Address 0xfffe out of bounds why and how to
solve.MyConfbridgeCount(conferencenumber,variablename )return total number
of user in conference given by conferencenumber otherwise zero.At runtime
using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call
function
Amit,
I know how to play with SIP in asterisk and other tools . I want to know
weather asterisk natively support or is there any extra patch or any
workaround for SIP-T/SIP-I.
Regarding packets and other things I am still not integrating it . I am
searching some open-source tool which can send
Vladimir Mikhelson vlad at mikhelson.com writes:
Hi,
Here is the reply from the developer as to what can be done
immediately to remove the offending logging.
He can just ignore these messages, they say that chan_ooh323 don't
known indication signal 33
Hello,
I've got a small install with Asterisk 11.
This box is connected to PSTN through a SIP trunk.
I need to add a cellular phone as a remote agent of an existing queue.
At the moment, this queue is configured according a ringall strategy and
busy agent can be dialed.
My plan is :
1- to
Hi, I have asterisk-server version of 1.8.11-cert7.
When external enemy try to using it for calling, server write this string to
log:
2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Executing
[9810972592309759@from-sip-external:1] NoOp(SIP/external-ip-00065fd2,
Received incoming SIP
On Thu, Mar 13, 2014 at 4:18 AM, hkc323 hkc...@gmail.com wrote:
Address 0xfffe out of bounds why and how to
solve.MyConfbridgeCount(conferencenumber,variablename )return total number
of user in conference given by conferencenumber otherwise zero.At runtime
using
Thanks Steve.
I think my problem is NAT. I'm using iptables, but I don't sure if I'm
doing right steps.
In the principal router I've forwarded the ports, but in my firewall
(iptables on PBX server) I'm not sure. 201.237.180.154 is my remote place.
#El NAT para el 5060 y el 1-3 (rtp)
Guys, but ALL MY INCOMING CALLS (in remote extensions) WORKS FINE. Should
be a NAT issue?
On Thu, Mar 13, 2014 at 8:43 AM, alp...@gmail.com alp...@gmail.com wrote:
Thanks Steve.
I think my problem is NAT. I'm using iptables, but I don't sure if I'm
doing right steps.
In the principal
(If you want to reply to this message, this is not where your reply goes)
Please, for the benefit of anyone reading the archives in search of answers to
a question, when replying to messages on this list, can everyone try to follow
the natural flow of conversation? That is, position your
-1
Prefer top posting.
Easy to see if I want to scroll down to see if it is something
interesting to me.
I get a lot of e-mails each day and scrolling wastes too much time.
But if you have a solution to a problem that I raise, please feel free
to post it anywhere you like.
On 13/03/2014
On Thu, 13 Mar 2014, Ron Wheeler wrote:
-1
Prefer top posting.
Your preferences are in conflict with the mailing list rules
(http://www.asterisk.org/community/discuss), specifically #5.
It has to be all one way or the other. This is an English language list.
Thus, the natural expectation
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, March 13, 2014 1:39 PM
To: rwhee...@artifact-software.com; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only
CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I
did get back a name and a number and everything was displayed correctly. So I think the calling
site should
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] on behalf of Eric Wieling
[ewiel...@nyigc.com]
Sent: Thursday, March 13, 2014 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry
Brummell
Sent: Thursday, March 13, 2014 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
rwhee...@artifact-software.com
Subject: Re: [asterisk-users] Replying to
After each line of text, please also dip the corner of your keyboard into your
ink well to ensure your writing can been seen.
Calling something natural because it used to be that way isn't always correct.
-MD-
P.S. Notice how little we see PS in posts...now that we can also edit our own
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Thursday, March 13, 2014 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
rwhee...@artifact-software.com
Subject: Re:
On Thu, Mar 13, 2014 at 1:42 PM, jg webaccou...@jgoettgens.de wrote:
When I set CONNECTEDLINE() info for an incoming ISDN call, the calling
party sees only CONNECTEDLINE(num) and the name does not get displayed.
Some time ago I called a number, where I did get back a name and a number
and
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.
On 13/3/14 5:13 pm, Ron Wheeler wrote:
-1
Prefer top posting.
Easy to see if I want to scroll
AFAIK, It is some kind of character set code. That byte is intentionally put
there for
switches that are not Q.SIG or ETSI. Technically, the display IE is only to be
sent
from the network to the user. So sending it to the network is undefined and
switch
dependent.
