Re: [asterisk-users] Are there AMI commands to manipulate a voice mailbox?

[Andrew Furey]

On 14/05/2010, Motiejus Jakštys  wrote:
> Talking about file permissions, on Linux everything is possible using
>  POSIX ACLs. You can set specific rights to files/directories for
>  certain users.
>  Note 1: if setting "group" permissions is enough, use that.
>  Note 2: Asterisk and web server should be on separate machines (at
>  least virtual machines) for many reasons... I would mount my voicemail
>  over NFS... which itself has enough access control.
>  Note 3: if you decide to experiment with ACLs (IMHO, most flexible) -
>  do not forget to remout your file system:
>  mount -o remount,acl /usr

Not quite everything - you're still limited to read/write/execute
granularity (unless something has changed in the 5 years since I
experimented with it). If you're expecting Full Control / Modify /
Delete etc as per Windows 2000 and its ilk, you might have to look at
something else...

Andrew

-- 
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Re: [asterisk-users] Are there AMI commands to manipulate a voice mailbox?

[Motiejus Jakštys]

Talking about file permissions, on Linux everything is possible using
POSIX ACLs. You can set specific rights to files/directories for
certain users.
Note 1: if setting "group" permissions is enough, use that.
Note 2: Asterisk and web server should be on separate machines (at
least virtual machines) for many reasons... I would mount my voicemail
over NFS... which itself has enough access control.
Note 3: if you decide to experiment with ACLs (IMHO, most flexible) -
do not forget to remout your file system:
mount -o remount,acl /usr

Hope it helps

On Fri, May 14, 2010 at 4:49 AM, Jim Dickenson  wrote:
> You might be able to use local channels to do what you want.
>
> As for the user asterisk runs as and the user the web server run as you can 
> maybe have both users belong to the same secondary group and gain the access 
> you need that way.
>
> Partly depends on what exactly you are wanting to do.
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com
>
> CfMC
> http://www.cfmc.com/
>
>
>
> On May 13, 2010, at 5:09 PM, Carlos Chavez wrote:
>
>>       I want to make a web interface so my users can listen/erase voicemails.
>> Is there a way to do this from the Asterisk manager interface?  Since
>> Asterisk and the web server do not run as the same user I cannot do a
>> direct manipulation of the voicemail files
>> in /var/spool/asterisk/voicemail.  Maybe there are some AMI commands to
>> delete a specific voicemail from a mailbox?  I have not found any so far
>> but documentation is often behind of implementation.
>>
>>       Any ideas on how to approach this?
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez Prats
>> Director de Tecnología
>> +52-55-91169161 ext 2001
>> --
>> _
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] Do you think my server is being attacked?

[Steve Edwards]

On Fri, 14 May 2010, bruce bruce wrote:

> Are these indications of attacks on this system? I specifically have 
> port 22 disabled at all times and only port forward it to server when I 
> access SSH for a minute or so. Shouldn't UNKNOWN be an actual IP 
> address?
> 
> /var/log/secure:
> 
> May 14 00:35:39 pbx sshd[9011]: Did not receive identification string from 
> UNKNOWN
> May 14 00:36:09 pbx sshd[9040]: Did not receive identification string from 
> UNKNOWN

Not an Asterisk question.

If you had asked Google, you may have seen (second result from my Google 
query) a post ("http://www.freepbx.org/v2/ticket/3461";) about FreePBX 
scanning port 22 every thirty seconds.

Maybe this is what you are seeing.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] aastra pt 480e phone

[bruce bruce]

Unplugging just turns off the phone and has no effect on the settings. You
can not "damage" the phone by tampering configurations but you can mess up
the settings and it might not register, send, or receive calls.

User manu for your reference:

http://www.aastra.com/cps/rde/xbcr/SID-3D8CCB6A-E9BBA0B7/04/480e_ma_en_0306.pdf

Overall instructions:
http://www.aastra.com/cps/rde/xchg/SID-3D8CCB73-FE677A10/04/hs.xsl/19493.htm#dl_instructions

I
couldn't find how to access Web Configurator but I guess it would be the IP
address. And if username/password on this is the same as the newer phones
from Aastra then it would be admin/2

-Bruce

On Fri, May 14, 2010 at 12:38 AM, michael capelle <
michael.cape...@charter.net> wrote:

> hello
> i hope i am posting to the right list, i am a totally blind user, and i
> want
> to reprogram my aastra pt 480e phone, my friend used the web configurator,
> but i think he programmed thw wrong codes, a few questions, is it possible
> to damage the phone by programming it wrong? also, how does one reset it
> literally to factory defaults? as unplugging it and replugging it back into
> the wall didn't work.
> regards
> a happy aastra user.
>
>
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[asterisk-users] Do you think my server is being attacked?

[bruce bruce]

Hello Everyone,

Are these indications of attacks on this system? I specifically have port 22
disabled at all times and only port forward it to server when I access SSH
for a minute or so. Shouldn't UNKNOWN be an actual IP address?

