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
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
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
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
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
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
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
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
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:
{
//
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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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
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
Hi!
Sorry for not knowing how hash table works, thus my questions may be a
bit stupid:
How many items can be stored in hashtable? Is it limited to size
parameter, e.g. 10 means max. 1024 entries? Or is it as much as memory
is available (what for is the size parameter in this case)?
How can I
hi.
is out there any how to install druid whit out the iso?
thanks
David
2009/3/4 Ming Yong m...@voiceroute.net
Dear Asterisk users,
We would like to announce that Druid, Open Source Unified Communications
project has just made a major release: Druid 2.0. It is out!It has a ton of
new
Hi there,
has anyone seen specifications of the codec g711-HD? This is right now
spreading fast in the wake up CATiq (the DECT successor), for example in
the AVM products (www.avm.de).
Is this a re-branded g711.1 (rfc5391) and therefore compatiable with it,
or a different animal? And what are
2009/3/4 Mikel Lindsaar raasd...@gmail.com
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?
I think you should get problems for faxing.
If
Looks like good!!
Congratulations i expect testing your solution
Regards,
On Wed, Mar 4, 2009 at 2:07 AM, Ming Yong m...@voiceroute.net wrote:
Dear Asterisk users,
We would like to announce that Druid, Open Source Unified Communications
project has just made a major release: Druid 2.0. It
Philipp von Klitzing wrote:
Hi there,
has anyone seen specifications of the codec g711-HD? This is right now
spreading fast in the wake up CATiq (the DECT successor), for example in
the AVM products (www.avm.de).
Is this a re-branded g711.1 (rfc5391) and therefore compatiable with it,
Damn you for solving this before he upped the bounty by a pack of tictacs!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: March 3, 2009 10:51 PM
To: Asterisk Users List
Subject: Re:
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:
{
//
sorry - wrong mailing list ...
Klaus Darilion schrieb:
Hi!
Sorry for not knowing how hash table works, thus my questions may be a
bit stupid:
How many items can be stored in hashtable? Is it limited to size
parameter, e.g. 10 means max. 1024 entries? Or is it as much as memory
is
Dean Collins wrote:
http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio_codec.html?tk=rss_news
any thoughts?
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
On Wed, Mar 04, 2009 at 09:32:43AM +0100, Klaus Darilion wrote:
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
Install a Microsoft product.
(Sorry I couldn't resist when I saw the subject)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: March 4, 2009 8:48 AM
To: Asterisk Users List
Subject: Re:
BJ Weschke wrote:
Cheaper to give away for hopes of proliferation what you've already
implemented versus having someone else get theirs proliferated and
popular first and then you are strapped with the cost of implementation
of someone else's popular and free codec?
Polycom's Siren7
Klaus Darilion schrieb:
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:
It's conceivable that the combined effort of these two responders
required less than ten minutes of time, yielding a theoretical pay rate
of $120/hour.
I wonder how much effort went into the other responses.
That will be $6 for my commentary, please.
Folks wrote:
Message: 1
Date: Tue, 3 Mar
Steve Underwood wrote:
CAT-iq supports G.722 for wideband voice. the CAT-iq web site describes
this as CD quality. I guess the person who wrote that has severely
impaired hearing. :-)
Maybe they meant 'CD quality after compression with MPEG layer 3 to 128
kilobits per second' :-)
--
Kevin
On 03/04/09 19:31, Michael wrote:
On Wed, 04 Mar 2009 19:25:38 Joseph wrote:
I'm faxing from stand alone fax machine via linksys SPA3102 but most of
the time only half or quarter page goes through.
Did anybody have any experience like this?
Should be obvious but does your up line SIP
I know it is possible, as this is how park works. Steve Edwards can answer
this better since he's always dissing my replies :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Mutuku
Sent: Tuesday, March
On 03/04/09 19:31, Michael wrote:
On Wed, 04 Mar 2009 19:25:38 Joseph wrote:
I'm faxing from stand alone fax machine via linksys SPA3102 but most of
the time only half or quarter page goes through.
Did anybody have any experience like this?
Should be obvious but does your up line SIP
What app are you using to receive the fax? If it is RXFAX, try turning on
the ECM and or DEBUG options.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Wednesday, March 04, 2009 8:36 AM
To:
Hi Joseph.
