECORDINGS)}/${RecordedFilename}.wav)
exten = h,1,Verbose(2,Cleanup the file)
same = n,System(rm -f
${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav)
...or something like that.
Leif Madsen.
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Re: [asterisk-users] MeetMe
Hi Felipe,
Glad to know that it worked , could you kindly post the complete dialplan
you wrote to achieve it.
I would also love to test it.
Thanks in Advance
Shiju V.Joseph
From:
Felipe Figueiredo felipe.figueired...@gmail.com
To:
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date:
07/28/2010 01:27 AM
Subject:
Re: [asterisk-users] MeetMe
Sent by:
asterisk-users-boun...@lists.digium.com
Guys,
I put the option X and used the MEETME_EXIT_CONTEXT and it's working
thanks for the help!!!
=)
On Tue, Jul 27, 2010 at 9:07 AM, Felipe Figueiredo
felipe.figueired...@gmail.com wrote:
1) Ok, I'm using now self/peer on the feature map
2) there's no space in features.conf.
toca_macaco = 1,self/caller,Playback,tt-monkeys
But it's not working yet.
On Mon, Jul 26, 2010 at 11:33 PM, Tilghman Lesher tles...@digium.com
wrote:
On Monday 26 July 2010 15:20:26 Felipe Figueiredo wrote:
Hi guys,
i'm trying to use the featuremap of features.conf inside the app
meetme,
but it's no working.
like:
_5XXX = {
Set(DYNAMIC_FEATURES=toca_macaco);
MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF
Hangup();
};
in features.conf:
toca_macaco = 123, peer, Playback,tt-monkeys
1) There is no peer when you invoke MeetMe. There is only a single call
leg.
You therefore want self or caller.
2) Kill the spaces on this line. All of them. Note that self,
caller,
or peer do not match anything and will thus signal Invalid
'ActivateOn'
specification for feature... at boot or reload. Similarly, there is a
dialplan application named Playback, but there is no dialplan
application
named Playback.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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