*/var/log/secure:*

May 14 00:35:39 pbx sshd[9011]: Did not receive identification string from
UNKNOWN
May 14 00:36:09 pbx sshd[9040]: Did not receive identification string from
UNKNOWN
May 14 00:36:39 pbx sshd[9075]: Did not receive identification string from
UNKNOWN
May 14 00:37:10 pbx sshd[9102]: Did not receive identification string from
UNKNOWN
May 14 00:37:40 pbx sshd[9139]: Did not receive identification string from
UNKNOWN
May 14 00:38:11 pbx sshd[9166]: Did not receive identification string from
UNKNOWN
May 14 00:38:41 pbx sshd[9195]: Did not receive identification string from
UNKNOWN
May 14 00:39:11 pbx sshd[9230]: Did not receive identification string from
UNKNOWN
May 14 00:39:42 pbx sshd[9250]: Did not receive identification string from
UNKNOWN
May 14 00:40:13 pbx sshd[9294]: Did not receive identification string from
UNKNOWN
May 14 00:40:44 pbx sshd[9329]: Did not receive identification string from
UNKNOWN
May 14 00:41:14 pbx sshd[9366]: Did not receive identification string from
UNKNOWN
May 14 00:41:44 pbx sshd[9401]: Did not receive identification string from
UNKNOWN
May 14 00:42:18 pbx sshd[9437]: Did not receive identification string from
UNKNOWN
May 14 00:42:48 pbx sshd[9457]: Did not receive identification string from
UNKNOWN
May 14 00:43:19 pbx sshd[9492]: Did not receive identification string from
UNKNOWN
May 14 00:43:49 pbx sshd[9521]: Did not receive identification string from
UNKNOWN
May 14 00:44:20 pbx sshd[9564]: Did not receive identification string from
UNKNOWN
May 14 00:44:50 pbx sshd[9600]: Did not receive identification string from
UNKNOWN
May 14 00:45:20 pbx sshd[9636]: Did not receive identification string from
UNKNOWN
May 14 00:45:51 pbx sshd[9663]: Did not receive identification string from
UNKNOWN
May 14 00:46:21 pbx sshd[9692]: Did not receive identification string from
UNKNOWN
May 14 00:46:51 pbx sshd[9721]: Did not receive identification string from
UNKNOWN
May 14 00:47:21 pbx sshd[9756]: Did not receive identification string from
UNKNOWN
May 14 00:47:52 pbx sshd[9792]: Did not receive identification string from
UNKNOWN

Thanks,
Bruce
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[asterisk-users] aastra pt 480e phone

[michael capelle]

hello
i hope i am posting to the right list, i am a totally blind user, and i want 
to reprogram my aastra pt 480e phone, my friend used the web configurator, 
but i think he programmed thw wrong codes, a few questions, is it possible 
to damage the phone by programming it wrong? also, how does one reset it 
literally to factory defaults? as unplugging it and replugging it back into 
the wall didn't work.
regards
a happy aastra user.


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[asterisk-users] Channel cannot be released

[kamrun nahar bina]

Dear all,
using asterisk-1.4.23.1, I encountered a problem of asterisk that cannot
release the channel.* *
We have several of asterisk server(client ,Guest). Now channels remaining
problem occurs only in the server where the number of user agent  is more
than 660 and where many simultaneous calling occurs.
Physically, it is being released, but in programming logic, it is not being
released. If we execute "core show channels concise" then we see that the
channels is remaining in server which is not using long time.
Is it the bugs of asterisk or something else? if asterisk has limitation
then how many concurrent call can occur in asterisk? Or how many user agent
can register in one asterisk server? Or is it the server load problem? Or is
it the problem of configuration file settings? We have specified the value
of canreinvite is "no" .
Please let me know.
We have got the channels remaining problem in the following hand set.

Acrobits Softphone version 3.2.2 (iPhone)
SipSimple v4.0/iPhoneOS
snom300/7.1.30
Grandstream HT487 1.0.8.16
Linphone/Linphone-3.1.2 (eXosip2/unknown)...for fax
Sipdroid(Linksys/PAP2-3.1.22(
LS)

Is there any one who knows the solution? Please help me.


Thanks in advance
Nahar
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[asterisk-users] Delay on DTMF with SpeechBackground and Vestec

[Richard Kenner]

I have a delay of "0" on SpeecBackGround, but when I enter DTMF, there's an
almost-exactly five second delay before it returns.  Where is this
delay controlled?  How can I shorten it?

Is there a way to set the maximum number of digits to look for?

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Re: [asterisk-users] Voicemail() app not available?

[Andrew Furey]

On 13/05/2010, Tzafrir Cohen  wrote:
> Specifically, builds 3 different variants of app_voicemail.so as
>  different modules (app_voicemail.so, app_voicemail_imap.so,
>  app_voicemail_odbc.so).

Correct; the other two were noload(ed) by default so I left them.


> What happens if you run:
>
>   module unload app_voicemail.so
>   module   load app_voicemail.so

Aha; I get the following:

asteriskdemo*CLI> module unload app_voicemail.so
Unable to unload resource app_voicemail.so
[May 14 11:06:04] WARNING[15747]: loader.c:501 ast_unload_resource:
Firm unload failed for app_voicemail.so
asteriskdemo*CLI> module load app_voicemail.so
[May 14 11:06:22] ERROR[15747]: app_voicemail.c:7941 load_module:
app_voicemail.so depends upon res_adsi.so

... which I had noload(ed) as well - didn't realise it was required.
(Further looking shows it was in the messages log the whole time;
didn't think to check, been connecting to -r console...)

Loading res_adsi.so fixes the problem. I'll have to go over the
Asterisk Slimming wiki entry...

Thanks Tzafrir!

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett

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Re: [asterisk-users] Are there AMI commands to manipulate a voice mailbox?

[Jim Dickenson]

You might be able to use local channels to do what you want.

As for the user asterisk runs as and the user the web server run as you can 
maybe have both users belong to the same secondary group and gain the access 
you need that way.

Partly depends on what exactly you are wanting to do.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On May 13, 2010, at 5:09 PM, Carlos Chavez wrote:

>   I want to make a web interface so my users can listen/erase voicemails.
> Is there a way to do this from the Asterisk manager interface?  Since
> Asterisk and the web server do not run as the same user I cannot do a
> direct manipulation of the voicemail files
> in /var/spool/asterisk/voicemail.  Maybe there are some AMI commands to
> delete a specific voicemail from a mailbox?  I have not found any so far
> but documentation is often behind of implementation.
> 
>   Any ideas on how to approach this?
> 
> -- 
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
> -- 
> _
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[asterisk-users] Are there AMI commands to manipulate a voice mailbox?

[Carlos Chavez]

I want to make a web interface so my users can listen/erase voicemails.
Is there a way to do this from the Asterisk manager interface?  Since
Asterisk and the web server do not run as the same user I cannot do a
direct manipulation of the voicemail files
in /var/spool/asterisk/voicemail.  Maybe there are some AMI commands to
delete a specific voicemail from a mailbox?  I have not found any so far
but documentation is often behind of implementation.

Any ideas on how to approach this?