I've spent some time tuning the SPA3102 FXS line input and output gain
and I think that this is an important variable.
Let's try to record incoming and outgoing fax tones with asterisk on SIP
channel (disabling the fax detection on SPA and sending fax inband) and
look at the recorded
On Wed, Mar 4, 2009 at 9:11 AM, Bill Michaelson b...@cosi.com wrote:
It's conceivable that the combined effort of these two responders required
less than ten minutes of time, yielding a theoretical pay rate of $120/hour.
I wonder how much effort went into the other responses.
I'm reminded of
On 03/04/09 08:38, Danny Nicholas wrote:
What app are you using to receive the fax? If it is RXFAX, try turning on
the ECM and or DEBUG options.
It is a stand alone fax machine.
Receiving faxes works OK, only when I try to send a fax it cuts it off.
I know the problem is setting on Linksys
Wednesday, March 4, 2009, 3:22:59 PM, Joseph wrote:
FAX Passthru Codec: G711u
for me FAX works better with G711a
--
Best regards,
Gergomailto:csi...@gmail.com
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Hi,
There is a long call setup time untill the call connects. How can I play a beep
tone say every 4 seconds to the caller untill the call connects?
Tx.
Shaun___
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asterisk-users
On 4 Mar 2009, at 15:02, Shaun Wingrin wrote:
There is a long call setup time untill the call connects. How can I
play a beep tone say every 4 seconds to the caller untill the call
connects?
Surely its better to try and diagnose the long call setup time?
hi
you can use the r option to send ring sound to the caller until the call is
answered or you can use m option to put music on hold where you can record a
4 sec length audio whit a beep.
check the dial coammand options.
David
2009/3/4 Shaun Wingrin voi...@gmail.com
Hi,
There is a long call
Yes. We have a number of customers in CR connecting to E1 PRIs using the
Redfone fonebridge and it works fine.
Are you having a particular issue or just looking for general confirmation that
Asterisk and E1 in Costa Works?
Good luck.
- Original Message -
From: Luis Morales
Kevin P. Fleming wrote:
Steve Underwood wrote:
CAT-iq supports G.722 for wideband voice. the CAT-iq web site describes
this as CD quality. I guess the person who wrote that has severely
impaired hearing. :-)
Maybe they meant 'CD quality after compression with MPEG layer 3 to 128
Ok guys I have to jump in here.
Seems some of you took affront to my $20 paypal bounty.
Not sure how long some of you have been around here but the history
behind the Weather app on the Trixbox (or Asterisk @ home as it used to
be known back then) was that my wife used to always ask me what the
Hello
With 1.4.23.1, I can't really see any difference between setting this value
to yes or no.
Can you explain ?
Regards
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Olivier wrote:
Hello
With 1.4.23.1, I can't really see any difference between setting this
value to yes or no.
Can you explain ?
Regards
It seems that you're only going to see a difference if you are using a phone
which subscribes to hints and uses the application/dialog-info+xml
Getting our own VOIP conference server going will be the 2nd part of the
ProgrammingParty until it is accomplished. - The first part will be
getting Ekiga ver 3 working on KUbuntu 8.04, whatever other OSs people
have.
Come help out on the BTIP conference server, if you like. :)
=
Since there is a PHP version of the *WA, it seems to me that that would
convert into tropo with very little effort. Just a matter of finding the
person with the means to try it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I saw some of the heat about the $20 bounty earlier. So I don't want to
put a low bounty out.
Quote me a bounty, and I'll see if I can get it approved by management. :-)
I'm in need of getting this bug fixed. Bug has all of the details, but
basically 1.4.22 broke it all.
I've waited as long
Robert Broyles wrote:
I saw some of the heat about the $20 bounty earlier. So I don't want to
put a low bounty out.
Quote me a bounty, and I'll see if I can get it approved by management. :-)
I'm in need of getting this bug fixed. Bug has all of the details, but
basically 1.4.22 broke
Yes, I've already posted notes on the bug.
I applied the patch, and when attempting to recompile, it fails.
