date:20090304

Re: [asterisk-users] Access sip.conf's mailbox from dialplan ?

2009-03-04 Thread Klaus Darilion

core show function SIPPEER


Olivier schrieb:
 Hello,
 
 In sip.conf, each peer/friend/user entry gathers several parameters such 
 as type, canreinvite or mailbox.
 How can you specifically access to mailbox value from dialplan ?
 
 I know how to access custom parameters (ie setvar=FOO=value) but I don't 
 know to access standard parameters.
 
 I'm specifically concerned to access to mailbox's value (from a given 
 entry) but would be delighted to discover a general mechanism to access 
 to other parameters (canreinvite, ...)
 
 Regards
 
 
 
 
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Re: [asterisk-users] How to generate core dump?

2009-03-04 Thread Klaus Darilion



Danny Nicholas schrieb:
 You would think this, but I've seen asterisk create 100 or more dumps in an
 hour of 10+Mb.  Depending on Inode size, etc., this situation could push a
 system into a hurting capacity rather quickly.  Also, many shops use older
 technology and compound this by RAID striping, which can reduce your
 effective capacity by up to 70%.  Just an observation.

If my Asterisk crashes 100 times per hour I would not be concerned about 
disk sizes, but about the service Asterisk should offer.

klaus
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RE Kushner
 List Account
 Sent: Monday, March 02, 2009 10:58 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] How to generate core dump?
 
 Danny Nicholas wrote:
 You could change your asterisk command to asterisk -vvg, but this will eat
 your disk space if you have a large number of faults since each core.*
 file
 produced takes up 1-13 Mb.

   
 
 In the day and age where 500GB hard drives are $75 at Micro Center, hard 
 drive space shouldn't be a concern to many unless you've recycled a 
 system older than four years old. I was out yesterday and the cheapest 
 and smallest drive they carried was a WD 160GB for $40, and until the 
 last year the smallest drive you could get was 80GB, and the year before 
 that 60GB.
 
 It'd take weeks of core dumps before a blip would show up in df unless 
 it's constantly core dumping, which from what he said I suspect is not 
 the case.
 
 -Ron
 
 
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Re: [asterisk-users] after install the zaptel but the rtp failed

2009-03-04 Thread Grygoriy Dobrovolskyy

2009/3/4 邱磊 qiulei...@163.com

  hi Grygoriy :
 appreciate your reply ,
 that's my cli command:
 CLI zap show status
 Description  Alarms IRQbpviol
 CRC4
 ZTDUMMY/1 1  UNCONFIGUR 0  0
 0

 Is't all right? forward your echo .
 thanks


Yes normally you should have meetme working. Paste your extensions.conf here
(only the context with the conference) Also the config of the sip peer who
is trying to join the conference and more cli output during that join.
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Re: [asterisk-users] How to generate core dump?

2009-03-04 Thread Klaus Darilion



Mark Michelson schrieb:
 Ken D'Ambrosio wrote:
 Asterisk segfaulted on me the other day; how do I tell it to generate a
 core file so -- if it happens again -- I can attempt to debug?  I looked
 in the obvious places in make menuconfig and didn't see anything
 appropriate.

 Thanks,

 -Ken


 
 Run Asterisk with the -g option and it will dump a core file if it should 
 crash.

If you also want to specify the location/file name this can be useful 
too (man core)

echo /tmp/core.%p  /proc/sys/kernel/core_pattern

klaus

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Re: [asterisk-users] SIP dialog matching problem? (1.4.23.1)

2009-03-04 Thread Klaus Darilion

Hi!

Actually I would consider this as a bug, thus you should report it at 
bugs.digium.com.

Are you using pedantic=yes (sip.conf)? If not, it would be interesting 
if the pedantic mode has the same problem.

regards
klaus

Santiago Gimeno schrieb:
 Hello all,
 
 Not sure if this mail belongs to this users or dev list. Sorry about
 that.
 
 We have the following scenario:
 
   PhoneA OpenSER   AsteriskPhoneB PhoneC
  |  |  |  |  |
  |  |  |  |  |
  |  |  |  |  |
  |INVITE B  |  |  |  |
  |-|  |  |  |
  |  |INVITE B  |  |  |
  |  |-|  |  |
  |  |  |INVITE B  |  |
  |  |  |-|  |
  |  |  |486 Busy Here |  |
  |  |  |-|  |
  |  |  |ACK   |  |
  |  |  |-|  |
  |  |486 Busy Here |  |  |
  |  |-|  |  |
  |  |ACK   |  |  |
  |  |-|  |  |
  |302 MOVED (to C) |  |  |
  |-|  |  |  |
  |ACK   |  |  |  |
  |-|  |  |  |
  |INVITE C  |  |  |  |
  |-|  |  |  |
  |  |INVITE C  |  |  |
  |  |-|  |  |
  |  |503 Unavailable  |  |
  |  |-|  |  |
  |  |ACK   |  |  |
  |  |-|  |  |
  |503 Unavailable  |  |  |
  |-|  |  |  |
  |ACK   |  |  |  |
  |-|  |  |  |
  |  |  |  |  |
  |  |  |  |  |
 
