date:20111229

Re: [asterisk-users] DTMF Testing software to test IVR system

2011-12-29 Thread Sammy Govind

Easy, use Read() to capture the incoming DTMF from Server-B

Server-A <> Server-B
Initiate-Call -> AnswerCall()
SendDTMF(5)--> Read()
Read()<-SendDTMF(4)
SendDTMF(3)--> Read()
Read()<-SendDTMF(2)
SendDTMF(1)--> Read()


Put proper GOTOIFs after reads if you like.

--
Regards,
Sammy

On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati  wrote:

> I originate calls from .call file and 1 channel I have at A server A and
> another channel at B server.
>
> *A server code is below:-*
>
> exten => 43689956,1,Answer()
> same => n,Wait(5)
> same => n,SendDTMF(1)
> same => n,NoOp(==   ${CHANNEL(state)}==> state)
> same => n,wait(2)
> same => n,SendDTMF(123456789012345#)
> same => n,NoOp(==   ${CHANNEL(state)}==> state)
> same => n,Hangup()
>
>  _  _
> |  A server  |  ___DTMF Send_=> | B server   |
> |_|  <=--- Responce -   |_|
>
> *B server code is below:-*
> At B server call come to 201 extension which is mention here..
>
> exten => _20[1-6],1,Answer()
> same => n,Ringing()
> same => n,wait(2)
> same => n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
> AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
> same => n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
> $[${EXTEN}=205] ||
> $[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
> same => n,Hangup()
>
> Now I can send the DTMF from A to B. But How I will get the responce at
> server A. I checked all the channels variable but they didn't reply status
> of B server channel. All information I will get of server A. Main problem
> is that control reach to AGI and then I don't have any rights to do any
> update or modification on AGI. So if I can work on request and responce
> then it will be the last solution as per my knowledge.
>
> Is this possible with the dialplan or I am just westing time?
>
>
>
> On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger wrote:
>
>> On 11-12-28 03:25 AM, virendra bhati wrote:
>>
>>> Hi list,
>>>
>>> Is there any way in asterisk by which I make a call from server and then
>>> dialplan(IVR system) gets DTMF from it. I mean to say that automatically
>>> DTMF is sended by channels as per user defined,
>>>
>>> I read there is an application sendDTMF but I don't know how we can used
>>> it?
>>>
>>> like A script make the call by using localdail, .call file or any method.
>>> And after landing the call we send dtmf to IVR system automatically as
>>> per
>>> my script..
>>>
>>>
>>> *extensions.conf:-*
>>>
>>>
>>> exten =>  1234,1,Answer()
>>>  same =>  n,Read(value,**pleasePress1forSupportPress2fo**
>>> rHelp,1,,10)
>>>  same =>  n,NoOp(${value})
>>>  same =>  n,ExecIf($[${value}=1]?Goto(**suppot,1))
>>>  same =>  n,ExecIf($[${value}=2]?Goto(**help,1))
>>>  same =>  n,Hangup()
>>>
>>> exten=>  support,1,Answer()
>>>  same =>  n,NoOp(you are at support section)
>>>  same =>  n,Hangup()
>>>
>>> exten=>  help,1,Answer()
>>>  same =>  n,NoOp(you are at help section)
>>>  same =>  n,Hangup()
>>>
>>>  We have DTMF based tests for the testsuite[1] that you could use.
>>
>> [1] 
>> http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/
>> --
>> Paul Belanger
>> Digium, Inc. | Software Developer
>> twitter: pabelanger | IRC: pabelanger (Freenode)
>> Check us out at: http://digium.com & http://asterisk.org
>>
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>  http://www.asterisk.org/hello
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>>  
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-user

Re: [asterisk-users] Function TESTTIME example [SOLVED]

2011-12-29 Thread Olivier

2011/12/29, Mindaugas Jasiulis :
> AEL2 ifTime use it too. AEL2 ifTime become the same GoToIfTime in the
> dialplan :)

I feel a bit shameful but I must say that it's now working on my system !
I don't understand why it didn't yesterday.

Anyway, thank you very much for you patience and guidance!

>
>
> On Dec 29, 2011, at 8:40 AM, Olivier wrote:
>
>> OK !
>> But AEL2's ifTime keyword do not use it, does it ?
>>
>> 2011/12/28, Mindaugas Jasiulis :
>>> Hi,
>>>
>>> This function sets TESTTIME global variable and if TESTTIME variable is
>>> set,
>>> then GoToIfTime use time from this variable.
>>>
>>>
>>> On 2011.12.28, at 17:28, Olivier  wrote:
>>>
 Hi,

 Thanks for replying.

 I'm afraid this :
 [foobar]
 exten => 123,1,Verbose(0,Into context ${CONTEXT})
 exten => 123,n,Verbose(0,Time is ${STRFTIME()})
 exten => 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
 exten => 123,n,Verbose(0,Time is ${STRFTIME()})
 exten => 123,n,HangUp()

 ... gives this:
   -- Executing [123@foobar:1] Verbose("SIP/7005-006b", "0,Into
 context foobar") in new stack
 Into context foobar
   -- Executing [123@foobar:2] Verbose("SIP/7005-006b", "0,Time
 is Wed Dec 28 16:25:59 2011") in new stack
 Time is Wed Dec 28 16:25:59 2011
   -- Executing [123@foobar:3] Set("SIP/7005-006b",
 "TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius") in new stack
   -- Executing [123@foobar:4] Verbose("SIP/7005-006b", "0,Time
 is Wed Dec 28 16:25:59 2011") in new stack
 Time is Wed Dec 28 16:25:59 2011
   -- Executing [123@foobar:5] Hangup("SIP/7005-006b", "") in new
 stack


 Do you see the same behaviour ?


 2011/12/28, Mindaugas Jasiulis :
> Hi,
>
> I do not know, whether this is the best way to use TESTTIME function,
> but
> for me it is working in that way:
> exten => 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
> OR
> You can use this:
> Set(__TESTTIME=${STRPTIME(2011-12-25 18:00:00,Europe/Vilnius,%Y-%m-%d
> %H:%M:%S)})
>
> Best regards,
> Mindaugas
>
>
>> Hi,
>>
>> Has someone a dialplan example using TESTTIME function (see core show
>> function TESTTIME) ?
>> I'm only getting replies such as "Function TESTTIME cannot be read".
>>
>> Regards
>
> --
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
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Re: [asterisk-users] DTMF Testing software to test IVR system

2011-12-29 Thread virendra bhati

In server B if I use SendDTMF then it means I am changing programming at
server B. Actually I don't have right or permission to change programming
in server B.

otherwise your suggestion is best for channel base communication.