--
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145
Jason Parker wrote:
Robert Broyles wrote:
I saw some of the
By the way, I'm more than happy to send murf a case of rootbeer (or real
beer assuming he's legal :-P ) if this bug and/or related bugs can be
resolved soon. :-)
--
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145
Jason
On Wed, Mar 4, 2009 at 6:24 PM, Robert Broyles rob...@poornam.com wrote:
By the way, I'm more than happy to send murf a case of rootbeer (or real
beer assuming he's legal :-P ) if this bug and/or related bugs can be
resolved soon. :-)
Bottle of Riga Black Balsam (45%), just have to figure out
On 03/04/09 15:56, Gergo Csibra wrote:
Wednesday, March 4, 2009, 3:22:59 PM, Joseph wrote:
FAX Passthru Codec: G711u
for me FAX works better with G711a
Can you folks compare my setting below with your settings and let me know if
something differ.
I was experimenting with echo in the past and
On 03/04/09 15:44, Marco Signorini wrote:
Hi Joseph.
I've spent some time tuning the SPA3102 FXS line input and output gain
and I think that this is an important variable.
Let's try to record incoming and outgoing fax tones with asterisk on SIP
channel (disabling the fax detection on SPA and
Joseph wrote:
On 03/04/09 15:44, Marco Signorini wrote:
Hi Joseph.
I've spent some time tuning the SPA3102 FXS line input and output gain
and I think that this is an important variable.
Let's try to record incoming and outgoing fax tones with asterisk on SIP
channel (disabling the fax
2009/3/4 Atis Lezdins a...@iq-labs.net
Bottle of Riga Black Balsam (45%), just have to figure out a way to send it
:)
Balsam??? By mail? Doesn't that count as liquid explosive? ;-)
Chris
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Actually, that's alcohol abuse. :-)
Regards,
Robert Broyles
Christian Victor wrote:
2009/3/4 Atis Lezdins a...@iq-labs.net mailto:a...@iq-labs.net
Bottle of Riga Black Balsam (45%), just have to figure out a way
to send it :)
Balsam??? By mail? Doesn't that count as liquid
On Wednesday 04 March 2009 10:24:16 Robert Broyles wrote:
By the way, I'm more than happy to send murf a case of rootbeer (or real
beer assuming he's legal :-P ) if this bug and/or related bugs can be
resolved soon. :-)
Murf is plenty legal; he simply doesn't consume alcohol.
--
Tilghman
I have been receiving a lot of hack attempts today (home and work)
multiple SIP registration requests (none of them managed to find a
relevant username before fail2ban kicked in).
Is this happening to a lot of people now?
I only have SIP available externally for enum purposes, is it possible
Hey, all. I was just wondering if there were any
tools/utilities/what-have-you out there that would allow a user to click
on a contact in Outlook, and have their phone dial it? (Or, I guess, have
Asterisk dial both their phone and the destination number, and put the two
into a conference.)
On Wednesday 04 March 2009 11:34:23 Thomas Kenyon wrote:
I have been receiving a lot of hack attempts today (home and work)
multiple SIP registration requests (none of them managed to find a
relevant username before fail2ban kicked in).
Is this happening to a lot of people now?
I only have
http://outcall.sourceforge.net/
--
Godson Gera
Asterisk Consultant
Indiahttp://godson.in/voip-asterisk-consultant-hyderabad-india
On Wed, Mar 4, 2009 at 11:12 PM, Ken D'Ambrosio k...@jots.org wrote:
Hey, all. I was just wondering if there were any
tools/utilities/what-have-you out there
On 03/04/09 18:01, Marco Signorini wrote:
Joseph wrote:
As I remember I have experimented with gain on PSTN line as well but have
reset back to default.
I have:
SPA To PSTN Gain:0
PSTN To SPA Gain:0
I think 0 is the default.
Yes, 0 is the default.
Is the fax machine connected to the
Tilghman Lesher wrote:
On Wednesday 04 March 2009 10:24:16 Robert Broyles wrote:
By the way, I'm more than happy to send murf a case of rootbeer (or real
beer assuming he's legal :-P ) if this bug and/or related bugs can be
resolved soon. :-)
Murf is plenty legal; he simply doesn't consume
Marco Signorini wrote:
Joseph wrote:
On 03/04/09 15:44, Marco Signorini wrote:
Hi Joseph.
