Hi Jonas,
I get this error when I incorrectly set my PBX gateway AND I have a sip
peer trying to register outside (i.e.: a sip provider).
Are you sure about your sip.conf?
Giorgio Incantalupo
Jonas Kellens wrote:
> Hello,
>
> my Asterisk CLI is flooded with the following message :
>
> [Jun 25
Hey Guys
I have an indial range of 6128[01234]X being trunked sip to
xxx.yyy.189.65
Now I want to break this down into 61280x going to xxx.yyy.188.145 and
61284x going to xxx.yyy.189.199
reminder being used for fax->email etc etc etc
I have created the outbound routes and
Has anyone ever integrated the software from order logix into their system?
This is primarily an API driven, pulled from a SQL database and stored for a
client to access... Order Logix deals primarily with Call Centers, it pulls
the information from the SQL database, and will allow access for the c
On Tue, Jun 29, 2010 at 6:42 PM, Philipp von Klitzing
wrote:
> Hi!
>
>> Because the codec is already chosen before the call is made, and you
>> told it that g722 is permitted.
>>
>> There are all sorts of discussions in play about codec negotiation,
>> but at this point in time, if you want differ
I have an *ipphone 7965G* which has to be connected to Asterisk. It has been
flashed with SIP firmware but the config file doesn't seem to work maybe I
am missing something in it.
I appreciate it if you can share your working sample config file with me.
Thanks
--
Hi!
> Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??
Most probably - who on this list would you like to test it for you? ;->
Philipp
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hi!
> Because the codec is already chosen before the call is made, and you
> told it that g722 is permitted.
>
> There are all sorts of discussions in play about codec negotiation,
> but at this point in time, if you want different behaviour you'll need to
> go and code it yourself
Look at the l
Thanks, but I don't have any *dahdi*.conf file here... (I check in
/etc/asterisk)
Anahi Ludueña
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 29 Jun 2010 16:54:01 -0500
Subject: Re: [asterisk-users] Dial options not working
Check your DTMF sett
Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi files
this lives in). Sounds like your DAHDI doesnt like DTMF input.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Tuesday, June 2
Thank you Andrew,
I will check it out. We are currently running 1.4.
-Matt
On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham wrote:
> Remote Party ID in trunk, it works There are hacks for other versions.
>
>
> ~
> Andrew "lathama" Latham
> lath...@gmail.com
>
> * Learn more about OSS http:/
Hi, I have an extension which has the follow me option activated. The followme
option should go to a IVR if no answer...
The problem that I have is that everything works when I'm calling it from my
extension, but if I use any landline phone or a cell phone, I'm unable to enter
any options. When
On Tue, Jun 29, 2010 at 2:36 PM, Myles Wakeham wrote:
> What I'm looking for is some sort of advice on 'best practice' to handle
> call routing to my cell phone vs. keeping it on my LAN to my desktop
> phone. As I mentioned, the desktop phone works flawlessly and I'm
> trying to get the same resu
On Tue, Jun 29, 2010 at 10:06 AM, William Stillwell (Lists)
wrote:
> Any idea what could be causing this?
>
Yes, network delay, packet loss, the Internet. Implement QoS and
bandwidth monitoring.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelang
It depends on the issue. If you have a carrier that has say 5-10 different
routes, they may want to confirm that the issue occurs on the same route
every time, or see if it is hitting the same box on their end.
Theoretically they could gather all the info on their end given the
caller/called numbe
Hi,
I managed to get a remote extension to work through a router which can now
call all the other local extensions in asterisk. For some reason, nobody
can call me back. They get failed upon trying. Keep thinking there must be
some caller group to which I need be added. Or perhaps I need to ad
On Tue, Jun 29, 2010 at 12:51 PM, Kenny Watson
wrote:
> Is it simply a case of converting the prompts into other codecs and asterisk
> will pick these up?
>
Yes, install both g729 and ulaw/alaw prompts to avoid trans-coding altogether.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul
Ok list users, this is a question born out of curiosity, but if I'm
having an intermittent problem and the carrier wants some examples of
calls where the problem happened, what can they actually do with that
information?
I guess my implementation is relatively simple here and all I've got to
l
Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??
I have reported a codec-issue, but there is no solution. Will this patch
also answer my question ??
https://issues.asterisk.org/view.php?id=17020
Jonas.