On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind  wrote:

> Easy, use Read() to capture the incoming DTMF from Server-B
>
> Server-A <> Server-B
> Initiate-Call -> AnswerCall()
> SendDTMF(5)--> Read()
> Read()<-SendDTMF(4)
> SendDTMF(3)--> Read()
> Read()<-SendDTMF(2)
> SendDTMF(1)--> Read()
>
>
> Put proper GOTOIFs after reads if you like.
>
> --
> Regards,
> Sammy
>
> On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati wrote:
>
>> I originate calls from .call file and 1 channel I have at A server A and
>> another channel at B server.
>>
>> *A server code is below:-*
>>
>> exten => 43689956,1,Answer()
>> same => n,Wait(5)
>> same => n,SendDTMF(1)
>> same => n,NoOp(==   ${CHANNEL(state)}==> state)
>> same => n,wait(2)
>> same => n,SendDTMF(123456789012345#)
>> same => n,NoOp(==   ${CHANNEL(state)}==> state)
>> same => n,Hangup()
>>
>>  _  _
>> |  A server  |  ___DTMF Send_=> | B server   |
>> |_|  <=--- Responce -   |_|
>>
>> *B server code is below:-*
>> At B server call come to 201 extension which is mention here..
>>
>> exten => _20[1-6],1,Answer()
>> same => n,Ringing()
>> same => n,wait(2)
>> same => n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
>> AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
>> same => n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
>> $[${EXTEN}=205] ||
>> $[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
>> same => n,Hangup()
>>
>> Now I can send the DTMF from A to B. But How I will get the responce at
>> server A. I checked all the channels variable but they didn't reply status
>> of B server channel. All information I will get of server A. Main problem
>> is that control reach to AGI and then I don't have any rights to do any
>> update or modification on AGI. So if I can work on request and responce
>> then it will be the last solution as per my knowledge.
>>
>> Is this possible with the dialplan or I am just westing time?
>>
>>
>>
>> On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger wrote:
>>
>>> On 11-12-28 03:25 AM, virendra bhati wrote:
>>>
 Hi list,

 Is there any way in asterisk by which I make a call from server and then
 dialplan(IVR system) gets DTMF from it. I mean to say that automatically
 DTMF is sended by channels as per user defined,

 I read there is an application sendDTMF but I don't know how we can
 used it?

 like A script make the call by using localdail, .call file or any
 method.
 And after landing the call we send dtmf to IVR system automatically as
 per
 my script..


 *extensions.conf:-*


 exten =>  1234,1,Answer()
  same =>  n,Read(value,**pleasePress1forSupportPress2fo**
 rHelp,1,,10)
  same =>  n,NoOp(${value})
  same =>  n,ExecIf($[${value}=1]?Goto(**suppot,1))
  same =>  n,ExecIf($[${value}=2]?Goto(**help,1))
  same =>  n,Hangup()

 exten=>  support,1,Answer()
  same =>  n,NoOp(you are at support section)
  same =>  n,Hangup()

 exten=>  help,1,Answer()
  same =>  n,NoOp(you are at help section)
  same =>  n,Hangup()

  We have DTMF based tests for the testsuite[1] that you could use.
>>>
>>> [1] 
>>> http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/
>>> --
>>> Paul Belanger
>>> Digium, Inc. | Software Developer
>>> twitter: pabelanger | IRC: pabelanger (Freenode)
>>> Check us out at: http://digium.com & http://asterisk.org
>>>
>>>
>>> --
>>> __**__**
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>  http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>  
>>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>>
>>
>>
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-8885268942
>> Software Engineer
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>

Re: [asterisk-users] Interesting attack tonight & fail2ban them

2011-12-29 Thread Mikhail Lischuk

 

Jeroen Eeuwes писал 29.12.2011 07:29: 

> Probably my
understanding is limited, but it seems to me that they
> have already
'access' to your Asterisk for them to be able to try to
> make outgoing
calls. Wouldn't it be better to make sure they get the
> "usual" errors
like "Registration from failed - no matching peer
> found"?
> 
> In
other words, how did they get this far in the first place?
> 
> Best
regards,
> Jeroen Eeuwes

Agreed. If you didn't get the "Failed to
authenticate on INVITE" (or whatever error should Asterisk log for not
authenticated user trying to place a call, I might be wrong here) - your
problem is way more serious. 

As I can advice you from my wast (despite
not always successfull) intruders fighting experience - banning by
useragent can help. I always dreamed of Asterisk to implement that, but
until then - if all your users are like "Linksys blablabla" or "eyeBeam
blablabla" and you see any other agent on the Asterisk log - just ban
it. Ofcourse, there are 2 limitations: 

1) If he doesnt register,
Asterisk wont show his useragent in log. And as for yor issue - neither
will it show IP. I think we might ask devs to correct that some day 

2)
if you dont have some standard for user sip devices and they use
whatever they want to, it wont help either 

-- 
With Best
Regards
Mikhail Lischuk

ITX Ukraine

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Re: [asterisk-users] DTMF Testing software to test IVR system

2011-12-29 Thread Sammy Govind

o in that case you need to observer the call flow in Server-B, i.e what is
the length of sound file playing. what DTMF it requires etc etc and once
you detect the call flow for a successful IVR traversal then mimic the
behaviour of the call from Server-A.
Thats all you can do.
Think of it exactly the same as Answering Machine Detection Algorithm, but
in your case its like Server-B Detection Algorithm :)

--
Regards,
Sammy

On Thu, Dec 29, 2011 at 2:15 PM, virendra bhati  wrote:

