Re: [asterisk-users] Results of security honeypot experiment - scraping for IP's/credentials ?
Very interesting ! Thank you very much for sharing this here. 2015-06-03 3:12 GMT+02:00 Michelle Dupuis mdup...@ocg.ca: The results of a security experiment were published this week, in which an Asterisk PBX was set out in the wild to see who would attack it and how: http://www.telium.ca/?honeypot1 What I find particularly interesting is that people/bots are scraping support websites looking for valid IP's of PBX's, and valid credentials! A good reminder to everyone on this list to not publish the IP of their PBX's, or even account names (in postings) as they will be quickly targeted -- _ -- 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] Forward loop protection...
On Tuesday 02 Jun 2015, Carlos Chavez wrote: Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensión number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What is an easy way to prevent dumb people from creating a loop? There currently is no easy way to prevent an infinite forwarding loop. If you come up with one, then you might well earn yourself a Nobel Prize for solving the Halting Problem . The obvious bodge is to set a hard limit on depth of recursion; if an actual real, live person is not reached within, say, five hops then the call should go to (the originally-called party's) voicemail. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan?
Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a button in the interface of its SIP application. For example, we can use the lib of Linphone and implement a GUI over it, having a new button to open doors and gates. So, the callee will not have to remember about codes, because there will be a button in someplace to be touched. When the button be touched, during an ongoing call, the software (SIP client) will sends a request to Asterisk executes the gate = 9,self/callee,System,insert command here , for example. So, it will works like the user pressing number 9. I will take a look at applicationmap in features.conf to understand what exactly can be done. But, let me ask you: This idea seems to be good to run during ongoing calls. What about moments when there is no ongoing call? That is, can Asterisk execute a dial plan (maybe by means of some kind of SIP request received from the SIP client) even without establishing a call? Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen [kevin.lar...@pioneerballoon.com] Enviado: quarta-feira, 3 de junho de 2015 10:29 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan? Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten = 1234,1,System(echo This is a test / var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a call. During the call, the caller asks me to execute a command (ex: to open a door or gate). In this case, what have I to program in dial plan to Asterisk execute System() again? Is it possible to execute a dial plan even during an ongoing call? Finally, lets suppose I want to use my softphone to execute a dial plan, even without establishing a call (no session with target 1234). For example, If I decide to open a dor or gate using my softphone, without existing an ongoing call, what have I to program in dial plan to Asterisk executes System(). Is this idea possible? Any hint will be very hepful! I love this question, simply because it allows me to talk about one of the neatest features I programmed into my system that barely anyone knows exists. Plus it lines up pretty much exactly with what you are trying to do. We have our gate control system tied into our Asterisk phone system so it is possible to dial a code on the phone and open the entrance gate to let someone in after hours. Only problem is this happens so rarely that no one (myself included) ever remembered the code. Thus a search for a better way. Now, when someone uses the gate phone to request entry, I change the caller ID on the display of the person who answers to read Press 9 to open gate. During the call, they can hit 9 at any time and the gate will open for them. Up until they answer, the caller ID reads Gate Phone, but when they answer, it changes to that text. The part about opening the gate is the magic piece you want to look into. Read up on applicationmap in features.conf. It's pretty simple and very effective. Here is what mine looks like. I am going to replace my actual command with insert command here. gate = 9,self/callee,System,insert command here ; Custom application to open the gate. This says that this feature is active in the 'gate' context of my dialplan. The dialing pattern it is looking for is a 9. 'self' tells it to activate on the channel that dialed it and callee says that the person receiving the call is the only one that can activate it (otherwise the person at the gate phone could hit 9 to open it). I am running the System dialplan application and passing it the insert command here value. Everything after the ';' is a comment as normal. The insert command here is equivalent to what you would put inside the '()' if it were in the dialplan (i.e. 'System(insert command here)'). Pretty straightforward to get it working once you know what to look for. Let me know if you want to know how I manipulate the Caller ID upon answering the call to give the instructions to the callee on how to open the gate/door. -- _ -- 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] RES: RES: RES: How to invoke a binary file from the dial plan?