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] Asterisk Call Recording *1 Status Indication

[Steve Johnson]

When you press *1 in Asterisk (1.6.2.7) to start/stop call recording,
the console CLI> shows:

> User hit '*1' to record call. filename: wav,auto-1273791789-103-5551212,m

Is it possible to play a sound to back to the person who pressed *1 to
indicate to them that recording has actually started or stopped?
Something like "Recording" / "Record Off", or else sounds like people
are used to hearing when they plug/unplug a USB device into a PC.

Also, I'd also like to have the completed recording go to the person's
voicemail box as a message if that's possible when the recording stops
by toggling *1 or the parties hang up, if anyone has suggestions for
doing that or can point to a link.

Thanks much!  S.

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Re: [asterisk-users] What does Asterisk give to reject a re-invite?

[Kevin P. Fleming]

On 05/13/2010 05:16 PM, David Cunningham wrote:

> We're having an issue where a peer's response to an INVITE includes
> "a=sendonly". Later it sends a re-invite with "a=sendrecv", however
> Asterisk responds to that with an OK that includes "a=recvonly". The
> end result is the called party can't hear the caller.
> 
> Do you have any idea why this is, or where I could go for more information?

That would seem to indicate that the peer is placing Asterisk 'on hold',
and then taking it back 'off hold' later. I do not know why Asterisk
would respond with 'recvonly', it should only do that when it thinks the
channel is still on hold. Are you using 'mohinterpret=passthrough',
where Asterisk would send the hold indication to the bridged channel
instead of reacting to it locally?

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] What does Asterisk give to reject a re-invite?

[David Cunningham]

Kevin,

Thank you for that reply!

We're having an issue where a peer's response to an INVITE includes
"a=sendonly". Later it sends a re-invite with "a=sendrecv", however
Asterisk responds to that with an OK that includes "a=recvonly". The
end result is the called party can't hear the caller.

Do you have any idea why this is, or where I could go for more information?

Thanks for the help.


On Thu, May 13, 2010 at 11:06 PM, Kevin P. Fleming  wrote:
> On 05/13/2010 01:41 PM, David Cunningham wrote:
>
>> If you have canreinvite=no and a peer sends you a re-invite, what will
>> Asterisk reply with?
>
> It will accept it. 'canreinvite' is mis-named, and that's why in more
> modern versions of Asterisk it has been renamed to 'directmedia'.
> Asterisk will *always* accept properly formed re-INVITEs that don't
> require capabilities that are not available, and it will also generate
> them for non-directmedia purposes (like switching to and from T.38) when
> necessary, regardless of whether 'canreinvite' is set to yes or no.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
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>



-- 
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180

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Re: [asterisk-users] What does Asterisk give to reject a re-invite?

[Kevin P. Fleming]

On 05/13/2010 01:41 PM, David Cunningham wrote:

> If you have canreinvite=no and a peer sends you a re-invite, what will
> Asterisk reply with?

It will accept it. 'canreinvite' is mis-named, and that's why in more
modern versions of Asterisk it has been renamed to 'directmedia'.
Asterisk will *always* accept properly formed re-INVITEs that don't
require capabilities that are not available, and it will also generate
them for non-directmedia purposes (like switching to and from T.38) when
necessary, regardless of whether 'canreinvite' is set to yes or no.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Continuing after a TIMEOUT(absolute)

[Zeeshan Zakaria]

It is possible. I do a whole lot of processing after dial and before hanging
up a call. In your case you can try using something like:

exten => h,1,Playback(blah)
exten => h,2,HangUp()

And make sure these lines are in the same context where the Dial command is.

There are other ways too to achieve similar results, but try this first and
it should work.

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.

On 2010-05-13 2:36 PM, "lesouvage"  wrote:

The whole idea of TIMEOUT(absolute) is to end to call after a certain
time. My advice is to explain what you are trying to achieve, there
might be a solutions but I doubt you will find it while using
TIMEOUT(absolute). If the dial plan reaches the t or T extension there
are, as far as I know,  no (bridged) legs of a call left, just some
channel variables you can use.


Erik

On 30 apr 2010, at 18:31, Brendan Sterne wrote:

> CF,
>
> When I comment out the timeout the call c...
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[asterisk-users] What does Asterisk give to reject a re-invite?

[David Cunningham]

Hello,

If you have canreinvite=no and a peer sends you a re-invite, what will
Asterisk reply with?

Thanks,

-- 
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http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180

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Re: [asterisk-users] Continuing after a TIMEOUT(absolute)

[lesouvage]

The whole idea of TIMEOUT(absolute) is to end to call after a certain  
time. My advice is to explain what you are trying to achieve, there  
might be a solutions but I doubt you will find it while using  
TIMEOUT(absolute). If the dial plan reaches the t or T extension there  
are, as far as I know,  no (bridged) legs of a call left, just some  
channel variables you can use.


Erik
On 30 apr 2010, at 18:31, Brendan Sterne wrote:

> CF,
>
> When I comment out the timeout the call continues as expected.  I
> believe the timeout is kicking in.
>
> Can anyone point me to an example where TIMEOUT(absolute) is used as a
> general timer, where the call continues after the expiry?  I'm not
> sure which extension to use "T" or "t".  I've tried both but neither
> seem to work.
>
> Cheers,
> - Brendan
>
> Brendan Sterne
> QA Lead, Callvine
>
>
>
> On Apr 30, 2010, at 9:38 AM, C F wrote:
>
>> I don't think you are actually hitting the time out. Comment out the
>> set timeout line I think the results will be the same. Which tells me
>> the timeout is not kicking in.
>>
>> On 4/29/10, Brendan Sterne  wrote:
>>> Greetings,
>>>
>>> I'm trying to continue to do some processing after a TIMEOUT
>>> (absolute).  In my dialplan below, when a call comes in to  
>>> [default],
>>> I call macro-phonenum and pass it a timeout of 20 seconds.  macro-
>>> phonenum sets TIMEOUT(absolute), then loops saying the phone number
>>> that was called (in MACRO_EXTEN).  When the timeout expires I want  
>>> to
>>> call my macro-hangup (so it can say "goodbye" or whatever).  But the
>>> system is just hanging up.  The dialplan and log output is below.
>>> Any
>>> info is appreciated.  This is on version 1.6.0.5.
>>>
>>>
>>>
>>> [macro-answer-and-join]
>>> exten => s,1,NoOp()
>>> exten => s,n,Answer()
>>> exten => s,n,Wait(4)
>>> exten => s,n,SendDTMF(1)
>>> exten => s,n,Wait(1)
>>> exten => s,n,SendDTMF(1)
>>> exten => s,n,MacroExit
>>>
>>> [macro-hangup]
>>> exten => s,1,NoOp()
>>> exten => s,n,Playback(goodbye)
>>> exten => s,n,Hangup()
>>> ;
>>> exten => T,1,NoOp()
>>> exten => T,n,Playback(goodbye)
>>> exten => T,n,Hangup()
>>>
>>> [macro-phonenum]
>>> exten => s,1,NoOp()
>>> exten => s,n,Macro(answer-and-join)
>>> exten => s,n,Set(TIMEOUT(absolute)=${ARG1})
>>> exten => s,n,Set(i=1000)
>>> exten => s,n,While($[${i} >= 1])
>>> exten =>  s,n,SayDigits(${MACRO_EXTEN})
>>> exten =>  s,n,Wait(5)
>>> exten =>  s,n,Set(i=$[${i} - 1])
>>> exten => s,n,EndWhile()
>>> exten => s,n,MacroExit
>>> ;
>>> exten => T,1,NoOp()
>>> exten => T,n,Macro(hangup)
>>> exten => T,n,MacroExit
>>>
>>>
>>> [default]
>>> exten => _X.,1,NoOp()
>>> exten => _X.,n,Macro(phonenum,20)
>>> exten => _X.,n,Macro(hangup)
>>> ;
>>> exten => T,1,NoOp()
>>> exten => T,n,Macro(hangup)
>>>
>>>
>>>
>>> The log when the timeout occurs:
>>>
>>>  (I'm in macro-phonenum)
>>>   --  Playing 'digits/5.ulaw' (language
>>> 'en')
>>>--  Playing 'digits/1.ulaw' (language
>>> 'en')
>>>--  Playing 'digits/2.ulaw' (language
>>> 'en')
>>>--  Playing 'digits/1.ulaw' (language
>>> 'en')
>>>--  Playing 'digits/2.ulaw' (language
>>> 'en')
>>>-- Executing [...@macro-phonenum:7] Wait("SIP/
>>> 70.124.61.17-082a69a8", "5") in new stack
>>>  == Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
>>> 70.124.61.17-082a69a8' in macro 'phonenum'
>>>  == Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
>>> 70.124.61.17-082a69a8'
>>> Scheduling destruction of SIP dialog 'D8FE9724-1DD1-11B2-9F1A-
>>> a4ef9db84...@192.168.1.98' in 32000 ms (Method: ACK)
>>> set_destination: Parsing  for address/port to
>>> send to
>>> set_destination: set destination to 70.124.61.17, port 5060
>>> Reliably Transmitting (NAT) to 70.124.61.17:5060:
>>> BYE sip:70.124.61.17:5060 SIP/2.0
>>> 
>>>
>>>
>>>
>>> Cheers,
>>> - Brendan
>>>
>>> Brendan Sterne
>>> QA Lead, Callvine
>>>
>>>
>>>
>>>
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[asterisk-users] Sip session timers.

[Leonardo Pistone]

Dear all,

I have a question about session timers. I have one of my installations 
(* 1.6.2.7) where all SIP calls get stuck, like this:

cs4wall*CLI> sip show channels
Peer User/ANR Call ID  Format   Hold 
 Last MessageExpiry
192.168.40.178   42   3c291b87c66e-sl  0x8 (alaw)   No 
  Rx: BYE
192.168.40.179   41   3c29160969a5-3o  0x8 (alaw)   No 
  Rx: BYE
192.168.40.101   34   35e52e9a70318d2  0x8 (alaw)   No 
  Rx: BYE
192.168.40.179   41   249be1a944ffaad  0x8 (alaw)   Yes 
  Rx: BYE Done
192.168.40.179   41   474d9912211df31  0x8 (alaw)   No 
  Rx: BYE
192.168.40.101   patton-quartu460edaf8a643fea  0x8 (alaw)   No 
  Tx: ACK
192.168.40.178   42   3b6b87342aa1299  0x8 (alaw)   No 
  Rx: BYE
192.168.40.179   41   613596f5278d8bc  0x8 (alaw)   Yes 
  Rx: BYE Done
192.168.40.101   patton-quartu7dc4ff993231df7  0x8 (alaw)   No 
  Rx: BYE
192.168.40.178   42   3c291834ab92-x9  0x8 (alaw)   No 
  Rx: BYE
192.168.40.178   42   26c6c35f5659fe8  0x8 (alaw)   No 
  Tx: ACK
11 active SIP dialogs

Strange thing is, I enabled sip session timers=originate, and 
rtptimeout, but this doesn't seem to do anything.

I noticed that a "sip show channel xxx" while the call is actually up says:

>   Session-Timer:  Active
>   S-Timer Interval:   1800
>   S-Timer Refresher:  uac
>   S-Timer Expirys:0
>   S-Timer Sched Id:   2267
>   S-Timer Peer Sts:   Active
>   S-Timer Cached Min-SE:  90
>   S-Timer Cached SE:  1800
>   S-Timer Cached Ref: auto
>   S-Timer Cached Mode:Originate

A few seconds later, when the phone actually hung up, for the same sip 
channel I get:

 >   Session-Timer:  Inactive

This seems strange to me. Any ideas?


Thanks a lot.

Leo
-- 
Leonardo Pistone
SISPAC

Tel. 0114540111, Fax 0114540160

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[asterisk-users] Sending SIP credentials in INVITE

[Mike A. Leonetti]

Is it possible to have Asterisk resend the SIP credentials in every INVITE?

-- 
Mike A. Leonetti
As warm as green tea


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[asterisk-users] Skype for Asterisk and instant messages

[Enrique Mora]

Can Skype for Asterisk process instant messages from Skype users?

I'm wondering if they can be forwarded via email or SMS.


TIA and regards to all

Enrique Mora
Context M.I.S.  SL
em...@context.es
Skype: context-m.i.s.