> In server B if I use SendDTMF then it means I am changing programming at
> server B. Actually I don't have right or permission to change programming
> in server B.
>
> otherwise your suggestion is best for channel base communication.
>
>
>
>
> On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind  wrote:
>
>> Easy, use Read() to capture the incoming DTMF from Server-B
>>
>> Server-A <> Server-B
>> Initiate-Call -> AnswerCall()
>> SendDTMF(5)--> Read()
>> Read()<-SendDTMF(4)
>> SendDTMF(3)--> Read()
>> Read()<-SendDTMF(2)
>> SendDTMF(1)--> Read()
>>
>>
>> Put proper GOTOIFs after reads if you like.
>>
>> --
>> Regards,
>> Sammy
>>
>> On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati wrote:
>>
>>> I originate calls from .call file and 1 channel I have at A server A and
>>> another channel at B server.
>>>
>>> *A server code is below:-*
>>>
>>> exten => 43689956,1,Answer()
>>> same => n,Wait(5)
>>> same => n,SendDTMF(1)
>>> same => n,NoOp(==   ${CHANNEL(state)}==> state)
>>> same => n,wait(2)
>>> same => n,SendDTMF(123456789012345#)
>>> same => n,NoOp(==   ${CHANNEL(state)}==> state)
>>> same => n,Hangup()
>>>
>>>  _  _
>>> |  A server  |  ___DTMF Send_=> | B server   |
>>> |_|  <=--- Responce -   |_|
>>>
>>> *B server code is below:-*
>>> At B server call come to 201 extension which is mention here..
>>>
>>> exten => _20[1-6],1,Answer()
>>> same => n,Ringing()
>>> same => n,wait(2)
>>> same => n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
>>> AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
>>> same => n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
>>> $[${EXTEN}=205] ||
>>> $[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
>>> same => n,Hangup()
>>>
>>> Now I can send the DTMF from A to B. But How I will get the responce at
>>> server A. I checked all the channels variable but they didn't reply status
>>> of B server channel. All information I will get of server A. Main problem
>>> is that control reach to AGI and then I don't have any rights to do any
>>> update or modification on AGI. So if I can work on request and responce
>>> then it will be the last solution as per my knowledge.
>>>
>>> Is this possible with the dialplan or I am just westing time?
>>>
>>>
>>>
>>> On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger 
>>> wrote:
>>>
 On 11-12-28 03:25 AM, virendra bhati wrote:

> Hi list,
>
> Is there any way in asterisk by which I make a call from server and
> then
> dialplan(IVR system) gets DTMF from it. I mean to say that
> automatically
> DTMF is sended by channels as per user defined,
>
> I read there is an application sendDTMF but I don't know how we can
> used it?
>
> like A script make the call by using localdail, .call file or any
> method.
> And after landing the call we send dtmf to IVR system automatically as
> per
> my script..
>
>
> *extensions.conf:-*
>
>
> exten =>  1234,1,Answer()
>  same =>  n,Read(value,**pleasePress1forSupportPress2fo**
> rHelp,1,,10)
>  same =>  n,NoOp(${value})
>  same =>  n,ExecIf($[${value}=1]?Goto(**suppot,1))
>  same =>  n,ExecIf($[${value}=2]?Goto(**help,1))
>  same =>  n,Hangup()
>
> exten=>  support,1,Answer()
>  same =>  n,NoOp(you are at support section)
>  same =>  n,Hangup()
>
> exten=>  help,1,Answer()
>  same =>  n,NoOp(you are at help section)
>  same =>  n,Hangup()
>
>  We have DTMF based tests for the testsuite[1] that you could use.

 [1] 
 http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/
 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com & http://asterisk.org


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[asterisk-users] Help_In Voicemail , vedio play but voice is not here out.

2011-12-29 Thread Durgesh Mishra



Hi all, 

I am using to Xlite to save video voice mail. 

when i retreive it, then only video show , no voice is here out. 

Plz tell me where ,i am wrong , and how i can able to see video plus here audio 
in voice mail box. 

I did following configuration 

In Sip.conf 



videosupport=yes 



[phone1] 
type=friend 
host=dynamic 
context= employees 
mailbox=101@default 
callerid="phone1<101>" 
disallow=all 
allow=ilbc 
allow=g729 
allow=gsm 
allow=g723 
allow=ulaw 
allow=alaw 
allow=adpcm 
allow=h263p 
allow=h264 
allow=h263 

[phone2] 
 type=friend 
host=dynamic 
context= employees 
mailbox=102@default 
callerid="phone2<102>" 
disallow=all 
allow=ilbc 
allow=g723 
allow=gsm 
allow=g723 
allow=ulaw 
allow=alaw 
allow=adpcm 
allow=h263 
allow=h263p 
allow=h261 





In extension.conf 





exten => 102,1,VoiceMail( 102@default,u ) 

exten => 102,n,Hangup() 

exten => 704,1,VoiceMailMain() 



In voicemail.conf 

format=h263|alaw|gsm 

101 =>1234,phone1 
102 =>1234,phone2 



I call to 102 and save video voice mail 



check in 102/INBOX 

    267 Dec 29 13:22 msg0020.txt 
   61503 Dec 29 13:22 msg0020.h263 

    330 Dec 29 13:22 msg0020.gsm 
  1600 Dec 29 13:22 msg0020.alaw 



My Xlite phone codecs 

aLaw,uLaw,H.263 





Thanks&Regards 

Durgesh Mishra 



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Re: [asterisk-users] Question on hung channel

2011-12-29 Thread Jerry Geis


On 12/28/2011 03:57 PM, Jerry Geis wrote:

I ran into a "rare" situation today.

A really short message is being played over the ALSA or console 
channel from one asterisk box to another. Both running 1.4.30.
the incoming context on the ALSA or Console port box first runs an AGI 
before connecting the audio path.

The AGI got hung up for an non-asterisk issue. then finally continued.
The audio from the server was DONE, the server already hung up the call.
So the console port is just setting there with nothing to do.
the next call in says  - opps the port is busy so just give busy signal.

The original call never "frees" up and goes away.

How do I get the orphaned call to just close and go away?

Thanks,

Jerry

I have updated to 1.4.43 and I have the same situation.

Any thoughts?