On 06/29/2010 09:42 PM, Mindaugas Kezys wrote:
Try this: http://www.b2bua.org/w
We are experiencing intermittent DTMF problems here, with the following
setup:
ITSP -> PIX -> Asterisk (g729, RFC2833 for DTMF).
I am running Ubuntu server 10.04, but Asterisk is compiled by us and not
installed from the software repository. Essentially, DTMF works for some
time, but at some poi
The Asterisk Development Team has announced the release of version 1.4.11.3 of
libpri. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/
This release fixes a regression in the calling number assignment logic:
* Calling Number assignment logi
>From what I have seen if your sip provider does not take g722 then you will
have problems with outgoing calls. When I tried this, the same way you did,
I could make calles externally but had no audio each way reguardless of what
I tried to pass to the sip provider. Best bet is to use what your sip
I currently have 4 lines coming into the house. We currently have an Avaya
standard analog key system which has served us well, but running extensions is
a major pain and requires a dedicated run per extension. I have ethernet run
throughout the house though.
The first two lines are "home" l
Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists
I have a small Asterisk 1.4.2 system that I run out of my home based
business, and my Dialplan has it set to send any incoming call to my
desk Grandstream phone. Works really well.
When I leave the office, I need to re-direct the calls to my cell phone.
I tried to do this through my Grandstr
I'll try it out tomorrow.
Youre my hero of the day!
Regards,
Remco
Op 29 jun. 2010 om 17:45 heeft Zeeshan Zakaria het
volgende geschreven:
> Just put exten => _,s, before the MYSQL ...
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
>> On 2010-06-29 11:41 AM, "Remco Bressers" wrot
Thanks again.
But it was a question pending. It's possible AMI show failure resgisters and
wrong password? Because I already have a Java program for AMI and a few
lines of modification would solve my problem if asterisk sends the
information to the AMI.
Thanks,
Rodrigo Lang.
2010/6/29 Andrew
Please start here http://www.spamhaus.org/drop/ with your BGP
routes Then move up to log parsing.
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more ab
If I didn't have fail2ban, I would have way over 20k of these entries in my
asterisk log.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-29 1:36 PM, "Rodrigo Lang" wrote:
Good afternoon.
Thanks to everyone for answers. What I find strange is the asterisk does not
have any native tool for
Good afternoon.
Thanks to everyone for answers. What I find strange is the asterisk does not
have any native tool for him to SIP server security. Here's an example of
the syslog messages from asterisk:
[Jun 15 03:05:46] NOTICE [25284] chan_sip.c: Registration from '"213"
' failed for '116 .124.12
On Tue, 2010-06-29 at 10:04 +0800, Zhang Shukun wrote:
> hi, list
> i want to know what is the best OS for install Asterisk 1.6.2.9,
> which should work properly on working system.
>
> i want to use CentOS5.2 or CentOS 5.4. Which is better and stable?
> Thanks for your help.
>
>
Th
On 26 June 2010 22:08, Ryan Wagoner wrote:
> I have Polycom phones that support the g722 codec. Adding allow=g722
> to the [general] section of sip.conf works great and I can make calls
> between the phones using g722. However Asterisk is negotiating g722
> for calls going out my voip provider and
Hi, I am running asterisk 1.6.1.6 with a howler screamer card.
I have g729 and alaw trunks from a pbx /sip providers.
The howler screamer will only transcode from g729 to alaw / ulaw but my voice
prompts are in SLIN and throws errors when i try and access these applications.
Is it simply
Yep, I saw that and it's just not the case. I was having it dial my desk
extension, which was decidedly not busy at the time...
On 6/28/10 5:30 PM, "Philipp von Klitzing"
wrote:
>> Well, I¹ve tried this, and something just isn¹t right.
>
> Look here:
>
>> Event: Hangup
>> Channel: SIP/ShoreT
Just put exten => _,s, before the MYSQL ...
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-29 11:41 AM, "Remco Bressers" wrote:
Thanks Zeeshan, but i don't use (and understand) AEL :)
Any regular examples out there? :)
regards,
Remco
On 06/29/2010 05:27 PM, Zeeshan Zakaria wrote:
>
Thanks Zeeshan, but i don't use (and understand) AEL :)
Any regular examples out there? :)
regards,
Remco
On 06/29/2010 05:27 PM, Zeeshan Zakaria wrote:
> Let me make it simple for you:
>
> Add a column to your table, e.g. `my column`.