[cid:image001.jpg@01CAF2BF.0F497C70]


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Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

[Zoa]

Hello,

Can you try trunk = no ?
How much jitter do you see on the link ?

Zoa

Gareth Blades wrote:
> There should be no noticeable difference between slin, ulaw and alaw so 
> what you have is fine. The problem must be elsewhere.
>
> Vieri wrote:
>   
>> --- On Thu, 5/13/10, Gareth Blades  wrote:
>>
>> 
>>> Show the details on the active
>>> channels when using both methods and 
>>> check what codecs are being used.
>>>   
>> The audio codecs are different:
>>
>>Type: SIP
>>   State: Up (6)
>>   Rings: 0
>>   NativeFormats: 0x4 (ulaw)
>> WriteFormat: 0x40 (slin)
>>  ReadFormat: 0x40 (slin)
>>  WriteTranscode: Yes
>>   ReadTranscode: Yes
>>
>>Type: IAX2
>>   State: Up (6)
>>   Rings: 0
>>   NativeFormats: 0x8 (alaw)
>> WriteFormat: 0x8 (alaw)
>>  ReadFormat: 0x8 (alaw)
>>  WriteTranscode: No
>>   ReadTranscode: No
>>
>> By the way, I have this in iax.conf:
>>
>> [interboxIAX2]
>> deny=all
>> allow=ulaw
>> allow=gsm
>> type=friend
>> host=192.168.250.111
>> secret=mysecret
>> auth=plaintext
>> requirecalltoken=no
>> qualify=yes
>> context=mycontext
>> trunk=yes
>> username=interbox
>>
>> Shouldn't the channel details report ulaw instead of alaw?
>>
>> Also, if I change [interboxIAX2] and replace ulaw with alaw, the result is 
>> the same (I still experience bad audio quality).
>>
>> Maybe I should try slin but how do I "force it"?
>>
>> 
>>> Vieri wrote:
>>>   
 Hi,

 I have an audio quality problem regarding IAX2. I have
 
>>> 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps
>>> (no nat, no firewall).
>>>   
 One trunk is SIP and the other IAX2.
 Normally, I use IAX2 but have noticed easily
 
>>> reproducible audio quality problems (voice in/out is OK but
>>> there's a "third" noise overlapping with a "scratchy sound"
>>> as if it were some kind of interference).
>>>   
 So lately I setup calls to go through the SIP trunk
 
>>> and audio quality is OK (no "third overlapping noise").
>>>   
 This is happening between Asterisk 1.4.31 and a
 
>>> 1.2.40.
>>>   
 I'm wondering if there's something I can tweak in IAX2
 
>>> to eliminate this artifact.
>>>   
 Could the IAX2 jitter buffer between 1.2 and 1.4 be an
 
>>> issue (I believe it's enabled by default)?
>>>   
 Thanks,

 Vieri





 
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>>>   
>>   
>>
>> 
>
>
>   


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[asterisk-users] Asterisk Sip Proxies and SIP persistence

[Seann Clark]

All,

   I am looking into open source idea's for something I play with on 
the closed source side. What I am thinking is to get two Asterisk PBX's 
behind a single SIP proxy to load balance calls inbound, and potentially 
outbound to an external sip provider, with the potential of multiple 
provider type lines (SPA3102, and a sip provider) that allows the call 
to persist.



What I am looking at is something like what I do at work. Having an F5 
with a SIP VIP configured, with persistence set up to follow call-id, or 
from, or to, or what ever ends up being best for my environment 
(Typically call-id) between 2 to 30 sip servers/engines.


Since this is more dev/research as a way of saving 12,000-38,000 on a 
device for testing and learning more in depth on sip in transit in a 
home lab environment, I am after something that can do something like 
that, though not expecting the performance of an actual F5, or anything 
like that.



Thanks in advance,
Seann Clark


smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] Voicemail() app not available?

[Tzafrir Cohen]

On Wed, May 12, 2010 at 05:29:29PM +0800, Andrew Furey wrote:
> Hi all,
> 
> I have a demo machine I'm running up on Lenny - it has the packaged
> Asterisk version installed (1.4.21.2+stuff).

Specifically, builds 3 different variants of app_voicemail.so as
different modules (app_voicemail.so, app_voicemail_imap.so,
app_voicemail_odbc.so).

> 
> I'm trying to add an extension to leave a voicemail message, just with
> Voicemail(1234), which I've done before (on 1.2 at least), but it's
> saying "no application 'Voicemail' ".
> 
> "module show like voi" shows "app_voicemail.so" and
> "app_hasnewvoicemail.so" loaded (I have "autoload=yes" in modules.conf
> and have "noload=" a bunch, but I even explicitly set
> "load=app_voicemail.so" just in case.
> 
> However, "core show applications like voi" only lists "HasVoicemail"
> and "HasNewVoicemail", with no sign of "Voicemail". The wiki seems to
> show that it should all be included, with no sign of deprecation...
> 
> Any ideas where I can look?

What happens if you run:

  module unload app_voicemail.so
  module   load app_voicemail.so

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Need fax solution for 1.4.xx (Resolution)

[William Stillwell (Lists)]

Ok, I ended up upgrading 2 of my 5 boxes to 1.6.2.7 , and using spandsp
0.0.6pre17, dahdi-linux-complete-2.3.0+2.3.0 , and enabled app_fax.

Hint: you need to install spandsp then run ./configure then make menuselect
:)


I was able to send over a 50 page fax from coast to coast with 0 issues 

However, did get this message in CLI: 

[May 13 07:27:31] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not
found
[May 13 07:27:31] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not
found
[May 13 07:28:14] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not
found
[May 13 07:28:14] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not
found
[May 13 07:28:27] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not
found
[May 13 07:28:27] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not
found
[May 13 07:28:39] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not
found
[May 13 07:28:39] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not
found

However, there was no noticeable errors in the fax., googling, the error
didn't seem to make much since.

This was via copper pair, over traditional LD carrier, into PRI terminating
into a Sangoma card.

Intel Xeon x3460, 8 gb ram, 320gb raid 0 sata


Thanks to all who offered suggestions, and such, I will try this out, and
hopefully should work well, as Steve Hinted to a year ago.