Jerry

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Re: [asterisk-users] 1.6 and 1.8

2011-12-29 Thread Ryan Wagoner

On Thu, Dec 29, 2011 at 12:05 AM, Bruce B  wrote:

> I have been running 1.8.7 with a few fixes back ported from the 1.8.8
>> release candidate for the last 2.5 months. The system processes around
>> 4,000 calls per day over PRIs for 250 Polycom phones.
>>
>> Previously I was running 1.6.1.18 with a bunch of back ports for fixes
>> and features. Overall it was stable but every few months I had an issue
>> where a channel would get hung. When this happened core show channels would
>> crash the console and I would eventually have to restart Asterisk.
>>
>> Ryan
>>
>
> What od you mean by, "been running 1.8.7 with a few fixes back ported
> from the 1.8.8 release candidate". So, this is a version 1.8.7 release that
> you are using or a 1.8.8 or is this a mix of both that you come up with?
> Can you please be specific with fixes?
>
> Thanks
>
>
It was a mix I came up with as I was hitting a few bugs in 1.8.7 and 1.8.8
wasn't released. At this point I would just go for 1.8.8. The issue was
mainly 17541 which was filling my logs and basically made Asterisk unusable.

https://issues.asterisk.org/jira/browse/ASTERISK-17541
https://issues.asterisk.org/jira/browse/ASTERISK-18570
https://issues.asterisk.org/jira/browse/ASTERISK-18101

I had tested 1.8.4 before and was hit by a bunch of dtmf issues that were
fixed in 1.8.5. When 1.8.7 came out it looked fairly stable so I switched
from 1.6.1.18. I was running the 1.6.1 branch as I needed TCP SIP support.
Right now I have been testing 1.8.8 which looks to be a good release. The
1.8 series has come a long way in a few releases as far as fixing major
bugs.

Ryan
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Re: [asterisk-users] Asterisk fail2ban filters - show us yours

2011-12-29 Thread Diego Aguirre (DagMoller)

Hi,

I Have added this line for asterisk 1.8 (i have allowguest=yes and 
context=default in sip.conf):
NOTICE.* .*: Call from '.*' () to extension '.*' rejected because 
extension not found in context 'default'.

Em 29-12-2011 13:03, Patrick Lists escreveu:
> Hi,
> 
> In the thread "Interesting attack tonight & fail2ban them" Bruce B mentioned 
> it would be nice to have input from the Community to come up with the best 
> set of fail2ban filters. That's a great idea. So let's start with Bruce's 
> filters (thanks!) and take it from there. Anyone have any improvements and/or 
> additions? Apologies for the line wrap. No idea how to prevent that in 
> Thunderbird. The filters are also at http://pastebin.com/6T9M1W3F
> 
> Not sure but it may be possible that logging has changed between Asterisk 
> 1.4, 1.6, 1.8 and 10 so please mention the asterisk version with your filters.
> 
> For Asterisk 1.8:
> 
> failregex = Registration from '.*' failed for '(:[0-9]{1,5})?' - Wrong 
> password
> Registration from '.*' failed for '(:[0-9]{1,5})?' - No 
> matching peer found
> Registration from '.*' failed for '(:[0-9]{1,5})?' - Device 
> does not match ACL
> Registration from '.*' failed for '(:[0-9]{1,5})?' - 
> Username/auth name mismatch
> Registration from '.*' failed for '(:[0-9]{1,5})?' - Peer 
> is not supposed to register
> NOTICE.*  failed to authenticate as '.*'$
> NOTICE.* .*: No registration for peer '.*' (from )
> NOTICE.* .*: Host  failed MD5 authentication for '.*' (.*)
> VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-noservice' 
> (language '.*')
> 
> 
> There are 2 lines that I have which are not in this list:
> 
> NOTICE.* .*: Registration from '.*' failed for '' - ACL error 
> (permit/deny)
> NOTICE.* .*: Failed to authenticate user .*@.*
> 
> How about those (no idea for which Asterisk version they are)?
> 
> Regards,
> Patrick
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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-- 
Diego Aguirre (DagMoller)
Infodag Consultoria
FWD#: 459696
Enum#: +55 21 8871-4916 (e164.org)
DUNDi-br#: 21 8871-4916

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Re: [asterisk-users] IAX2 woes

2011-12-29 Thread Carlos Rojas

Hello

Asterisk only says that the iax2 channel don't work maybe you look the
iax.conf. you trunk. Is iax I think

Regards
On Dec 29, 2011 6:49 AM, "--[ UxBoD ]--"  wrote:

> Hello all,
>
> I attempted to make a couple of outbound calls this morning and always got
> the busy tone.  I checked the Asterisk console and was greeted with:
>
> [Dec 29 11:29:22] WARNING[12039]: app_dial.c:2218 dial_exec_full: Unable
> to create channel of type 'IAX2' (cause 20 - Unknown)
>   == Everyone is busy/congested at this time (1:0/0/1)
>
> I proceeded to restart Asterisk and dialed the same number again and it
> worked without fault. What could cause this type of error and is there any
> way to auto-remediate when it does arise ?
>
> voip*CLI> core show version
> Asterisk 10.0.0 built by root @ voip.my.server on a x86_64 running Linux
> on 2011-12-19 16:16:46 UTC
> --
> Thanks, Phil
>
>
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Re: [asterisk-users] 1.6 and 1.8

2011-12-29 Thread Bruce B

Log are being filled with g729 transcoding error in 1.8.7x now :-(
I don't dare to test 1.8.8x as it might have something else broken.
Unfortunately I can no longer trust the release candidates. Thanks for the
input.

