[asterisk-users] Send the same message to list of users

[Apa Minerala]

Customer is delivering stuff over the ocean.

Time of delivery is between 1 month to 1.5 months.

So
customers need some sort of a tracking system ( hard to implement given
the conditions ) or he needs to let tjem know when the packages
arrived. 

Customers in Europe all have mobile phones, while
senders in North America rarely have them ( they have answering
machines, though ). Could someone point me in the right direction so
that my customer :

A ) Sends the same vocal message to all his customers

or 

B) Sends SMS to everybody in Europe who is expecting packages. 

Any idea would be highly appreciated.

Thank you,

Tudor



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Re: [asterisk-users] clever ways to "share" an extension between sip and fxs

[Leif Neland]

Ira skrev:
> At 07:06 AM 11/18/2009, you wrote:
>   
>> I know that I'm not looking for Dial(SIP/x&SIP/y) - as documented, this
>> handles nothing like what I'm looking for.
>> 
>
> It's not the answer you're looking for, but that feature is built 
> into a Aastra 480i-CT and I think a 57i-CT.
>
>   
Do you know if this phone can also connect to a dect headset ?
(which currently connects to a Siemens base, which also supports 
standard dect phones)


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[asterisk-users] Asterisk crashes : Failed to start PBX

[Neo Anderson]

Hello,

I am using Asterisk 1.4.24.1 version in production.
OS is Centos 5.3 64 bit & RAM is 8 GB.
I am facing crash in asterisk approx each 12 hour.
When it crashes I see  below linesin asterisk logs.
[Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread
[Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :(
I debugged asterisk source code in details &  I found that it happens because 
it can not allocate memory to create thread.

Another thing is, when I check coredump using gdb, it's not showing any debug 
symbols.

Would you please let me know how to prevent or resolve this?

Thanks in advance!!

--
Regards,
voipexpert


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[asterisk-users] Dahdi and Junghanns QuadBRI

[Olivier]

Hi,

I'm using a revision 6822-enabled Dahdi-Tools (see
https://issues.asterisk.org/view.php?id=13897) with a Junghanns QuadBRI.

1. Do I still need qozap driver ? If positive, how is it recommended to get
it ?
2. Which line should be included in /etc/dahdi/modules to have the
appropriate driver loaded ?
3. The process I'm planning to use is :
A- Hand edit /etc/dadhi/modules, /etc/dadhi/genconf_parameters and
/etc/asterisk/chan_dadhi.conf.
B- Use dahdi_genconf to generate /etc/dadhi/system.conf  and
/etc/asterisk/dadhi_channels.conf.
Any suggestion ?


Regards
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[asterisk-users] SIP Calls on Asterisk fails after 25000 calls

[A A ANEES-RJD876]

Hi,
 
I am trying to use asterisk open source version(asterisk-1.6.0.5) with
MySQL (using res_odbc)support for extensions and users list.
 
The call rate is 7 calls / second and each call stays for 120 seconds.
after making 25000 successful calls , calls started 
failing with following message on CLI.
 
[Nov 11 08:50:04] WARNING[2258]: app_dial.c:1502 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
[Nov 11 08:50:04] WARNING[2259]: app_dial.c:1502 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
 
 
Is there any configuration parameters I missed out ?  please provide
your valuable suggestions on the same.
 
Regards
Anees

 
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[asterisk-users] Gain

[David @ULC]

Anyway to Increase Volume gain in Asterisk ?

USING g729 codec.
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Re: [asterisk-users] Security Against brute force attack

[Rasmus Männa]

Hi All,

I must say that there are many ways to detect password attack cause this
information actually goes into logs and it's possible to analyze them.
Couple of hours thinking + day or 2 creating gives a really nice result.
Bad thing is that by the time someone will start guessing password with
dictionary attack or brute force (it doesn't matter) he already knows
what is the account name/ID.

All this leads me to question which is (from my point of view) a bit
more important. Is there any way to detect SIP/IAX account guessing
without actually dumping UDP flow ? I tried some _hacking_ tools and
these create only some logs in debug mode. Using debug is not always an
option cause in some cases it creates ~5MB log in a minute - such flow
is quite impossible to handle.