William Stillwell



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell (Lists)
Sent: Wednesday, May 12, 2010 11:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Need fax solution for 1.4.xx

But that can't handle the call volume, and doesn't support (2) 23B+D now
does it?



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rod Boileau
Sent: Wednesday, May 12, 2010 11:23 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Need fax solution for 1.4.xx

You are right that PIKA no longer just sells Fax licenses to be used
with 3rd party boards.

However the PIKA Warp appliance is great for Faxing with Asterisk.
http://www.pikatechnologies.com/english/View.asp?x=1009

Rod


==
On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists)
 wrote:
> Anybody know a reliable fax solution for 1.4.30 branch?
>
>
> I am using PikaFax  on another server and works very well (about 3000
faxes
> a week), but it appears they no longer offer their product to open
source
> asterisk, only for there "WARP" appliance.
>
> NOT really looking to migrate from 1.4.x to 1.6.x



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Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 30

[Nasir Javaid]

sorry, you r right i just checked it with registration so there were astdb
entries for SIP registration.

anyhow after clearing settings frm astdb i tried the same scenario you
advised but no luck.
I think i told that i am not using server as peer but want to use a user
[abc] as peer so that when ever i use

dial(SIP/${ext...@abc)  or  dial(SIP/abc/${EXTEN})

the call will be out from server using [abc]'s account.
i hope you understand what i mean.

also i will like to know is there any way that i can include registration
information in my dial string so that i have no need to write

register => abc:mysec...@nasir.server.com:8060

regards,

Nasir Javaid


Look, you do again with registration.
remove any registration information.
Look this config, I think it can help you


Server1:

sip.conf

[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729

extensions.conf

[calltoserver2]
 exten =>  _X.,1,Noop(Call to server2)
 exten =>  _X.,2,Dial(SIP/
interboxserver2/${EXTEN})
 exten =>  _X.,3,Hangup

[callfromserver2]

exten => _X.,1,Noop(Call from server2)
exten => _X.,2,Dial(SIP/${EXTEN})
exten => _X.,3,Hangup


Server2:

sip.conf

[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729

extensions.conf

[calltoserver1]
 exten =>  _X.,1,Noop(Call to server1)
 exten =>  _X.,2,Dial(SIP/interboxserver1/${EXTEN})
 exten =>  _X.,3,Hangup

[callfromserver1]

exten => _X.,1,Noop(Call from server1)
exten => _X.,2,Dial(SIP/${EXTEN})
exten => _X.,3,Hangup


Try so, I think it must work.
And also, look and delete any another records in both servers in
sip.conf about this servers settings.

Vardan
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Re: [asterisk-users] app_addon_sql_mysql.c:116 find_identifier

[Doug Lytle]

Jonas Kellens wrote:
> exten => s,n,NoOp(fetchid = ${fetchid})
> exten => s,n,MYSQL(Clear ${resultid})
> exten => s,n,MYSQL(Disconnect ${connid})

The only different between yours and mine is that I do a disconnect 
before I do the clear.

Try:

exten => s,n,MYSQL(Fetch fetchid ${resultid} extensie)
exten => s,n,NoOp(fetchid = ${fetchid})
exten => s,n,MYSQL(Disconnect ${connid})
exten => s,n,MYSQL(Clear ${resultid})


My guess is (If it works), that you can't clear an open channel.

Doug


-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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[asterisk-users] app_addon_sql_mysql.c:116 find_identifier

[Jonas Kellens]

Hello list,

I have the following problem with MySQL-queries : it seems that the 
resultid and connid are not cleared !


[macro-GetMailboxFromSIPuserID]
exten => s,1,MYSQL(Connect connid localhost xxx xxx xxx)
exten => s,n,MYSQL(Query resultid ${connid} SELECT\ extensie FROM\ 
tbl_SIPaccounts\ WHERE\ ID="${ARG1}")

exten => s,n,MYSQL(Fetch fetchid ${resultid} extensie)
exten => s,n,NoOp(fetchid = ${fetchid})
exten => s,n,MYSQL(Clear ${resultid})
exten => s,n,MYSQL(Disconnect ${connid})
exten => s,n,MacroExit()

[May 13 16:00:34] -- Executing [...@macro-getmailboxfromsipuserid:1] 
MYSQL("SIP/testcorp-0827e670", "Connect connid localhost xxx xxx xxx") 
in new stack
[May 13 16:00:34] -- Executing [...@macro-getmailboxfromsipuserid:2] 
MYSQL("SIP/testcorp-0827e670", "Query resultid 19 SELECT extensie FROM 
tbl_SIPaccounts WHERE ID="105002"") in new stack
[May 13 16:00:34] -- Executing [...@macro-getmailboxfromsipuserid:3] 
MYSQL("SIP/testcorp-0827e670", "Fetch fetchid 20 extensie") in new stack
[May 13 16:00:34] -- Executing [...@macro-getmailboxfromsipuserid:4] 
NoOp("SIP/testcorp-0827e670", "fetchid = 0") in new stack
[May 13 16:00:34] -- Executing [...@macro-getmailboxfromsipuserid:5] 
MYSQL("SIP/testcorp-0827e670", "*Clear 20*") in new stack
[May 13 16:00:34] -- Executing [...@macro-getmailboxfromsipuserid:6] 
MYSQL("SIP/testcorp-0827e670", "*Disconnect 19*") in new stack
[May 13 16:00:34] -- Executing [...@macro-getmailboxfromsipuserid:7] 
MacroExit("SIP/testcorp-0827e670", "") in new stack



[macro-SDgeenopname])
exten => s,1,MYSQL(Connect connid5 localhost xxx xxx xxx)
exten => s,n,MYSQL(Query resultid5 ${connid5} DELETE\ FROM\ recordings\ 
WHERE\ file="${xxx}")

exten => s,n,MYSQL(Clear ${resultid5})
exten => s,n,MYSQL(Disconnect ${connid5})