>
> In the dialplan do the following (AEL example):
>
>
Let me make it simple for you:
Add a column to your table, e.g. `my column`.
In the dialplan do the following (AEL example):
MYSQL(Connect connid localhost username password database);
MYSQL(Query resultid ${connid} INSERT INTO `cdr` (`mycolumn`)
VALUES('${SIPCHANINFO(ip)
Hi, you can use fail2ban
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
Which works well, when a pattern is found in a log file it addes in an iptables
rules to block the traffic for a period.
On debian you can apt-get install fail2ban and on centos/redhat yum -i fail2
Hi!
> Do you already have script to capture user's IP address? If not, check
> it here how I am capturing it:
>
> http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-
> within-the-dialplan
Or simply use one fo these:
${SIPCHANINFO(peerip)}
${SIPCHANINFO(recvip)}
${SIPCHAN
Actually putty does it all. I don't know which putty you are using, maybe
try downloading it again and explore its settings.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-29 11:12 AM, "Roderick A. Anderson"
wrote:
On 06/29/2010 06:53 AM, bruce bruce wrote:
> Hi Everyone,
>
> I am accusto
Rodrigo Lang wrote:
> Hello list.
>
> I'm trying to find a way to block any ip that tries to login more than
> three times with the wrong password and try to log in three different
> extensions. For I have suffered some brute force attacks on my asterisk
> in the morning period.
>
> The idea w
I use PUTTY 0.58 and have Window title and scroll control for 20K+ lines.
It could use some improvements, but it is more than adequate for "green
screen" control. The quality of Putty and many other applications depends
on how you choose to control it.
-Original Message-
From: asterisk-us
On 06/29/2010 06:53 AM, bruce bruce wrote:
> Hi Everyone,
>
> I am accustomed to PUTTY and it's very nice as in it allows many many
> SSH profiles to be saved and allows tunneling etcbut it's not very
> good when it comes to scrolling up and down, colors, text size, and
> specially it doesn't g
There are some good suggestions here as a starting point:
http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/
Rgds,
mcr
On 29 June 2010 15:39, Rodrigo Lang wrote:
> Hello list.
>
> I'm trying to find a way to block any ip that tries to login more than
> three times with the wr
I use SecureCRT+FX , and use ansi graphics.
Putty is nice w/WinSCP as well.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Tuesday, June 29, 2010 10:17 AM
To: Asterisk Users Mailing List -
Hello list.
I'm trying to find a way to block any ip that tries to login more than three
times with the wrong password and try to log in three different extensions. For
I have suffered some brute force attacks on my asterisk in the morning
period.
The idea would be: Any ip with three attempts wit
I do not use windows on the desktop/laptop, but when I have to I use putty.
Darkbasic
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thur
Remco Bressers wrote:
> Hi,
>
> Sorry, but i forgot to notice that i am already using the 'userfield'
> column so that's not a possibility. Is there any way i can add the IP
> address to a custom MySQL field in CDR? With AGI possibly? The problem
> is, that the CDR entry is written in MySQL when t
I like putty too. There are many features included in this client, for
example an freindly interface to setup tunnels, X11, and another
features.
Take a look into putty website.
http://the.earth.li/~sgtatham/putty/0.60/htmldoc/
Regards,
On Tue, Jun 29, 2010 at 9:52 AM, Tzafrir Cohen wrote:
>
Hi,
Sorry, but i forgot to notice that i am already using the 'userfield'
column so that's not a possibility. Is there any way i can add the IP
address to a custom MySQL field in CDR? With AGI possibly? The problem
is, that the CDR entry is written in MySQL when the call is hungup, so i
have no po
On Tue, Jun 29, 2010 at 09:53:42AM -0400, bruce bruce wrote:
> Hi Everyone,
>
> I am accustomed to PUTTY and it's very nice as in it allows many many SSH
> profiles to be saved and allows tunneling etcbut it's not very good when
> it comes to scrolling up and down, colors, text size, and speci
bruce bruce wrote:
> Hi Everyone,
>
> I am accustomed to PUTTY and it's very nice as in it allows many many
> SSH profiles to be saved and allows tunneling etcbut it's not very
> good when it comes to scrolling up and down, colors, text size, and
> specially it doesn't give a title to the o
Remco Bressers wrote:
> Hi,
>
> The subject says it all. Is it possible to put the IP address of the
> peer in the CDR records? Using AGI maybe?