Hi Kevin.
Thank you!
I will examine it.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen
[kevin.lar...@pioneerballoon.com]
Enviado: quarta-feira, 3 de junho de 2015 10:34
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] RES: RES: How to invoke a binary file from the
dial plan?
I love this question, simply because it allows me to talk about one
of the neatest features I programmed into my system that barely
anyone knows exists. Plus it lines up pretty much exactly with what
you are trying to do.
We have our gate control system tied into our Asterisk phone system
so it is possible to dial a code on the phone and open the entrance
gate to let someone in after hours. Only problem is this happens so
rarely that no one (myself included) ever remembered the code. Thus
a search for a better way.
Now, when someone uses the gate phone to request entry, I change the
caller ID on the display of the person who answers to read Press 9
to open gate. During the call, they can hit 9 at any time and the
gate will open for them. Up until they answer, the caller ID reads
Gate Phone, but when they answer, it changes to that text.
The part about opening the gate is the magic piece you want to look
into. Read up on applicationmap in features.conf. It's pretty simple
and very effective. Here is what mine looks like. I am going to
replace my actual command with insert command here.
gate = 9,self/callee,System,insert command here ; Custom
application to open the gate.
This says that this feature is active in the 'gate' context of my
dialplan. The dialing pattern it is looking for is a 9. 'self' tells
it to activate on the channel that dialed it and callee says that
the person receiving the call is the only one that can activate it
(otherwise the person at the gate phone could hit 9 to open it). I
am running the System dialplan application and passing it the
insert command here value. Everything after the ';' is a comment
as normal. The insert command here is equivalent to what you would
put inside the '()' if it were in the dialplan (i.e. 'System(insert
command here)').
Pretty straightforward to get it working once you know what to look
for. Let me know if you want to know how I manipulate the Caller ID
upon answering the call to give the instructions to the callee on
how to open the gate/door.
I just realized I said one piece wrong in this. 'gate' is not the context, it
is the dynamic feature designator. I can illustrate this better by posting my
front gate context.
[front_gate]
exten = number gate dials goes here,1,Set(__DYNAMIC_FEATURES=gate)
same = n,Goto(frontgate_queue,${EXTEN},1)
--
_
-- 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] RES: RES: How to invoke a binary file from the dial plan?
Great On Jun 3, 2015 9:29 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten = 1234,1,System(echo This is a test / var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a call. During the call, the caller asks me to execute a command (ex: to open a door or gate). In this case, what have I to program in dial plan to Asterisk execute System() again? Is it possible to execute a dial plan even during an ongoing call? Finally, lets suppose I want to use my softphone to execute a dial plan, even without establishing a call (no session with target 1234). For example, If I decide to open a dor or gate using my softphone, without existing an ongoing call, what have I to program in dial plan to Asterisk executes System(). Is this idea possible? Any hint will be very hepful! I love this question, simply because it allows me to talk about one of the neatest features I programmed into my system that barely anyone knows exists. Plus it lines up pretty much exactly with what you are trying to do. We have our gate control system tied into our Asterisk phone system so it is possible to dial a code on the phone and open the entrance gate to let someone in after hours. Only problem is this happens so rarely that no one (myself included) ever remembered the code. Thus a search for a better way. Now, when someone uses the gate phone to request entry, I change the caller ID on the display of the person who answers to read Press 9 to open gate. During the call, they can hit 9 at any time and the gate will open for them. Up until they answer, the caller ID reads Gate Phone, but when they answer, it changes to that text. The part about opening the gate is the magic piece you want to look into. Read up on applicationmap in features.conf. It's pretty simple and very effective. Here is what mine looks like. I am going to replace my actual command with insert command here. gate = 9,self/callee,System,insert command here ; Custom application to open the gate. This says that this feature is active in the 'gate' context of my dialplan. The dialing pattern it is looking for is a 9. 'self' tells it to activate on the channel that dialed it and callee says that the person receiving the call is the only one that can activate it (otherwise the person at the gate phone could hit 9 to open it). I am running the System dialplan application and passing it the insert command here value. Everything after the ';' is a comment as normal. The insert command here is equivalent to what you would put inside the '()' if it were in the dialplan (i.e. 'System(insert command here)'). Pretty straightforward to get it working once you know what to look for. Let me know if you want to know how I manipulate the Caller ID upon answering the call to give the instructions to the callee on how to open the gate/door. -- _ -- 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] RES: RES: How to invoke a binary file from the dial plan?