Does anyone have any experience catching account guessing attempts
automatically ? Any kind of ideas would be wonderful :)

thx a lot,
--
razu

On 11/18/2009 10:01 PM, Ioan Indreias wrote:
> Hello Xavier,
>
> Unfortunately we are not aware of any Asterisk configuration which
> will protect against of a brute force attack on SIP. 
>
> We use BFD - http://www.rfxn.com/projects/brute-force-detection/ .
>
> We have found first details here: http://engineertim.com/?cat=15 and
> we are currently maintaining 4 rules (SIP and IAX) . All of them could
> be downloaded from
> here: http://www.modulo.ro/Modulo/downloads/tools/tenora.bfd.tar.gz
>
> We have tried to document the installation of BFD on an Asterisk
> server
> here: 
> http://www.modulo.ro/Modulo/ro/Articole/Securitate_pentru_servere_Asterisk.html
>  (in
> Romanian)
>
>
> HTH,
> Ioan (Nini) Indreias
> www.modulo.ro 
>
>
> On Mon, Nov 16, 2009 at 7:24 PM, TDF  > wrote:
>
> fail2ban
>
> 
> http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
>
>
> 2009/11/16 Xavier Mesquida  >
>
> Has Asterisk any protection against brute force attack for SIP
> authentication?
> Something like a maximum login attempt limit
> Thanks
>
>
>
>
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Re: [asterisk-users] softphone/debug panel with BLF

[Leif Neland]

Philipp Kempgen skrev:
> Leif Neland schrieb:
>   
>> Mostly to debug/test BLF, is there a softphone or another app. which can 
>> subscribe to hints on Asterisk?
>> 
>
> X-Lite?
> http://www.counterpath.com/x-lite.html
>
>
> Philipp Kempgen
>   
It does not

subscribe to hints on Asterisk.

Leif



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Re: [asterisk-users] can't call through voip provider

[Michael Wyres]

To be perfectly complete, exactly which inbound ports to open will depend on 
the phones in use.  For example, a Cisco 7940 (using this example because I 
have one on my desk at the moment), the default ports from the config are:

voip_control_port : 5060
start_media_port : 16384
end_media_port : 32766

Meaning, you have to have 5060 open (obviously), and all the ports between the 
start and end media port.  Many phones will let you adjust where these 
boundaries lie, but some won't.  You'll need enough range to cover every kind 
of phone (soft or hard) that you are using.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Thursday, 19 November 2009 09:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] can't call through voip provider


Ok. I do NOT have ports 1-2 opened in. I guess I should try that and 
see if it works.

I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I 
will keep you posted.

Thanks. 
--- On Wed, 11/18/09, Danny Nicholas  wrote:

> From: Danny Nicholas 
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> 
> Date: Wednesday, November 18, 2009, 5:18 PM
> According to what I know, you have to
> have 5060 open out and 1-2
> open in (you can cut this to as small as 1-10004).
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Landy Landy
> Sent: Wednesday, November 18, 2009 4:13 PM
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] can't call through voip
> provider
> 
> According to the provider he says he doesn't see anything
> coming in on their
> side. I've had all ports FORWARD out to ACCEPT but,
> blocking incoming new
> connections. I thought when asterisk starts a communication
> with a remote
> server using an unprivate port to port 5060 theres already
> an ESTABLISHED
> communication. I don't know if I'm having problems with my
> firewall script
> or what but, since there isn't any new connections coming
> form outside I
> think I'm ok to accept only ESTABLISHED,RELATED coming in.
> 
> I don't know but, I'm stuck with this problem and don't
> know what else to
> do.
> 
> --- On Wed, 11/18/09, Warren Selby 
> wrote:
> 
> > From: Warren Selby 
> > Subject: Re: [asterisk-users] can't call through voip
> provider
> > To: "Asterisk Users Mailing List - Non-Commercial
> Discussion"
> 
> > Date: Wednesday, November 18, 2009, 5:03 PM
> > What does your provider see when you
> > attempt to call them?
> > 
> > 
> > 
> > Thanks,
> > --Warren Selby
> > 
> > On Nov 18, 2009, at 3:38 PM, Landy Landy  
> > 
> > wrote:
> > 
> > > Thanks for replying.
> > >
> > > But how come I'm able to use a softphone to
> place
> > calls from withing  
> > > the lan? I really dont get it. What ports should
> I
> > enable in the  
> > > INPUT chain?
> > >
> > >
> > >
> > > --- On Wed, 11/18/09, Jared Smith 
> > wrote:
> > >
> > >> From: Jared Smith 
> > >> Subject: Re: [asterisk-users] can't call
> through
> > voip provider
> > >> To: "Asterisk Users Mailing List -
> Non-Commercial
> > Discussion"  > 
> > >> >
> > >> Date: Wednesday, November 18, 2009, 9:28 AM
> > >> On Wed, 2009-11-18 at 06:01 -0800,
> > >> Landy Landy wrote:
> > >>> Please help me with this, I can find any
> > solution on
> > >> this pls help. Your help will be very
> appreciated.
> > Thanks.
> > >>
> > >> It appears that Asterisk keeps sending an
> SIP
> > INVITE
> > >> message to your
> > >> provider, but not getting any kind of
> > response.  After
> > >> a number of
> > >> attempts at re-transmitting the message,
> it's
> > giving up.
> > >>
> > >> You need to check your network configuration
> and
> > find out
> > >> why responses
> > >> from the provider aren't getting back to
> your
> > Asterisk
> > >> system.  This is
> > >> typically a problem with firewalls, either on
> the
> > Asterisk
> > >> system itself
> > >> or between Asterisk and your VoIP provider.
> > >>
> > >>
> > >>
> > >> -- 
> > >> Jared Smith
> > >> Training Manager
> > >> Digium, Inc.
> > >>
> > >>
> > >>
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> > >>
> > >> asterisk-users mailing list
> > >> To UNSUBSCRIBE or update options visit:
> > >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>
> > >
> > >
> > >
> > >
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Re: [asterisk-users] Problem with sounds DTMF's phone keys