I've spent some time tuning the SPA3102 FXS line input and output gain
and I think that this is an important variable.
Let's try to record incoming and outgoing fax tones with asterisk on
Yea, that patch was tried, and doesn't resolve the issue either.
I will hold out on the bounty a little longer... maybe it will be
resolved soon. It's pretty important for us.
--
Regards,
Robert Broyles
Jason Parker wrote:
Tilghman Lesher wrote:
On Wednesday 04 March 2009 10:24:16
You want ADA which is the new name for the old snapanumber
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Tilghman Lesher wrote:
Yes, you can use the permit/deny labels to specify an IP mask that is eligible
to authenticate:
deny=0.0.0.0/0
permit=192.168.0.0/16
permit=172.16.0.0/12
permit=10.0.0.0/8
By the way, after the slash, you can use either CIDR notation or a netmask.
Thanks.
Courier mail server at exa.billmerriam.com schrieb:
This is a delivery status notification from exa.billmerriam.com,
running the Courier mail server, version 0.54.1.
The original message was received on Wed, 04 Mar 2009 09:10:55 -0500
from localhost (localhost [127.0.0.1])
Philipp Kempgen wrote:
Courier mail server at exa.billmerriam.com schrieb:
This is a delivery status notification from exa.billmerriam.com,
running the Courier mail server, version 0.54.1.
The original message was received on Wed, 04 Mar 2009 09:10:55 -0500
from localhost (localhost
Robert Broyles wrote:
I saw some of the heat about the $20 bounty earlier. So I don't want to
put a low bounty out.
Quote me a bounty, and I'll see if I can get it approved by management. :-)
I'm in need of getting this bug fixed. Bug has all of the details, but
basically 1.4.22 broke
I just want to confirm but it seems that if-then-else is not permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
No, if-then-else works fine inside a case statement. See inline
comments.
switch(${DIALSTATUS})
{
case
Can I assume that you want this only for blind transfers?
I have done this previously, but I lost my copy of the work (and it was
a proof of concept only)
It involved the ${BLINDTRANSFER} variable, which catches the number that
made the blind transfer and making macro-stdexten (or your
Thanks guys.
It was the If vs if that was causing the problem.
This is probably due to my good coding practice of other languages in
the past :-)
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Watkins,
I tried upgrading to 1.6.0.6 but when i compile and install that, it seems
that support for SIP is missing completely?
Reverting back to 1.6.0.5 gets SIP going again...
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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.
I'm not sure that this isn't stretching things a bit. There are no handsets or
headsets (AFAIK) that
Hi guys,
Has anyone had any luck with getting the Cisco IP Communicator working with
your Asterisk or primarily, Trixbox installation?
I've tried searching the net for information, and found someone said to set it
up like the 7970 hard phone, which I have, and I'm just running into the
Howdy,
I have the following issue and would like to know if anyone has got around this
before.
IP Phones - Linksys 942
Sip server - Asterisk 1.4.13
Stun server - Vovida
Ok heres the issue. We have multiple client phones on their own network behind
a natted connection. We have setup the
thanks very much for your reply,Grygoriy,you are so warm-heart! thank you
advance!
Here is my extensions.conf and meetme.conf. I don't use the digium card so I
just use the ztdummy modules.
[meetme]
exten = 4105,1,Answer()
exten = 4105,n,meetme(99008664105|Ap)
exten = 4105,n,Hangup()
meetm.conf
Have you tried using md5secret, not sure if that will do it - but that's how
we had to get our 7970 registered with freepbx/trixbox - unfortunately they
don' t have this ability built in (yet). I have a patch if you need it,
contact me off list. As a quick test you could enable it in the config
Outcall moved to http://code.google.com/p/outcall/
There's also Camrivox's Flexor (Snom and Asterisk versions).
Googling 'outlook click-to-call' will also show a bunch of related info
and tools.
Paul
Godson Gera wrote:
http://outcall.sourceforge.net/
--
Godson Gera
Asterisk Consultant
Hi List,
im running a test server with the 1.6.1-rc1-release of * and OpenAIS. Asterisk
is configured so far and running stable. Now i set up a second server to test
the distributed devstate. In a cluster on the same subnet it's no problem. But
we have a customer who wants that feature for
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