 
 
 1.- Phone A calls Phone B behind Asterisk.
 2.- Phone B rejects call by sending a '486 Busy Here' response.
 3.- When OpenSER receives the 486 it sends a '302 Moved Temporarily'
 to Phone A to redirect the call to Phone C.
 4.- Phone A perfoms the redirection and sends a new INVITE to Phone C
 (that is also behind Asterisk) with same call-id BUT DIFFERENT from-tag,
 CSeq.
 5.- Asterisk, for some reason, considers the new INVITE to belong to the
 previous call and then rejects the call with a
 '503 Unavailable'. But it cannot be considered to belong to the same
 dialog because the tags are different, although the call-id is the same.
 We have used pedantic checking. Could it be considered as a bug?
 
 Looking at the code of chan_sip.c (version 1.4.23.1), we have observed
 that in function 'find_call' line 4667, asterisk is considering the call
 as FOUND because of this test:
 !ast_test_flag(p-flags[1], SIP_PAGE2_DIALOG_ESTABLISHED).
 Commenting out this comparison, the call proceeds correctly. Sure, there
 is some reason for this checking and we would like to know which is and
 in what does it affect. How could we fix it?
 
 The following is the asterisk console output when the call does not
 proceed:
 [Mar  2 12:15:24] DEBUG[9989]: chan_sip.c:15813 handle_request: 
 Received INVITE (5) - Command in SIP INVITE [Mar  2 12:15:24]
 NOTICE[9989]: chan_sip.c:14724
 handle_request_invite: Unable to create/find SIP channel for this INVITE
 [Mar  2 12:15:24] DEBUG[9989]: chan_sip.c:4653 find_call: = Looking for
 Call ID: 9463d153-64f11de-8602e9bf-a87f5...@172.16.103.15 
 mailto:9463d153-64f11de-8602e9bf-a87f5...@172.16.103.15
 (Checking From) --From tag 182B3580-E9 --To-tag as62e21069
 
 Any feedback would be appreciated.
 
 Thank you in advance,
 
 Santi
 
 
 
 
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[asterisk-users] Question on phone line pass through

2009-03-04 Thread Mikel Lindsaar

Hi all,

I have a TDM2400 full of FXS cards and a TDM800 full of FXO cards.

If I have a fax machine on the FXS port dialing out through asterisk
on the TDM800 FXO, should I be expecting any problems?

Or should this just work as expected? (ie, flawlessly with the
asterisk box essentially transparent to the whole operation).

I am doing it this way to allow many faxes and modems to share a dial out pool.

Mikel


-- 
http://lindsaar.net/
Rails, RSpec and Life blog

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[asterisk-users] Master.csv - disposition value (based on?)

2009-03-04 Thread Tobias Steen

I need to get the effective time for a call and therefore I wonder if the 
disposition field in the Master.csv are based on the effective call time with 
an agent or does this value also including the callers holdtime in queue?

Many thanks!

Regards
Tobias Steén
_
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Re: [asterisk-users] SIP dialog matching problem? (1.4.23.1)

2009-03-04 Thread Santiago Gimeno

Hello,

Thanks for the reply.

Yes, I'm using pedantic=yes. I will report this asap.

One more thing that I have observed and might be also related to this issue.

The scenario is the same as the one I described in the previous mail, but in
this case, the SIP Phone that receives the 302 generates a new INVITE to the
new address with exactly the same dialog information as the initial
INVITE: call-id, from-tag and to-tag. (I think this is legal as stated in
the RFC 3261-8.1.3.4: *It is RECOMMENDED that the UAC reuse the same To,
From, and Call-ID used in the original redirected request, but the UAC MAY
also choose to update the Call-ID header field value for new requests, for
example.*). Asterisk answers to this INVITE with a 503 Unavailable because
it matches with the previous dialog. I'm not sure if this is how Asterisk
should behave, or it should allow the call to progress as the previous
dialog is already in the TERMINATED state. What do you think?

Best regards,

Santi

2009/3/4 Klaus Darilion klaus.mailingli...@pernau.at

 Hi!

 Actually I would consider this as a bug, thus you should report it at
 bugs.digium.com.

 Are you using pedantic=yes (sip.conf)? If not, it would be interesting
 if the pedantic mode has the same problem.

 regards
 klaus

 Santiago Gimeno schrieb:
  Hello all,
 
  Not sure if this mail belongs to this users or dev list. Sorry about
  that.
 