[May 13 16:00:39] -- Executing [...@macro-sdgeenopname:3] 
MYSQL("SIP/testcorp-0827e670", "Connect connid localhost xxx xxx xxx") 
in new stack
[May 13 16:00:39] -- Executing [...@macro-sdgeenopname:4] 
MYSQL("SIP/testcorp-0827e670", "Query resultid 19 DELETE FROM recordings 
WHERE file="xxx"") in new stack
[May 13 16:00:39] -- Executing [...@macro-sdgeenopname:5] 
MYSQL("SIP/testcorp-0827e670", "*Clear 20*") in new stack
[May 13 16:00:39] WARNING[4358]: app_addon_sql_mysql.c:116 
find_identifier: Identifier 20, identifier_type 2 not found in 
identifier list
[May 13 16:00:39] WARNING[4358]: app_addon_sql_mysql.c:355 aMYSQL_clear: 
Invalid result identifier 20 passed in aMYSQL_clear
[May 13 16:00:39] -- Executing [...@macro-sdgeenopname:6] 
MYSQL("SIP/testcorp-0827e670", "*Disconnect 19*") in new stack



With every call it's the same... something with this '19' and '20'...


Jonas.
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[asterisk-users] Asterisk Crashing with ERROR[1906] astobj2.c: refcount -1 on object 0xb1aab758 Ast Ver 1.6.2.6

[Steve Totaro]

Hello,

Anyone have any insight or fix for the error below?  It was the last error
in the log before Asterisk crashed.  I am running Asterisk 1.6.2.6 only for
the T.38 support.

06:21:49] ERROR[1906] astobj2.c: refcount -1 on object 0xb1aab758

Google has some vague references that there is a patch or something but
nothing I could really follow.  Anyone have any ideas?

The system was up for about a month before this error and subsequent crash
occurred.

Thanks,
Steve Totaro
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Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

[Gareth Blades]

There should be no noticeable difference between slin, ulaw and alaw so 
what you have is fine. The problem must be elsewhere.

Vieri wrote:
> --- On Thu, 5/13/10, Gareth Blades  wrote:
> 
>> Show the details on the active
>> channels when using both methods and 
>> check what codecs are being used.
> 
> The audio codecs are different:
> 
>Type: SIP
>   State: Up (6)
>   Rings: 0
>   NativeFormats: 0x4 (ulaw)
> WriteFormat: 0x40 (slin)
>  ReadFormat: 0x40 (slin)
>  WriteTranscode: Yes
>   ReadTranscode: Yes
> 
>Type: IAX2
>   State: Up (6)
>   Rings: 0
>   NativeFormats: 0x8 (alaw)
> WriteFormat: 0x8 (alaw)
>  ReadFormat: 0x8 (alaw)
>  WriteTranscode: No
>   ReadTranscode: No
> 
> By the way, I have this in iax.conf:
> 
> [interboxIAX2]
> deny=all
> allow=ulaw
> allow=gsm
> type=friend
> host=192.168.250.111
> secret=mysecret
> auth=plaintext
> requirecalltoken=no
> qualify=yes
> context=mycontext
> trunk=yes
> username=interbox
> 
> Shouldn't the channel details report ulaw instead of alaw?
> 
> Also, if I change [interboxIAX2] and replace ulaw with alaw, the result is 
> the same (I still experience bad audio quality).
> 
> Maybe I should try slin but how do I "force it"?
> 
>> Vieri wrote:
>>> Hi,
>>>
>>> I have an audio quality problem regarding IAX2. I have
>> 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps
>> (no nat, no firewall).
>>> One trunk is SIP and the other IAX2.
>>> Normally, I use IAX2 but have noticed easily
>> reproducible audio quality problems (voice in/out is OK but
>> there's a "third" noise overlapping with a "scratchy sound"
>> as if it were some kind of interference).
>>> So lately I setup calls to go through the SIP trunk
>> and audio quality is OK (no "third overlapping noise").
>>> This is happening between Asterisk 1.4.31 and a
>> 1.2.40.
>>> I'm wondering if there's something I can tweak in IAX2
>> to eliminate this artifact.
>>> Could the IAX2 jitter buffer between 1.2 and 1.4 be an
>> issue (I believe it's enabled by default)?
>>> Thanks,
>>>
>>> Vieri
>>>
>>>
>>>
>>>
>>>
>>
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>>
> 
> 
>   
> 


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Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

[Vieri]

--- On Thu, 5/13/10, Gareth Blades  wrote:

> Show the details on the active
> channels when using both methods and 
> check what codecs are being used.

The audio codecs are different:

   Type: SIP
  State: Up (6)
  Rings: 0
  NativeFormats: 0x4 (ulaw)
WriteFormat: 0x40 (slin)
 ReadFormat: 0x40 (slin)
 WriteTranscode: Yes
  ReadTranscode: Yes

   Type: IAX2
  State: Up (6)
  Rings: 0
  NativeFormats: 0x8 (alaw)
WriteFormat: 0x8 (alaw)
 ReadFormat: 0x8 (alaw)
 WriteTranscode: No
  ReadTranscode: No

By the way, I have this in iax.conf:

[interboxIAX2]
deny=all
allow=ulaw
allow=gsm
type=friend
host=192.168.250.111
secret=mysecret
auth=plaintext
requirecalltoken=no
qualify=yes
context=mycontext
trunk=yes
username=interbox

Shouldn't the channel details report ulaw instead of alaw?

Also, if I change [interboxIAX2] and replace ulaw with alaw, the result is the 
same (I still experience bad audio quality).

Maybe I should try slin but how do I "force it"?