I love this question, simply because it allows me to talk about one
of the neatest features I programmed into my system that barely
anyone knows exists. Plus it lines up pretty much exactly with what
you are trying to do.
We have our gate control system tied into our Asterisk phone system
so it is possible to dial a code on the phone and open the entrance
gate to let someone in after hours. Only problem is this happens so
rarely that no one (myself included) ever remembered the code. Thus
a search for a better way.
Now, when someone uses the gate phone to request entry, I change the
caller ID on the display of the person who answers to read Press 9
to open gate. During the call, they can hit 9 at any time and the
gate will open for them. Up until they answer, the caller ID reads
Gate Phone, but when they answer, it changes to that text.
The part about opening the gate is the magic piece you want to look
into. Read up on applicationmap in features.conf. It's pretty simple
and very effective. Here is what mine looks like. I am going to
replace my actual command with insert command here.
gate = 9,self/callee,System,insert command here ; Custom
application to open the gate.
This says that this feature is active in the 'gate' context of my
dialplan. The dialing pattern it is looking for is a 9. 'self' tells
it to activate on the channel that dialed it and callee says that
the person receiving the call is the only one that can activate it
(otherwise the person at the gate phone could hit 9 to open it). I
am running the System dialplan application and passing it the
insert command here value. Everything after the ';' is a comment
as normal. The insert command here is equivalent to what you would
put inside the '()' if it were in the dialplan (i.e. 'System(insert
command here)').
Pretty straightforward to get it working once you know what to look
for. Let me know if you want to know how I manipulate the Caller ID
upon answering the call to give the instructions to the callee on
how to open the gate/door.
I just realized I said one piece wrong in this. 'gate' is not the context,
it is the dynamic feature designator. I can illustrate this better by
posting my front gate context.
[front_gate]
exten = number gate dials goes here,1,Set(__DYNAMIC_FEATURES=gate)
same = n,Goto(frontgate_queue,${EXTEN},1)
--
_
-- 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] Forward loop protection...
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Wednesday, June 3, 2015 3:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Forward loop protection... On Tuesday 02 Jun 2015, Carlos Chavez wrote: Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensión number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What is an easy way to prevent dumb people from creating a loop? There currently is no easy way to prevent an infinite forwarding loop. If you come up with one, then you might well earn yourself a Nobel Prize for solving the Halting Problem . The obvious bodge is to set a hard limit on depth of recursion; if an actual real, live person is not reached within, say, five hops then the call should go to (the originally-called party's) voicemail. -- AJS Deciding on the mailbox to use is problematic! The dialed-party may be away for an extended period and wants voice mail handled by the forwarded-to party. --Don -- _ -- 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] RES: RES: How to invoke a binary file from the dial plan?
Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten = 1234,1,System(echo This is a test /var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a call. During the call, the caller asks me to execute a command (ex: to open a door or gate). In this case, what have I to program in dial plan to Asterisk execute System() again? Is it possible to execute a dial plan even during an ongoing call? Finally, lets suppose I want to use my softphone to execute a dial plan, even without establishing a call (no session with target 1234). For example, If I decide to open a dor or gate using my softphone, without existing an ongoing call, what have I to program in dial plan to Asterisk executes System(). Is this idea possible? Any hint will be very hepful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen [kevin.lar...@pioneerballoon.com] Enviado: terça-feira, 2 de junho de 2015 17:50 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: How to invoke a binary file from the dial plan? Ok. Thanks for the hint. But, what exactly is a System() dialplan application? Is it a kind of command that i can call in dial plan? I will look for System() related to dial plans. From the Asterisk CLI type: core show application System It will print out the syntax for the command. One of the easier dialplan applications. exten = 1234,1,System(echo This is a test /var/log/asterisk/test.txt) That line would use the Linux echo command to place the text This is a test into a file named test.txt located in the /var/log/asterisk directory. -- _ -- 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] Forward loop protection...