Got it. The single
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.
Outlook can quote correctly, but it is an all or nothing setting it would
appear. Lotus
On Thu, Mar 13, 2014 at 9:24 AM, Rusty Newton rnew...@digium.com wrote:
On Thu, Mar 13, 2014 at 4:18 AM, hkc323 hkc...@gmail.com wrote:
Address 0xfffe out of bounds why and how to
solve.MyConfbridgeCount(conferencenumber,variablename )return total number
of user in conference given by
[Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register:
Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching
peer found
I tried hard bit can's find a solution or even a hint
ru
--
_
[Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register:
Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching
peer found
What does the cli command sip show peers show? Do you have a definition for the sip device
2000 in sip.conf?
jg
--
On Thu, Mar 13, 2014 at 7:58 AM, Игорь Гайсин igor.gaj...@tts.tv wrote:
snip
mysql select calldate,
clid,src,dst,dcontext,channel,lastapp,duration,billsec,disposition,amaflags
from cdr where calldate like '2014-03-13 09:56:04';
This mean that you either do not have such peer in your "sip.conf"
or "permit" option of the peer 2000 don not include IP address 2000.
On 03/13/2014 09:43 PM, q...@vienna.at
wrote:
[Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812
On 3/13/2014 11:33 AM, A J Stiles wrote:
If you need to make a point-by-point argument, split up your reply --
a critical piece to this component is proper quoting.
the person replying needs to differentiate between what he is writing
and what is is replying to. notice the in front of what
Hello everyone,
I would be extremely glad if someone could help me with the
following issue:
cat /etc/asterisk/func_odbc.conf
[call_user]
prefix=GET
dsn=asterisk_odbc_sip
readsql=SELECT name FROM asterisk_sippeers WHERE name = '%477'
Asterisk CLI
readsql=SELECT name FROM asterisk_sippeers WHERE name = '%477'
Not sure what database you are accessing, but have you tried the
following:
readsql=SELECT name FROM asterisk_sippeers WHERE name LIKE '%477'--
_
-- Bandwidth and
On Thu, Mar 13, 2014 at 08:49:32PM +0100, jg wrote:
[Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register:
Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching
peer found
What does the cli command sip show peers show?
Not a single 2000, no 2000 at
Sure , here is the reasult.
mysql SELECT name FROM asterisk_sippeers WHERE name LIKE '%477'
;
+-+
| name |
+-+
| Y_MD_vlungu_477 |
+-+
1 row in set (0.00 sec)
On 03/13/2014 11:17 PM, Kevin
On Thu, Mar 13, 2014 at 10:41:58PM +0200, Vadim Lungu wrote:
This mean that you either do not have such peer in your sip.conf
or permit option of the peer 2000 don not include IP address 2000.
I neither have a 2000 in sip,conf nor I want to have one.
2000 doesn't have an IP and I want to get
I neither have a 2000 in sip,conf nor I want to have one.
2000 doesn't have an IP and I want to get rid of it, honestly.
I'd really want to know, where this 2000 is burned in
and how to erase it.
sip show peers does'nt show a peer 2000 nor I have a user 2000.
Something that lives at
Sure , here is the reasult.
mysql SELECT name FROM asterisk_sippeers WHERE name LIKE '%477' ;
+-+
| name|
+-+
| Y_MD_vlungu_477 |
+-+
1 row in set (0.00 sec)
What happens when you use that in your func_odbc.conf? Does your
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.
Kevin Larson sez:
Outlook can quote correctly, but it is an all or nothing setting it would
Don Kelly wrote:
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.
Kevin Larson sez:
Outlook can quote correctly, but it is an all or
On 13/03/2014 9:32 PM, Don Kelly wrote:
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.
Kevin Larson sez:
Outlook can quote correctly, but
Vadim Lungu
try this one .
readsql=SELECT name FROM asterisk_sippeers WHERE name LIKE '%477';
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
I neither have a 2000 in sip,conf nor I want to have one.
2000 doesn't have an IP and I want to get rid of it, honestly.
I'd really want to know, where this 2000 is burned in
and how to erase it.
This is IP is brute forcing you with peer that don't exist. So just block it.
iptalble
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