On Thu, Dec 29, 2011 at 8:29 AM, Ryan Wagoner  wrote:

> On Thu, Dec 29, 2011 at 12:05 AM, Bruce B  wrote:
>
>> I have been running 1.8.7 with a few fixes back ported from the 1.8.8
>>> release candidate for the last 2.5 months. The system processes around
>>> 4,000 calls per day over PRIs for 250 Polycom phones.
>>>
>>> Previously I was running 1.6.1.18 with a bunch of back ports for fixes
>>> and features. Overall it was stable but every few months I had an issue
>>> where a channel would get hung. When this happened core show channels would
>>> crash the console and I would eventually have to restart Asterisk.
>>>
>>> Ryan
>>>
>>
>> What od you mean by, "been running 1.8.7 with a few fixes back ported
>> from the 1.8.8 release candidate". So, this is a version 1.8.7 release that
>> you are using or a 1.8.8 or is this a mix of both that you come up with?
>> Can you please be specific with fixes?
>>
>> Thanks
>>
>>
> It was a mix I came up with as I was hitting a few bugs in 1.8.7 and 1.8.8
> wasn't released. At this point I would just go for 1.8.8. The issue was
> mainly 17541 which was filling my logs and basically made Asterisk unusable.
>
> https://issues.asterisk.org/jira/browse/ASTERISK-17541
> https://issues.asterisk.org/jira/browse/ASTERISK-18570
> https://issues.asterisk.org/jira/browse/ASTERISK-18101
>
> I had tested 1.8.4 before and was hit by a bunch of dtmf issues that were
> fixed in 1.8.5. When 1.8.7 came out it looked fairly stable so I switched
> from 1.6.1.18. I was running the 1.6.1 branch as I needed TCP SIP support.
> Right now I have been testing 1.8.8 which looks to be a good release. The
> 1.8 series has come a long way in a few releases as far as fixing major
> bugs.
>
> Ryan
>
>
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[asterisk-users] performance/memory

2011-12-29 Thread Matt Hamilton


I have a couple of performance/memory related questions:


Is there any downside to using long URIs as far as memory or database (mysql) 
performance is concerned, e.g.

sip:1234567890_1234567...@abc.com? Or is this negligible?


Also is there a performance hit if no pattern matching is used?  e.g.

exten => _XXX,Noop(...

vs

exten => 100,Noop(..
exten => 101,Noop(...
exten => 102,Noop(...
... 
exten => 999,Noop(...

If a call comes to 999, does Asterisk go through each extension sequentially 
from 100 to 999 until it finds the matching one?

Thanks,
Matt



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[asterisk-users] Softphones

2011-12-29 Thread Rebecca Robinson

 

We are currently using an older version of Eyebeam on our deployment and
keep having an issue with the disappearance of SIP accounts, and after
research found it is a bug on the version we currently have.

 

I am looking for a new softphone solution and I was wondering what
everyone was using out there with your Asterisk deployments.  Any
information would be helpful and most appreciated.

 

Our current user base is on Windows XP, but we would like the chosen
solution to be compatible with Windows 7 and MAC as well.   I have
looked into using Bria by Counterpath, but would like to keep my options
open. 

 



 

Rebecca Robinson

Telecom Administrator

 

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please reply to the above-referenced sender about the error and 
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[asterisk-users] Asterisk fail2ban filters - show us yours

2011-12-29 Thread Patrick Lists


Hi,

In the thread "Interesting attack tonight & fail2ban them" Bruce B 
mentioned it would be nice to have input from the Community to come up 
with the best set of fail2ban filters. That's a great idea. So let's 
start with Bruce's filters (thanks!) and take it from there. Anyone have 
any improvements and/or additions? Apologies for the line wrap. No idea 
how to prevent that in Thunderbird. The filters are also at 
http://pastebin.com/6T9M1W3F


Not sure but it may be possible that logging has changed between 
Asterisk 1.4, 1.6, 1.8 and 10 so please mention the asterisk version 
with your filters.


For Asterisk 1.8:

failregex = Registration from '.*' failed for '(:[0-9]{1,5})?' - 
Wrong password
Registration from '.*' failed for '(:[0-9]{1,5})?' - 
No matching peer found
Registration from '.*' failed for '(:[0-9]{1,5})?' - 
Device does not match ACL
Registration from '.*' failed for '(:[0-9]{1,5})?' - 
Username/auth name mismatch
Registration from '.*' failed for '(:[0-9]{1,5})?' - 
Peer is not supposed to register

NOTICE.*  failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from )
NOTICE.* .*: Host  failed MD5 authentication for '.*' 
(.*)
VERBOSE.* logger.c: -- .*IP/-.* Playing 
'ss-noservice' (language '.*')



There are 2 lines that I have which are not in this list:

NOTICE.* .*: Registration from '.*' failed for '' - ACL error 
(permit/deny)

NOTICE.* .*: Failed to authenticate user .*@.*

How about those (no idea for which Asterisk version they are)?

Regards,
Patrick

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Re: [asterisk-users] Interesting attack tonight & fail2ban them

2011-12-29 Thread Michelle Dupuis

1. I checked the log and I don't see any registration attempt, so I *assume* 
they simply send an invite, and so they are in the external/outside context of 
my dialplan.  So they are trying to reach extensions which don't exist.  If 
they succesfully registered they would be on the internal context, and their 
calls would have succeeded.  (Or am I missing something?).  I actually see 
nothing in the log but the notice (and nothing on the CLI but the notice)...so 
I assume it is only an invite?

2. I got their IP by turning on SIP DEBUG while they were attacking.

3. The NOTICE showed a call from '' - what normally goes there?  I can't 
reproduce this NOTICE so I'm not sure what causes it to be recorded.  Normal 
calls show "Accepting AUTHENTICATED call from x.x.x.x"

I'm thinking of using SIPCHANINFO and LOG to log the bad attempts, and let 
fail2ban takeover from there.

Thanks


From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Mikhail Lischuk 
[mlisc...@itx.com.ua]
Sent: Thursday, December 29, 2011 4:14 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Interesting attack tonight & fail2ban them


Jeroen Eeuwes писал 29.12.2011 07:29:



Probably my understanding is limited, but it seems to me that they
have already 'access' to your Asterisk for them to be able to try to
make outgoing calls. Wouldn't it be better to make sure they get the
"usual" errors like "Registration from failed - no matching peer
found"?

In other words, how did they get this far in the first place?

Best regards,
Jeroen Eeuwes



Agreed. If you didn't get the "Failed to authenticate on INVITE" (or whatever 
error should Asterisk log for not authenticated user trying to place a call, I 
might be wrong here) - your problem is way more serious.