Deciding on the mailbox to use is problematic! The dialed-party may be away for an extended period and wants voice mail handled by the forwarded-to party. And then you have the users who would work around this by sharing their voicemail passwords. Not quite as bad as sharing your computer log on credentials, but still, something I would like to avoid if possible.-- _ -- 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] RES: RES: How to invoke a binary file from the dial plan?
Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten = 1234,1,System(echo This is a test / var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a call. During the call, the caller asks me to execute a command (ex: to open a door or gate). In this case, what have I to program in dial plan to Asterisk execute System() again? Is it possible to execute a dial plan even during an ongoing call? Finally, lets suppose I want to use my softphone to execute a dial plan, even without establishing a call (no session with target 1234). For example, If I decide to open a dor or gate using my softphone, without existing an ongoing call, what have I to program in dial plan to Asterisk executes System(). Is this idea possible? Any hint will be very hepful! I love this question, simply because it allows me to talk about one of the neatest features I programmed into my system that barely anyone knows exists. Plus it lines up pretty much exactly with what you are trying to do. We have our gate control system tied into our Asterisk phone system so it is possible to dial a code on the phone and open the entrance gate to let someone in after hours. Only problem is this happens so rarely that no one (myself included) ever remembered the code. Thus a search for a better way. Now, when someone uses the gate phone to request entry, I change the caller ID on the display of the person who answers to read Press 9 to open gate. During the call, they can hit 9 at any time and the gate will open for them. Up until they answer, the caller ID reads Gate Phone, but when they answer, it changes to that text. The part about opening the gate is the magic piece you want to look into. Read up on applicationmap in features.conf. It's pretty simple and very effective. Here is what mine looks like. I am going to replace my actual command with insert command here. gate = 9,self/callee,System,insert command here ; Custom application to open the gate. This says that this feature is active in the 'gate' context of my dialplan. The dialing pattern it is looking for is a 9. 'self' tells it to activate on the channel that dialed it and callee says that the person receiving the call is the only one that can activate it (otherwise the person at the gate phone could hit 9 to open it). I am running the System dialplan application and passing it the insert command here value. Everything after the ';' is a comment as normal. The insert command here is equivalent to what you would put inside the '()' if it were in the dialplan (i.e. 'System(insert command here)'). Pretty straightforward to get it working once you know what to look for. Let me know if you want to know how I manipulate the Caller ID upon answering the call to give the instructions to the callee on how to open the gate/door.-- _ -- 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] sedwa...@sedwards.com causes me to be knocked off the list
Someone on this list uses the address @sedwards.com I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either. Part of my mail servers reject these emails because they cannot be replied to, or are likely to be spam. Every so often I get a mail from the list management to say that I've been unsubscribed because of excessive bounces and it takes a single click to re-register. It's a bit of a niggle for me. What do you think I should do? Change my servers so that I don't check sender domains? -- Regards, Giles Coochey, CCNP, CCNA, CCNAS NetSecSpec Ltd +44 (0) 8444 780677 +44 (0) 7584 634135 http://www.coochey.net http://www.netsecspec.co.uk gi...@coochey.net smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan?
Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a button in the interface of its SIP application. For example, we can use the lib of Linphone and implement a GUI over it, having a new button to open doors and gates. So, the callee will not have to remember about codes, because there will be a button in someplace to be touched. When the button be touched, during an ongoing call, the software (SIP client) will sends a request to Asterisk executes the gate = 9,self/callee,System,insert command here , for example. So, it will works like the user pressing number 9. I will take a look at applicationmap in features.conf to understand what exactly can be done. But, let me ask you: This idea seems to be good to run during ongoing calls. What about moments when there is no ongoing call? That is, can Asterisk execute a dial plan (maybe by means of some kind of SIP request received from the SIP client) even without establishing a call? The way I would probably approach what you want to do is that the button action state would be dependent on if you are in a call or not. If you are in a call, it sends whatever DTMF digits you want to use for this feature. If you are not in a call, it could dial an extension whose purpose is to do the same thing. I have an outside number that when dialed checks that your caller id number is in an approved list and if it is, sends the gate open signal. This is the same gate open signal that the feature code uses (the call to System()), it is just reached by making a sip call. Nothing says a call has to connect two phones together. You can answer the call inside of Asterisk and do stuff based on what number you called or what digits the caller enters with their keypads. Lot's of opportunity to make the system do exactly what you want.-- _ -- 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] sedwa...@sedwards.com causes me to be knocked off the list
On Jun 3, 2015, at 11:01 AM, Giles Coochey gi...@coochey.net wrote: Someone on this list uses the address @sedwards.com I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either. Really? That's Steve Edwards and he's actively posts here. I show his MX as: sedwards.commail exchanger = 20 smtp1.sedwards.com. sedwards.commail exchanger = 10 smtp0.sedwards.com. And whois reports it's registered under Steve out of California. Doug -- _ -- 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] sslv3 alert unexpected message
hello, my webrtc calls ends after ~60seconds with res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating. any ideas where can be problem? or howto debug this problem? asterisk13.4.0-rc1 + sipml5 latest (chrome,firefox) -- --- Marek Cervenka === -- _ -- 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] sedwa...@sedwards.com causes me to be knocked off the list
sedwa...@sedwards.com causes me to be knocked off the list I feel so powerful :) On Wed, 3 Jun 2015, Giles Coochey wrote: Someone on this list uses the address @sedwards.com That would be me. Although, I prefer to use asterisk@sedwards.com for 'list related' emails. I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either. It is, there is, and please look harder. Part of my mail servers reject these emails because they cannot be replied to, or are likely to be spam. Every so often I get a mail from the list management to say that I've been unsubscribed because of excessive bounces and it takes a single click to re-register. It's a bit of a niggle for me. What do you think I should do? Change my servers so that I don't check sender domains? DNS and email servers are not my area of expertise, so if you think I've misconfigured something on my side, please let me know. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: RES: RES: RES: How to invoke a binary file from the dial plan?
Ok Kevin. Thank you for the information. Now, I will try to build a prototype to see how everything works. If I have a new doubt, I will post it here. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Kevin Larsen [kevin.lar...@pioneerballoon.com] Enviado: quarta-feira, 3 de junho de 2015 12:26 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan? Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a button in the interface of its SIP application. For example, we can use the lib of Linphone and implement a GUI over it, having a new button to open doors and gates. So, the callee will not have to remember about codes, because there will be a button in someplace to be touched. When the button be touched, during an ongoing call, the software (SIP client) will sends a request to Asterisk executes the gate = 9,self/callee,System,insert command here , for example. So, it will works like the user pressing number 9. I will take a look at applicationmap in features.conf to understand what exactly can be done. But, let me ask you: This idea seems to be good to run during ongoing calls. What about moments when there is no ongoing call? That is, can Asterisk execute a dial plan (maybe by means of some kind of SIP request received from the SIP client) even without establishing a call? The way I would probably approach what you want to do is that the button action state would be dependent on if you are in a call or not. If you are in a call, it sends whatever DTMF digits you want to use for this feature. If you are not in a call, it could dial an extension whose purpose is to do the same thing. I have an outside number that when dialed checks that your caller id number is in an approved list and if it is, sends the gate open signal. This is the same gate open signal that the feature code uses (the call to System()), it is just reached by making a sip call. Nothing says a call has to connect two phones together. You can answer the call inside of Asterisk and do stuff based on what number you called or what digits the caller enters with their keypads. Lot's of opportunity to make the system do exactly what you want. -- _ -- 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] default features