[Danny Nicholas]

Here's an idea - let's say that 4,7 and 8 are "extensions" that you want to
have a valid action take place on exit from the conference; you already have
that set up in the dialplan.  Therefore, you just need to set up
1,2,3,5,6,9,0 and * to do something like either hangup or jump back to the
conference.  Voila! No DTMF and if you use the hangup option, they will only
do it once.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Villacís
Lasso
Sent: Wednesday, November 18, 2009 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with sounds DTMF's phone keys

Danny Nicholas escribió:
> Could it be your using option X when you have no extensions for the user
to
> exit to (therefore when they press dtmf instead of one and done, they just
> keep going?)
>
>  
>
>   _  
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Moises Silva
> Sent: Monday, November 16, 2009 2:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Problem with sounds DTMF's phone keys
>
>  
>
> On Mon, Nov 16, 2009 at 3:27 PM, Diana Lopez  wrote:
>
> Hello everybody,
>
> I need help, I have a problem with conferences in asterisk, when many
> people are in a conference sometimes there're users pressing phone keys
> and this action emits a sound (DTMF of the phone keys), so, I need to
> find the way of not listening this sound.. I'm using
> MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because
> users continue listening de DTMF's sounds...
>
> But the way, I'm using Asterisk 1.4.26.1
>
> Thanks a lot.
>
>  
>
> This is just a guess, but, I think that if any of the users is using
inband
> DTMF and its sip.conf is not enabled to detect inband dtmf, you may have
> this result. 
>
>  
>
> The 'F' options MUST work if all the peers are properly configured,
> otherwise is a bug. I order to asterisk to mute or ignore that DTMF it
must
> first properly detect it. Use the /etc/asterisk/logger.conf configuration
to
> display detected dtmf and make sure all users dtmf is being detected. 
>
>  
>
> If the dtmf is detected by asterisk and you use the F option in meetme and
> still you hear the dtmf, then is a bug, try to reproduce it with latest
1.4
> version, if you can reproduce it the bug must be filled in
> issues.asterisk.org.
>
>  
>   
On reading the responses to this thread, I think there is a big omission 
that needs to be mentioned. I am a co-worker of the original poster, and 
I need to point out that the issue of the DTMFs is happening on a 
DAHDI-only setup. To elaborate: the Asterisk server (with 
asterisk-1.4.26.1) has a set of ISDN cards dedicated to incoming calls 
that are directed to one of several conference rooms created with 
app_meetme. As far as we know, there are no SIP phones connected to this 
server. All calls come from the public PSTN system.

The problem with the DTMFs is that some (but not all) keys are used by 
our dialplan to implement some custom actions. When the conference 
members press keys that are unused in the dialplan, all of the other 
conference members hear the DTMF sounds. This is undesirable in this 
server, so we are looking for a way (if it exists) to prevent other 
conference members from hearing the DTMF sounds caused by any one 
conference member pressing the numeric keys, while still preserving the 
keypress functionality (switching to other conference rooms, other 
custom actions).

For this particular server, we cannot rely on cooperation from the 
conference members, as this is a public PBX (anyone is and must be 
allowed to dial in). Therefore, asking the conference members to "stop 
pressing keys at random" is not an (enforceable) option for us. The 
scenario we are trying to prevent is one where malicious users start 
pressing random keys for the sake of disturbing the conference and 
annoying other conference users.

I have the suspicion that in this particular setup, what we are asking 
is difficult (if not impossible) to implement, as these DTMFs are 
probably in-band and therefore require signal processing to remove. I 
would be more than happy to be proven wrong on this issue.