As I can advice you from my wast (despite not always successfull) intruders 
fighting experience - banning by useragent can help. I always dreamed of 
Asterisk to implement that, but until then - if all your users are like 
"Linksys blablabla" or "eyeBeam blablabla" and you see any other agent on the 
Asterisk log - just ban it. Ofcourse, there are 2 limitations:

1) If he doesnt register, Asterisk wont show his useragent in log. And as for 
yor issue - neither will it show IP. I think we might ask devs to correct that 
some day

2) if you dont have some standard for user sip devices and they use whatever 
they want to, it wont help either

--
With Best Regards
Mikhail Lischuk

ITX Ukraine

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Re: [asterisk-users] 1.6 and 1.8

2011-12-29 Thread David Backeberg

On Wed, Dec 28, 2011 at 4:10 PM, Danny Nicholas  wrote:
> Can somebody point me to an explanation from Kevin or Tzafir or someone else
> "up the food chain" explaining the differences/benefits of 1.6/1.8 vs
> 1.4/10.0?

What's the difference between a car released in 2006 versus a car
released in 2012? Very complex question. Aside from reading the
changelogs, like others have suggested, you should understand that
asterisk is developed in a very public way, with open access to
source, bugs, and the ability of lots of people to provide patches.

Some feature requests are very difficult to honor without a major
rewrite of various layers. And the intervening releases made something
substantially better (well, some people will disagree; if you don't
need any of the newer features, you might be able to ram more calls
through an older version) or added more flexibility or more features.
If you are running 1.4 and want a feature that comes built-in with
1.6, nobody is going to have any sympathy for you when you say you
wish 1.4 did X feature.

Basically, if you want to participate in the public ecosystem where
you can find something you want improved, or you come across a bug,
you need to be at the stage of development where code changes are
still happening.

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Re: [asterisk-users] Interesting attack tonight & fail2ban them

2011-12-29 Thread Bruce B

Maybe your logger is not setup properly?! You should get the IP in logs. I
can't think of when you won't get the IP in your logs unless the SIP
packets are manipulated. That IP is from Voxel.net. You don't have a VPS or
service from them do you?

2011/12/29 Michelle Dupuis 

>  1. I checked the log and I don't see any registration attempt, so I
> *assume* they simply send an invite, and so they are in the
> external/outside context of my dialplan.  So they are trying to reach
> extensions which don't exist.  If they succesfully registered they would be
> on the internal context, and their calls would have succeeded.  (Or am I
> missing something?).  I actually see nothing in the log but the notice (and
> nothing on the CLI but the notice)...so I assume it is only an invite?
>
> 2. I got their IP by turning on SIP DEBUG while they were attacking.
>
> 3. The NOTICE showed a call from '' - what normally goes there?  I can't
> reproduce this NOTICE so I'm not sure what causes it to be recorded.
> Normal calls show "Accepting AUTHENTICATED call from x.x.x.x"
>
> I'm thinking of using SIPCHANINFO and LOG to log the bad attempts, and let
> fail2ban takeover from there.
>
> Thanks
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [
> asterisk-users-boun...@lists.digium.com] On Behalf Of Mikhail Lischuk [
> mlisc...@itx.com.ua]
> *Sent:* Thursday, December 29, 2011 4:14 AM
>
> *To:* Asterisk Users List
> *Subject:* Re: [asterisk-users] Interesting attack tonight & fail2ban them
>
>   Jeroen Eeuwes писал 29.12.2011 07:29:
>
>
>
> Probably my understanding is limited, but it seems to me that they
> have already 'access' to your Asterisk for them to be able to try to
> make outgoing calls. Wouldn't it be better to make sure they get the
> "usual" errors like "Registration from failed - no matching peer
> found"?
>
> In other words, how did they get this far in the first place?
>
> Best regards,
> Jeroen Eeuwes
>
>
>  Agreed. If you didn't get the "Failed to authenticate on INVITE" (or
> whatever error should Asterisk log for not authenticated user trying to
> place a call, I might be wrong here) - your problem is way more serious.
>
> As I can advice you from my wast (despite not always successfull)
> intruders fighting experience - banning by useragent can help. I always
> dreamed of Asterisk to implement that, but until then - if all your users
> are like "Linksys blablabla" or "eyeBeam blablabla" and you see any other
> agent on the Asterisk log - just ban it. Ofcourse, there are 2 limitations:
>
> 1) If he doesnt register, Asterisk wont show his useragent in log. And as
> for yor issue - neither will it show IP. I think we might ask devs to
> correct that some day
>
> 2) if you dont have some standard for user sip devices and they use
> whatever they want to, it wont help either
>
> --
> With Best Regards
> Mikhail Lischuk 
>
> ITX Ukraine
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] IAX2 woes

2011-12-29 Thread --[ UxBoD ]--

Hello all, 

I attempted to make a couple of outbound calls this morning and always got the 
busy tone. I checked the Asterisk console and was greeted with: 

[Dec 29 11:29:22] WARNING[12039]: app_dial.c:2218 dial_exec_full: Unable to 
create channel of type 'IAX2' (cause 20 - Unknown) 
== Everyone is busy/congested at this time (1:0/0/1) 

I proceeded to restart Asterisk and dialed the same number again and it worked 
without fault. What could cause this type of error and is there any way to 
auto-remediate when it does arise ? 

voip*CLI> core show version 
Asterisk 10.0.0 built by root @ voip.my.server on a x86_64 running Linux on 
2011-12-19 16:16:46 UTC 

-- 
Thanks, Phil 

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Re: [asterisk-users] Asterisk fail2ban filters - show us yours