> Vieri wrote:
> > Hi,
> > 
> > I have an audio quality problem regarding IAX2. I have
> 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps
> (no nat, no firewall).
> > One trunk is SIP and the other IAX2.
> > Normally, I use IAX2 but have noticed easily
> reproducible audio quality problems (voice in/out is OK but
> there's a "third" noise overlapping with a "scratchy sound"
> as if it were some kind of interference).
> > 
> > So lately I setup calls to go through the SIP trunk
> and audio quality is OK (no "third overlapping noise").
> > 
> > This is happening between Asterisk 1.4.31 and a
> 1.2.40.
> > 
> > I'm wondering if there's something I can tweak in IAX2
> to eliminate this artifact.
> > 
> > Could the IAX2 jitter buffer between 1.2 and 1.4 be an
> issue (I believe it's enabled by default)?
> > 
> > Thanks,
> > 
> > Vieri
> > 
> > 
> > 
> >       
> > 
> 
> 
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> _
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> 


  

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Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

[Steve Totaro]

On Thu, May 13, 2010 at 4:17 AM, Vieri  wrote:

> Hi,
>
> I have an audio quality problem regarding IAX2. I have 2 Asterisk servers
> interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
> One trunk is SIP and the other IAX2.
> Normally, I use IAX2 but have noticed easily reproducible audio quality
> problems (voice in/out is OK but there's a "third" noise overlapping with a
> "scratchy sound" as if it were some kind of interference).
>
> So lately I setup calls to go through the SIP trunk and audio quality is OK
> (no "third overlapping noise").
>
> This is happening between Asterisk 1.4.31 and a 1.2.40.
>
> I'm wondering if there's something I can tweak in IAX2 to eliminate this
> artifact.
>
> Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe
> it's enabled by default)?
>
> Thanks,
>
> Vieri
>
>
> IAX2 has been borken since Jump Street.  You can find posts of mine dating
back years stating exactly what you just described.  Of course, plenty of
people have "No Problems" with IAX2, and the official Digium party line is
that is "Great"

I only use it for situations where NAT makes SIP impossible since it only
uses one port for everything.  Don't bother with it on a LAN or VPN, only
use it in a pinch.

I suggest that you use 1.4.X or newer if you 1.6.X if you are a bit daring.
I believe IAX2 has had alot of reworking, so using the latest and same
version on your boxen should help, but although Asterisk is not RFC SIP
compliant, it works well.  IAX2 looks and sound good on paper, just not the
every important phone.

I have made a good deal of money consulting, only to find out that the
customer was using IAX2 somewhere, after switching to SIP, they had perfect
audio, these were mostly ITSPs using "trunking" but I have seen the same
issue with no trunking (using same protocol overhead for all simultaneous
calls, rather than a separate overhead per call.

I advise bagging IAX2 for now if you can.  I am forced to use it when ISPs
are using several NATs in their networks, but then and only then.  If you
can setup OpenVPN, then do it and use SIP.  If you are forced to use IAX2,
don't trunk and try to use the latest stable, or a version or two prior.

I have a few friends at Digium that have told me that IAX2 was a work in
progress but not ready for large scale prime time (isn't everything) and
also that Realtime was also sub-par and needed to be re-written from
scratch.  This was a couple of years ago.

Thanks,
Steve Totaro
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Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

[Gareth Blades]

Show the details on the active channels when using both methods and 
check what codecs are being used.

Vieri wrote:
> Hi,
> 
> I have an audio quality problem regarding IAX2. I have 2 Asterisk servers 
> interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
> One trunk is SIP and the other IAX2.
> Normally, I use IAX2 but have noticed easily reproducible audio quality 
> problems (voice in/out is OK but there's a "third" noise overlapping with a 
> "scratchy sound" as if it were some kind of interference).
> 
> So lately I setup calls to go through the SIP trunk and audio quality is OK 
> (no "third overlapping noise").
> 
> This is happening between Asterisk 1.4.31 and a 1.2.40.
> 
> I'm wondering if there's something I can tweak in IAX2 to eliminate this 
> artifact.
> 
> Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe it's 
> enabled by default)?
> 
> Thanks,
> 
> Vieri
> 
> 
> 
>   
> 


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[asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

[Vieri]

Hi,

I have an audio quality problem regarding IAX2. I have 2 Asterisk servers 
interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
One trunk is SIP and the other IAX2.
Normally, I use IAX2 but have noticed easily reproducible audio quality 
problems (voice in/out is OK but there's a "third" noise overlapping with a 
"scratchy sound" as if it were some kind of interference).

So lately I setup calls to go through the SIP trunk and audio quality is OK (no 
"third overlapping noise").

This is happening between Asterisk 1.4.31 and a 1.2.40.

I'm wondering if there's something I can tweak in IAX2 to eliminate this 
artifact.

Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe it's 
enabled by default)?

Thanks,

Vieri



  

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Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]

[Vieri]

Issue solved.
Looks like all I was missing was one parameter:
"fromuser="
Thanks for your time!




  

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Re: [asterisk-users] Error at start of asterisk with cdr_addon_mysql.o

[Pham Quy]

On Wed, 2010-05-12 at 22:10 -0700, Steve Edwards wrote:
> On Thu, 13 May 2010, Pham Quy wrote:
> 
> > Hi all,
> >
> > I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
> >
> > It started ok with out cdr_addon_mysql.o. But when I put
> > cdr_addon_mysql.o in to modules folder, it fail at start and the
> > following out has been thrown:
> >
> > --
> > [r...@localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270
> > Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk
> > -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
> > Asterisk exited with exit status 139
> > Asterisk exited on signal 11
> > Automatically restarting Asterisk.
> > ---
> >
> > What is the problem?
> 
> The problem is...
> 
> You have no clue[s] :)
> 
> First off, the module should be cdr_addon_mysql.so, not cdr_addon_mysql.o. 
> If you don't have the "so" in "/usr/lib/asterisk/modules/" something is 
> wrong with your build.
> 
> Try something like this:
> 
>   sudo -u \
>   /usr/sbin/asterisk -c -d -d -d -f -g -n -v -v -v
> 
> Or, you can start Asterisk without loading cdr_addon_mysql.so and then 
> load it from the Asterisk CLI. It sounds like you are auto-loading modules 
> so you could add "noload=cdr_addon_mysql.so" to 
> "/etc/asterisk/modules.conf" to get Asterisk running and then load it with 
> something like "load cdr_addon_mysql.so"
> 
> I'm a 1.2 Luddite so the commands may have changed slightly. Also, 
> depending on the specifics of your installation, the paths may be 
> different.
> 
> See if this gives you any clues.
> 
> -- 
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
> 

Hi, 

Yes, it was a typing mistake, i meant cdr_addon_mysql.so. After manually
loadind the module, it turn out there is an mistake in my cdr_mysql.conf

I fixed it and everything work fine.

Thanks.
Quyps


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