>
Yes you can either put the information in the userfield if you are using
a plain text file.
If you are storing to a mysql table for example then yo
Ubuntu is not Debian.
I would recommend Debian tho, its rock solid and it jsut works (for
anything)
On 29 June 2010 12:29, Paul Belanger wrote:
> On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun wrote:
> > i want to know what is the best OS for install Asterisk 1.6.2.9,
> > which should work
I have several remote phones that experience a slight "call" delay when
answering phones, ie, they will answer, speak a few words, and then the
remote caller will hear them, and the first half is cutoff?
Any idea what could be causing this?
Thanks,
Bill.
--
_
AFAIK, this will only address successful authentications. I think the OP
wanted to be able to know what the user had entered on failed attempts.
Since I've added another layer to this onion, I think the best option is to
use Read followed by an AGI if you have bells and whistles like this and a
Go
Due to this reason I am doing authentications using Read().
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-29 9:36 AM, "Coco Richard" wrote:
Danny, Doug
thx for the replies. According to the documentation, there is no
change for Authenticate() in version 1.6.x.x. So it seems i have to
us
Hi!
> i need to save into a local variable the user's input dialed during
> the cmd Authenticate(). Is there a way to do it?
Use option a of Authenticate together with ${CDR(accountcode)}
Philipp
--
_
-- Bandwidth and Colocat
Hi,
There is usually an empty column in the cdr table named 'userfield'. You can
also add a column of your own. Then in the dialplan use:
Set(CDR(userfield)="user IP address")
And this will automatically add this information into the userfield column.
Do you already have script to capture user'
Hi Everyone,
I am accustomed to PUTTY and it's very nice as in it allows many many SSH
profiles to be saved and allows tunneling etcbut it's not very good when
it comes to scrolling up and down, colors, text size, and specially it
doesn't give a title to the opened instance. Maybe giving the I
Danny, Doug
thx for the replies. According to the documentation, there is no
change for Authenticate() in version 1.6.x.x. So it seems i have to
use Read().
rich
On Tue, Jun 29, 2010 at 3:26 PM, Doug Lytle wrote:
> Coco Richard wrote:
>> Hi,
>>
>> i need to save into a local variable the user's
Coco Richard wrote:
> Hi,
>
> i need to save into a local variable the user's input dialed during
> the cmd Authenticate(). Is there a way to do it?
>
>
core show application authenticate
hylafax*CLI>
-= Info about application 'Authenticate' =-
[Synopsis]
Authenticate a user
Options:
Depend of your hardware. For example if you plan use 8G or more in RAM
it's better choice 64Bits distro. There are others benefits for
example the size on databases, logs files, memory use, recording
files, etc.
Regards,
On Tue, Jun 29, 2010 at 8:38 AM, Steve Underwood wrote:
> On 06/29/2010
I believe that the information keyed is just trashed after authentication.
You could modify app_authenticate.c to set a variable with the passed
information (this might already be included in the 1.6/1.8 branches, I just
deal with 1.4). Your other option would be to use a read/gotoif pair in
place
Hello,
Just thought you might like to know that our popular call center
management and statistics package, OrderlyStats SE, has just got a new
release.
Version 1.6.2l includes a several configuration changes and enhancements
to provide seamless call integration with the popular Elastix
distri
On 06/29/2010 05:35 PM, Gareth Blades wrote:
> Zhang Shukun wrote:
>
>> hi, all
>> after a long time development, i need to deploy a production system.
>>
>> i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused
>> me.
>>
>> my computer hardware support 64 bit O
Hi,
i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?
thx
rich
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
=
| '-'
| "*"
It's either a range of days, e.g. 29-30, or * for don't care. So do 29-30.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, June 29, 2010 7:12 AM
To
Hello list,
why is it that GoToIfTime thinks a date of **|*|29-*|jun *is not valid ??
[Jun 29 14:06:34] -- Executing [...@macro-vac:10]
*GotoIfTime*("SIP/testcorp-0036", "**|*|29-*|jun*?onvac") in new stack
[Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day
'*', as
Simply set it to costume field of cdrs in dialplan and you will have
it a part of native cdr
Regards,
*Faisal Hanif*
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a liv
On Tue, 29 Jun 2010, Kiss András wrote:
2010/6/25 Remco Bressers :
On 06/25/2010 09:48 AM, Kiss András wrote:
You selected 5, G.729 Codec
Please enter your Key-ID: G729-10D2X----X
This product key cannot be registered! Please verify you entered the
correct product
Hi,
The subject says it all. Is it possible to put the IP address of the
peer in the CDR records? Using AGI maybe?