2011-12-29 Thread Bruce B

>
> Hi,
>
> I Have added this line for asterisk 1.8 (i have allowguest=yes and
> context=default in sip.conf):
> NOTICE.* .*: Call from '.*' () to extension '.*' rejected because
> extension not found in context 'default'.
>
> Em 29-12-2011 13:03, Patrick Lists escreveu:
> > Hi,
> >
> > In the thread "Interesting attack tonight & fail2ban them" Bruce B
> mentioned it would be nice to have input from the Community to come up with
> the best set of fail2ban filters. That's a great idea. So let's start with
> Bruce's filters (thanks!) and take it from there. Anyone have any
> improvements and/or additions? Apologies for the line wrap. No idea how to
> prevent that in Thunderbird. The filters are also at
> http://pastebin.com/6T9M1W3F
> >
> > Not sure but it may be possible that logging has changed between
> Asterisk 1.4, 1.6, 1.8 and 10 so please mention the asterisk version with
> your filters.
> >
> > For Asterisk 1.8:
> >
> > failregex = Registration from '.*' failed for '(:[0-9]{1,5})?' -
> Wrong password
> > Registration from '.*' failed for '(:[0-9]{1,5})?' -
> No matching peer found
> > Registration from '.*' failed for '(:[0-9]{1,5})?' -
> Device does not match ACL
> > Registration from '.*' failed for '(:[0-9]{1,5})?' -
> Username/auth name mismatch
> > Registration from '.*' failed for '(:[0-9]{1,5})?' -
> Peer is not supposed to register
> > NOTICE.*  failed to authenticate as '.*'$
> > NOTICE.* .*: No registration for peer '.*' (from )
> > NOTICE.* .*: Host  failed MD5 authentication for '.*'
> (.*)
> > VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-noservice'
> (language '.*')
> >
> >
> > There are 2 lines that I have which are not in this list:
> >
> > NOTICE.* .*: Registration from '.*' failed for '' - ACL error
> (permit/deny)
> > NOTICE.* .*: Failed to authenticate user .*@.*
> >
> > How about those (no idea for which Asterisk version they are)?
> >
> > Regards,
> > Patrick
>

Thanks Patrick. This is a great initiative. Let's all build the strongest
and most detailed filter possible. I actually looked at mine and now see
that it has weaknesses due Asterisk 1.8.8x giving different type of logs or
maybe FreePBX. Let's test, fix and append to the end of the filter.
Everyone is welcome to contribute.

So far we have:

*For Asterisk 1.8:*
failregex = Registration from '.*' failed for '(:[0-9]{1,5})?' -
Wrong password
   Registration from '.*' failed for '(:[0-9]{1,5})?' - No
matching peer found
   Registration from '.*' failed for '(:[0-9]{1,5})?' -
Device does not match ACL
   Registration from '.*' failed for '(:[0-9]{1,5})?' -
Username/auth name mismatch
   Registration from '.*' failed for '(:[0-9]{1,5})?' - Peer
is not supposed to register
   NOTICE.*  failed to authenticate as '.*'$
   NOTICE.* .*: No registration for peer '.*' (from )
   NOTICE.* .*: Host  failed MD5 authentication for '.*' (.*)
   VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-noservice'
(language '.*') *#Outdated?*
  #*Situation:* allowguest=yes and context=default in sip.con - *Tested
by **Diego Aguirre?*
   NOTICE.* .*: Call from '.*' () to extension '.*' rejected
because extension not found in context 'default'

The following are what I found to be insecure but need escaping and fine
tuning to work with filter:

*Asterisk 1.8 + FreePBX:*
*Situation:* When target is coming in from unknown DID -
Needs character escaping
Executing [unknown@from-sip-external:1] NoOp("SIP/10.0.0.6-0001",
"Received incoming SIP connection from unknown peer to unknown") in new
stack

*Situation:* Same as above except for an extension is called. Above was
just IP call. Extension 011x doesn't exist.
Executing [011566@from-sip-external:1] NoOp("SIP/10.0.0.6-0003",
"Received incoming SIP connection from unknown peer to 011566") in new
stack

*Situation: *Same as above except for extension 101 does exist but system
still rejects calls due to no guest allowed?!
Executing [101@from-sip-external:1] NoOp("SIP/10.0.0.6-0005", "Received
incoming SIP connection from unknown peer to 101") in new stack

*All of above have this following which can be used as a universal
filter: *Executing
[s@from-sip-external:8] Playback("SIP/10.0.0.6-0005", "ss-noservice")
in new stack *
*
*
***Notice how this ss-noservice is difference from current the outdated
filter one:
*VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-noservice' (language
'.*')*

-Bruce
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[asterisk-users] How to create SIP INVITE with different To: Header field than Request-Line URI

2011-12-29 Thread Michael Shore

Hello,

 

I'm trying to interface Asterisk with a third party voicemail system. This
voicemail system registers itself as extension 199. This voicemail system
gets the DID number (mailbox) from the SIP To: header field.

 

My problem is creating the SIP INVITE with a  To: field that's different
that the Request-Line URI

 

I need to create an INVITE that looks like this in Asterisk:

 

20:23:12 UDP Packet Received from 127.0.0.1:5060
<<<

INVITE sip:199@127.0.0.1:5070;target=101%40voicemailserver;cause=486 SIP/2.0

Via: SIP/2.0/UDP 10.11.11.205:5060;rport;branch=z9hG4bK253292

To: 

From: "101";tag=3277

Call-ID: 1325031862-3292-VOICEMAILSER@10.11.11.205

CSeq: 14 INVITE

 

Can this be accomplished?

 

Mike

 

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[asterisk-users] can't set up tcp sip - sip connection : digest problem

2011-12-29 Thread sean darcy

Trying to set up a simple sip - sip connection over tcp. Home : 10.0.0 - 
Office: 1.8.8.0


Home sip.conf:

register => 
tcp://office-going-to-home:password@/home-coming-from-office


[home-coming-from-office] ; receives  calls
type=friend
transport=tcp
dtmfmode=rfc2833
disallow=all
allow=ulaw
secret=password
context=incoming

Office:
[office-going-to-home] ;  places calls
type=peer ;; we only call out
transport=tcp
dtmfmode=rfc2833
disallow=all
allow=ulaw
fromuser=home-coming-from-office
remotesecret=password

sip show peer office-going-to-home


  * Name   : office-going-to-home
  Secret   : 
  MD5Secret: 
  Remote Secret: 

  Status   : OK (28 ms)
  Useragent: Asterisk PBX 10.0.0
  Reg. Contact : sip:home-coming-from-office@ipaddr>:5060;transport=TCP


But if I place a call from the office to home, on home cli:

[Dec 29 16:32:23] WARNING[4387]: chan_sip.c:14446 check_auth: username 
mismatch, have , digest has 
[Dec 29 16:32:23] NOTICE[4387]: chan_sip.c:22769 handle_request_invite: 
Failed to authenticate device "Office" 
>;tag=as150d4960


I thought the last part of the register line was supposed to tell the 
calling end point what to put in the digest header.


sean


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Re: [asterisk-users] 1.6 and 1.8

2011-12-29 Thread Ryan Wagoner

On Thu, Dec 29, 2011 at 12:18 PM, Bruce B  wrote:

> Log are being filled with g729 transcoding error in 1.8.7x now :-(
> I don't dare to test 1.8.8x as it might have something else broken.
> Unfortunately I can no longer trust the release candidates. Thanks for the
> input.
>
>
What are you using for transcoding? I'm running 1.8.7 with a Sangoma
transcoding card. I would give 1.8.8 a try as they fixed the transcoding
issue in 1.8.7 or at least try the patch I mentioned before.