  We have the following scenario:
 
PhoneA OpenSER   AsteriskPhoneB PhoneC
   |  |  |  |  |
   |  |  |  |  |
   |  |  |  |  |
   |INVITE B  |  |  |  |
   |-|  |  |  |
   |  |INVITE B  |  |  |
   |  |-|  |  |
   |  |  |INVITE B  |  |
   |  |  |-|  |
   |  |  |486 Busy Here |  |
   |  |  |-|  |
   |  |  |ACK   |  |
   |  |  |-|  |
   |  |486 Busy Here |  |  |
   |  |-|  |  |
   |  |ACK   |  |  |
   |  |-|  |  |
   |302 MOVED (to C) |  |  |
   |-|  |  |  |
   |ACK   |  |  |  |
   |-|  |  |  |
   |INVITE C  |  |  |  |
   |-|  |  |  |
   |  |INVITE C  |  |  |
   |  |-|  |  |
   |  |503 Unavailable  |  |
   |  |-|  |  |
   |  |ACK   |  |  |
   |  |-|  |  |
   |503 Unavailable  |  |  |
   |-|  |  |  |
   |ACK   |  |  |  |
   |-|  |  |  |
   |  |  |  |  |
   |  |  |  |  |
 
 
 
  1.- Phone A calls Phone B behind Asterisk.
  2.- Phone B rejects call by sending a '486 Busy Here' response.
  3.- When OpenSER receives the 486 it sends a '302 Moved Temporarily'
  to Phone A to redirect the call to Phone C.
  4.- Phone A perfoms the redirection and sends a new INVITE to Phone C
  (that is also behind Asterisk) with same call-id BUT DIFFERENT from-tag,
  CSeq.
  5.- Asterisk, for some reason, considers the new INVITE to belong to the
  previous call and then rejects the call with a
  '503 Unavailable'. But it cannot be considered to belong to the same
  dialog because the tags are different, although the call-id is the same.
  We have used pedantic checking. Could it be considered as a bug?
 
  Looking at the code of chan_sip.c (version 1.4.23.1), we have observed
  that in function 'find_call' line 4667, asterisk is considering the call
  as FOUND because of this test:
  !ast_test_flag(p-flags[1],

Re: [asterisk-users] AEL2: If-then-else not permitted in Switch-Case

2009-03-04 Thread Philipp Kempgen

Lee, John (Sydney) schrieb:
 I just want to confirm but it seems that if-then-else is not permitted
 in case structure.
 It was not really documented but it seems to be the case.
 
 Can anyone confirm?
 
 switch(${DIALSTATUS})
   {
 case NOANSWER:
  {
// if-then-else not permitted
If (${ael-var} = 1)
{
  Playback(beep); 
  return;
}
  }

I would have written this like so:

switch (${DIALSTATUS}) {
case NOANSWER:
if (${ael-var} = 1) {
Playback(beep);
}
break;
}

Give it a try.


Philipp Kempgen
-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Access sip.conf's mailbox from dialplan ? [SOLVED]

2009-03-04 Thread Olivier

2009/3/4 Klaus Darilion klaus.mailingli...@pernau.at

 core show function SIPPEER


Thanks : that's exactly what I was looking for !!
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Re: [asterisk-users] CDR

2009-03-04 Thread Gustavo A Gonzalez

Thanks Carlos for your response, I have in my Zapata.conf immediate=no, is
this field that affect start time and answer time?. Because of this I have
the fields DURATION and BILLSEC with the same value, it means that Im
billing from start time, what I need is billing from answer time. How I
would do it?

 

Cheers!!

Gustavo A. González



 

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Re: [asterisk-users] Silk for Free

2009-03-04 Thread Steve Underwood

Dean Collins wrote:

http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio_codec.html?tk=rss_news

any thoughts?

They have said it will be royalty free, but they have said little else.

From discussions with Skype people in the last few days they seem very
reluctant to hand out source code, so it looks like they will provide
binary blobs for whatever platforms they choose to support. They are
clearly eager to get Skype broadly connected to corporate networks, but
if they don't get this codec into a broad range of phones its a waste of
time. Transcoding looses too much quality.. If they don't hand out the
source, or at least provide a rigorous spec, I don't think this will
fly. Even rigorous specs aren't really enough. Pretty much all modern
codecs are defined by their reference implementation.

The bit rate is supposed to dynamically adapt to network conditions,
when the code is used in conjunction with a suitable network performance
monitor. Exactly what those bit rates are, however, still seems to be a
mystery. They claim audio up to 12kHz, and specifically say they are
suppressing the bass end below 70Hz as it just sounds nasty. That's
sad. 12kHz isn't really enough for high quality voice, and the extra bit
rate needed to push the bandwidth to 15kHz is small. Also, a deep man's
voice looses something when you cut off at 70Hz. You really want the
bass to extend to 40Hz or 50Hz.

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