--
Kind regards,
Signet bv
Remco Bressers
T 040 - 707 4 907
F 040 - 707 4 909
E rbress...@signet.nl
--
_
--
Hi,
If you use curl realtime for registrations you can add useragnet check
in your CGI and also lot of else as well.
Regards,
*Faisal Hanif
*On 6/29/2010 4:48 PM, Tarek Sawah wrote:
well there are two restrictions.. the IP address of the station they
are using it .. and the UserAgent..
one t
well there are two restrictions.. the IP address of the station they are using
it .. and the UserAgent..one thing my agents hardly understand Computers .. and
their computer skills are limited to Microsoft Office products and
telemarketing. i'm not afraid of hackers or cracker .. security is no
Lets say you did everything as it was mentioned in the tutorial .. then go into
Asterisk console and issue the command:sip show peer A2BILLINGCREATEDUSER
if you can't find it.. then simply include additional_a2billing_sip.conf in
your sip.conf file.Regards
-- Tarek Sawah
Integrated Digital Sys
On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun wrote:
> i want to know what is the best OS for install Asterisk 1.6.2.9,
> which should work properly on working system.
>
Ubuntu 10.04 Server ?
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelan
On Tue, Jun 29, 2010 at 3:23 AM, DHAVAL INDRODIYA
wrote:
> is there anyway to resolve it out, Means if SIP wants to send each call to
> 192.168.1.30 , but without entry in /etc/hosts.
>
Setup a DNS server.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC
Zhang Shukun wrote:
> hi, list
> i want to know what is the best OS for install Asterisk 1.6.2.9,
> which should work properly on working system.
>
> i want to use CentOS5.2 or CentOS 5.4. Which is better and stable?
> Thanks for your help.
>
>
>
Somewhat of a religious argument.
CentOS
Zhang Shukun wrote:
> hi, all
> after a long time development, i need to deploy a production system.
>
> i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused
> me.
>
> my computer hardware support 64 bit OS.
>
> my question is : should i use Centos 5.4 64bit or
Zhang Shukun wrote:
> hi, list
> i want to know what is the best OS for install Asterisk 1.6.2.9,
> which should work properly on working system.
>
> i want to use CentOS5.2 or CentOS 5.4. Which is better and stable?
> Thanks for your help.
>
>
Whatever system you go for it should have a
hi, all
after a long time development, i need to deploy a production system.
i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused me.
my computer hardware support 64 bit OS.
my question is : should i use Centos 5.4 64bit or Centos 5.4 32bit?
which is better for
2010/6/25 Remco Bressers :
> On 06/25/2010 09:48 AM, Kiss András wrote:
>> You selected 5, G.729 Codec
>> Please enter your Key-ID: G729-10D2X----X
>> This product key cannot be registered! Please verify you entered the
>> correct product key.
>> Server response: 404 -
Hi All,
I am newbie in this asterisk and a2billing technology . i had successfully
installed asterisk in my server fedora -8 [server behind NAT/STUN]
i after installation i can able to create users and tested the call
features with X-Lite . the was working fine .
after i installed the A2Billing
On Tuesday 29 Jun 2010, Tarek Sawah wrote:
> . is it possible to
> force the agents (users) to use a certain UserAgent which is the one
> built-in our system? this way will prevent the agents we are restricting
> them to only be able to dial through the software which is already
> restricted t
Dear All,
I have Asterisk and Kamailio Configuration.
everything works fine, now the situation is like i have following Dial
pattern in Dialplan.
exten => s,n, Dial(SIP/1...@glbvoice.com,20,m)
now in my /etc/hosts i have following entry
192.168.1.30 glbvoice.com
then call get forwarded to kam
On Tue, 29 Jun 2010, Rustam Kovhaev wrote:
> Hi there,
>
> I would like to setup up my Asterisk to do this:
> receptionist answers the call, caller says he wants to leave a
> voicemail message for Ashleigh, receptionist transfers the call to
> Ashleigh's voicemail
>
> I guess It has something to d
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