Ryan
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[asterisk-users] Asterisk Registrar / Trunk

2011-12-29 Thread Khaled W. Chehab

Dears, 

 

1-I have a GSM gateway (GOIP) with 8 ports, I used to let every port
register to VoIPSwitch in order to know how many minutes  does this GSM
card, ASR ,ACD on each card.

It's too simple on VoIPSwitch to add the registrar client to dial plan ,but
in asterisk only I can find trunks

How can I do that with asterisk .

 

 

2-Do any one know from where I can download a2billing prompts in Arabic for
free.

 

Regards

 

Khaled  Chehab

   NGN Eng.

 

Description: xplorium 

 Operations Office - Lebanon

 Office : +961 1 868686 ext 115

 Mobile: +961 3 045212

 E-mail:   kche...@xplorium.com

 MSN ID :khalidche...@hotmail.com  

 Web Site: http://www.xplorium.com

 

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[asterisk-users] Block Specific Number on Inbound

2011-12-29 Thread Kevin Oravits

Greetings,

Is there a way to block a specific inbound number? I've found code online for 
blocking all nocallerid and all 800, etc. but nothing for a specific number. My 
company is wanting me to block a specific number. Is this possible in Asterisk 
1.4 and 1.6 or do I need to go through my Service Provider?

Thanks,

Kevin Oravits
Phone Sys Admin

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Re: [asterisk-users] Block Specific Number on Inbound

2011-12-29 Thread Robert Huddleston

Take a look at Blacklist

 

I love that command and love to send nice intercept messages to the other
side J

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Thursday, December 29, 2011 8:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Block Specific Number on Inbound

 

Greetings,

 

Is there a way to block a specific inbound number? I've found code online
for blocking all nocallerid and all 800, etc. but nothing for a specific
number. My company is wanting me to block a specific number. Is this
possible in Asterisk 1.4 and 1.6 or do I need to go through my Service
Provider?

 

Thanks,

 

Kevin Oravits  

Phone Sys Admin

 

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Re: [asterisk-users] Block Specific Number on Inbound

2011-12-29 Thread Stuart Sheldon

Check out the X Boy/Girl friend feature.

http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf

Around the middle of the page.

Stu


-Original Message-
From: Kevin Oravits 
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion

To: asterisk-users@lists.digium.com 
Subject: [asterisk-users] Block Specific Number on Inbound
Date: Fri, 30 Dec 2011 01:39:46 +

Greetings,

 

Is there a way to block a specific inbound number? I’ve found code
online for blocking all nocallerid and all 800, etc. but nothing for a
specific number. My company is wanting me to block a specific number. Is
this possible in Asterisk 1.4 and 1.6 or do I need to go through my
Service Provider?

 

Thanks,

 

Kevin Oravits 

Phone Sys Admin

 


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[asterisk-users] Asterisk Video Playback - MP4 ad 3GP files

2011-12-29 Thread LL


Hi There,
I'm fairly new to asterisk and I'm trying to play a video file during a 
video call without success for a couple of days now.
I've posted a question at stack-overflow describing my problem - 
http://stackoverflow.com/questions/8675713/asterisk-video-playback-mp4-3gp

*
In short, what are the exact specs that a 3gp or mp4 file needs to have 
in order to play under asterisk ?*

Any tips, links or suggestions will be very welcome.

*Thank you all!
LL
*

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Re: [asterisk-users] Interesting attack tonight & fail2ban them

2011-12-29 Thread Jeroen Eeuwes

Hi Michelle,

> 1. I checked the log and I don't see any registration attempt, so I *assume*
> they simply send an invite, and so they are in the external/outside context
> of my dialplan.  So they are trying to reach extensions which don't exist.
> If they succesfully registered they would be on the internal context, and
> their calls would have succeeded.  (Or am I missing something?).  I actually
> see nothing in the log but the notice (and nothing on the CLI but the
> notice)...so I assume it is only an invite?

Are you saying that you have an "external/outside" context in which
you have "allowguest=yes"?

Which version of Asterisk are you running?

Best regards,
Jeroen Eeuwes

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Re: [asterisk-users] DTMF Testing software to test IVR system

Re: [asterisk-users] Function TESTTIME example [SOLVED]

Re: [asterisk-users] DTMF Testing software to test IVR system

Re: [asterisk-users] Interesting attack tonight & fail2ban them

Re: [asterisk-users] DTMF Testing software to test IVR system

[asterisk-users] Help_In Voicemail , vedio play but voice is not here out.

Re: [asterisk-users] Question on hung channel

Re: [asterisk-users] 1.6 and 1.8

Re: [asterisk-users] Asterisk fail2ban filters - show us yours

Re: [asterisk-users] IAX2 woes

Re: [asterisk-users] 1.6 and 1.8

[asterisk-users] performance/memory

[asterisk-users] Softphones

[asterisk-users] Asterisk fail2ban filters - show us yours

Re: [asterisk-users] Interesting attack tonight & fail2ban them

Re: [asterisk-users] 1.6 and 1.8

Re: [asterisk-users] Interesting attack tonight & fail2ban them

[asterisk-users] IAX2 woes

Re: [asterisk-users] Asterisk fail2ban filters - show us yours

[asterisk-users] How to create SIP INVITE with different To: Header field than Request-Line URI

[asterisk-users] can't set up tcp sip - sip connection : digest problem

Re: [asterisk-users] 1.6 and 1.8

[asterisk-users] Asterisk Registrar / Trunk

[asterisk-users] Block Specific Number on Inbound

Re: [asterisk-users] Block Specific Number on Inbound

Re: [asterisk-users] Block Specific Number on Inbound

[asterisk-users] Asterisk Video Playback - MP4 ad 3GP files

Re: [asterisk-users] Interesting attack tonight & fail2ban them

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