-- 
perl -e '$x=2.3;printf("%.0f + %.0f = %.0f\n",$x,$x,$x+$x);'


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Re: [asterisk-users] can't call through voip provider

[Landy Landy]

Ok. I do NOT have ports 1-2 opened in. I guess I should try that and 
see if it works.

I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I 
will keep you posted.

Thanks. 
--- On Wed, 11/18/09, Danny Nicholas  wrote:

> From: Danny Nicholas 
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> 
> Date: Wednesday, November 18, 2009, 5:18 PM
> According to what I know, you have to
> have 5060 open out and 1-2
> open in (you can cut this to as small as 1-10004).
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Landy Landy
> Sent: Wednesday, November 18, 2009 4:13 PM
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] can't call through voip
> provider
> 
> According to the provider he says he doesn't see anything
> coming in on their
> side. I've had all ports FORWARD out to ACCEPT but,
> blocking incoming new
> connections. I thought when asterisk starts a communication
> with a remote
> server using an unprivate port to port 5060 theres already
> an ESTABLISHED
> communication. I don't know if I'm having problems with my
> firewall script
> or what but, since there isn't any new connections coming
> form outside I
> think I'm ok to accept only ESTABLISHED,RELATED coming in.
> 
> I don't know but, I'm stuck with this problem and don't
> know what else to
> do.
> 
> --- On Wed, 11/18/09, Warren Selby 
> wrote:
> 
> > From: Warren Selby 
> > Subject: Re: [asterisk-users] can't call through voip
> provider
> > To: "Asterisk Users Mailing List - Non-Commercial
> Discussion"
> 
> > Date: Wednesday, November 18, 2009, 5:03 PM
> > What does your provider see when you
> > attempt to call them?
> > 
> > 
> > 
> > Thanks,
> > --Warren Selby
> > 
> > On Nov 18, 2009, at 3:38 PM, Landy Landy  
> > 
> > wrote:
> > 
> > > Thanks for replying.
> > >
> > > But how come I'm able to use a softphone to
> place
> > calls from withing  
> > > the lan? I really dont get it. What ports should
> I
> > enable in the  
> > > INPUT chain?
> > >
> > >
> > >
> > > --- On Wed, 11/18/09, Jared Smith 
> > wrote:
> > >
> > >> From: Jared Smith 
> > >> Subject: Re: [asterisk-users] can't call
> through
> > voip provider
> > >> To: "Asterisk Users Mailing List -
> Non-Commercial
> > Discussion"  > 
> > >> >
> > >> Date: Wednesday, November 18, 2009, 9:28 AM
> > >> On Wed, 2009-11-18 at 06:01 -0800,
> > >> Landy Landy wrote:
> > >>> Please help me with this, I can find any
> > solution on
> > >> this pls help. Your help will be very
> appreciated.
> > Thanks.
> > >>
> > >> It appears that Asterisk keeps sending an
> SIP
> > INVITE
> > >> message to your
> > >> provider, but not getting any kind of
> > response.  After
> > >> a number of
> > >> attempts at re-transmitting the message,
> it's
> > giving up.
> > >>
> > >> You need to check your network configuration
> and
> > find out
> > >> why responses
> > >> from the provider aren't getting back to
> your
> > Asterisk
> > >> system.  This is
> > >> typically a problem with firewalls, either on
> the
> > Asterisk
> > >> system itself
> > >> or between Asterisk and your VoIP provider.
> > >>
> > >>
> > >>
> > >> -- 
> > >> Jared Smith
> > >> Training Manager
> > >> Digium, Inc.
> > >>
> > >>
> > >>
> ___
> > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >>
> > >> asterisk-users mailing list
> > >> To UNSUBSCRIBE or update options visit:
> > >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>
> > >
> > >
> > >
> > >
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> > >
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> > > To UNSUBSCRIBE or update options visit:
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> > 
> 
> 
>   
> 
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> 


  

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Re: [asterisk-users] Problem with sounds DTMF's phone keys

[Alex Villací­s Lasso]