Hi We are provisioning some default features to our customers like automated answer when outside labor time, rerouting when Subscriber Absent, and so. These are macro calls embedded in key points inside the dialplan. Since not all customers need/want all features and they (the features) are customized, I named them like [macro-feature1-ClientA], [macro-feature4-ClientF], ... Question: is there some built-in way to know if macro feature1-ClientA is defined? Something like ExecIfMacro(feature1-ClientA)?macro(feature1-ClientA):Goto(...). For the moment, we are using an external command like asterisk -rx 'dialplan show'| grep ... to set a __variable and we use it later to decide to execute or not the suspicious macro. Cheers Ethy -- _ -- 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] default features
Question: is there some built-in way to know if macro feature1-ClientA is defined? Something liken ExecIfMacro(feature1-ClientA)?macro(feature1-ClientA):Goto(...). A macro is a context, so DIALPLAN_EXISTS should work if you specify an extension and priority that's in the macro (presumably, s,1). -- _ -- 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] sedwa...@sedwards.com causes me to be knocked off the list
I agree with Doug, everything looks legit for sedwards.com from my point of view. Giles, maybe there's something wrong with the DNS server you're using? James Cass http://goog_987864563 jcas...@gmail.com On Wed, Jun 3, 2015 at 11:01 AM, Giles Coochey gi...@coochey.net wrote: Someone on this list uses the address @sedwards.com I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either. Part of my mail servers reject these emails because they cannot be replied to, or are likely to be spam. Every so often I get a mail from the list management to say that I've been unsubscribed because of excessive bounces and it takes a single click to re-register. It's a bit of a niggle for me. What do you think I should do? Change my servers so that I don't check sender domains? -- Regards, Giles Coochey, CCNP, CCNA, CCNAS NetSecSpec Ltd +44 (0) 8444 780677 +44 (0) 7584 634135 http://www.coochey.net http://www.netsecspec.co.uk gi...@coochey.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sedwa...@sedwards.com causes me to be knocked off the list
Someone on this list uses the address @sedwards.com I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either. Part of my mail servers reject these emails because they cannot be replied to, or are likely to be spam. Every so often I get a mail from the list management to say that I've been unsubscribed because of excessive bounces and it takes a single click to re-register. It's a bit of a niggle for me. What do you think I should do? Change my servers so that I don't check sender domains? If the SMTP session is saying MAIL FROM: xx...@sedwards.com, and if sedwards.com has no MX or A addresses on file with DNS, then I think it's appropriate to reject the mail at that stage, either permanently or temporarily. The latter is probably better in case there's a transient DNS problem. Your server should send an error message in response to the MAIL FROM command. The Asterisk mailing list servers should *not* be forwarding messages to list subscribers in this way. Most of the big mail providers are now performing SPF (or similar) validation on the MAIL FROM addresses, and will reject a lot of mail which is reflected through mailing-list servers. Current practice is for mailing-list servers to rewrite the sender address in the envelope (to a form which identifies their own domain as the intermediate relay/sender) or to just use an address such as asterisk-us...@list.digium.com as the MAIL FROM address. Now... if you're digging into the message headers themselves (e.g. looking at the From: header) and rejecting the mail because the address therein isn't legitimate... that's a different issue and a bigger problem. Your mail server can't do this during the initial SMTP handshake... only after it accepts the entire message from the sending system. This creates an anomalous situation, because your server provisionally accepted the message (by saying OK to the MAIL FROM, RCPT TO, and DATA requests from the sender), and then rejected the message as undeliverable at the last moment. Not a good thing, according to the SMTP spec, and it's not surprising that some servers will consider this a practice which justifies blocking further deliveries to your system. In this case, you'd be better off accepting the mail normally via SMTP, using your spam filter to tag it with a suspicious label, and the filing it in a spam folder or just discarding it after reception. From the point of view of the sending system, it will have been accepted normally (rather than rejected or bounced) . One thing you definitely should *NOT* ever do, is accept a mail message via SMTP (saying OK), determine that you think it's spam, and then have your mail server mail back a rejected bounce message to the sender. This is bad, bad, bad. It causes back-scatter - if mail is sent with a forged sender address (which is quite common) the poor schlub whose address was stolen for this purpose is likely to get a reject message for every copy of the forged mail. This can put a horrible burden on the victim's mail server. -- _ -- 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