Danny Nicholas escribió:
> Could it be your using option X when you have no extensions for the user to
> exit to (therefore when they press dtmf instead of one and done, they just
> keep going?)
>
>  
>
>   _  
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Moises Silva
> Sent: Monday, November 16, 2009 2:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Problem with sounds DTMF's phone keys
>
>  
>
> On Mon, Nov 16, 2009 at 3:27 PM, Diana Lopez  wrote:
>
> Hello everybody,
>
> I need help, I have a problem with conferences in asterisk, when many
> people are in a conference sometimes there're users pressing phone keys
> and this action emits a sound (DTMF of the phone keys), so, I need to
> find the way of not listening this sound.. I'm using
> MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because
> users continue listening de DTMF's sounds...
>
> But the way, I'm using Asterisk 1.4.26.1
>
> Thanks a lot.
>
>  
>
> This is just a guess, but, I think that if any of the users is using inband
> DTMF and its sip.conf is not enabled to detect inband dtmf, you may have
> this result. 
>
>  
>
> The 'F' options MUST work if all the peers are properly configured,
> otherwise is a bug. I order to asterisk to mute or ignore that DTMF it must
> first properly detect it. Use the /etc/asterisk/logger.conf configuration to
> display detected dtmf and make sure all users dtmf is being detected. 
>
>  
>
> If the dtmf is detected by asterisk and you use the F option in meetme and
> still you hear the dtmf, then is a bug, try to reproduce it with latest 1.4
> version, if you can reproduce it the bug must be filled in
> issues.asterisk.org.
>
>  
>   
On reading the responses to this thread, I think there is a big omission 
that needs to be mentioned. I am a co-worker of the original poster, and 
I need to point out that the issue of the DTMFs is happening on a 
DAHDI-only setup. To elaborate: the Asterisk server (with 
asterisk-1.4.26.1) has a set of ISDN cards dedicated to incoming calls 
that are directed to one of several conference rooms created with 
app_meetme. As far as we know, there are no SIP phones connected to this 
server. All calls come from the public PSTN system.

The problem with the DTMFs is that some (but not all) keys are used by 
our dialplan to implement some custom actions. When the conference 
members press keys that are unused in the dialplan, all of the other 
conference members hear the DTMF sounds. This is undesirable in this 
server, so we are looking for a way (if it exists) to prevent other 
conference members from hearing the DTMF sounds caused by any one 
conference member pressing the numeric keys, while still preserving the 
keypress functionality (switching to other conference rooms, other 
custom actions).

For this particular server, we cannot rely on cooperation from the 
conference members, as this is a public PBX (anyone is and must be 
allowed to dial in). Therefore, asking the conference members to "stop 
pressing keys at random" is not an (enforceable) option for us. The 
scenario we are trying to prevent is one where malicious users start 
pressing random keys for the sake of disturbing the conference and 
annoying other conference users.

I have the suspicion that in this particular setup, what we are asking 
is difficult (if not impossible) to implement, as these DTMFs are 
probably in-band and therefore require signal processing to remove. I 
would be more than happy to be proven wrong on this issue.

-- 
perl -e '$x=2.3;printf("%.0f + %.0f = %.0f\n",$x,$x,$x+$x);'


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Re: [asterisk-users] can't call through voip provider

[Danny Nicholas]

According to what I know, you have to have 5060 open out and 1-2
open in (you can cut this to as small as 1-10004).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Wednesday, November 18, 2009 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] can't call through voip provider

According to the provider he says he doesn't see anything coming in on their
side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new
connections. I thought when asterisk starts a communication with a remote
server using an unprivate port to port 5060 theres already an ESTABLISHED
communication. I don't know if I'm having problems with my firewall script
or what but, since there isn't any new connections coming form outside I
think I'm ok to accept only ESTABLISHED,RELATED coming in.

I don't know but, I'm stuck with this problem and don't know what else to
do.

--- On Wed, 11/18/09, Warren Selby  wrote:

> From: Warren Selby 
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

> Date: Wednesday, November 18, 2009, 5:03 PM
> What does your provider see when you
> attempt to call them?
> 
> 
> 
> Thanks,
> --Warren Selby
> 
> On Nov 18, 2009, at 3:38 PM, Landy Landy  
> 
> wrote:
> 
> > Thanks for replying.
> >
> > But how come I'm able to use a softphone to place
> calls from withing  
> > the lan? I really dont get it. What ports should I
> enable in the  
> > INPUT chain?
> >
> >
> >
> > --- On Wed, 11/18/09, Jared Smith 
> wrote:
> >
> >> From: Jared Smith 
> >> Subject: Re: [asterisk-users] can't call through
> voip provider
> >> To: "Asterisk Users Mailing List - Non-Commercial
> Discussion"  
> >> >
> >> Date: Wednesday, November 18, 2009, 9:28 AM
> >> On Wed, 2009-11-18 at 06:01 -0800,
> >> Landy Landy wrote:
> >>> Please help me with this, I can find any
> solution on
> >> this pls help. Your help will be very appreciated.
> Thanks.
> >>
> >> It appears that Asterisk keeps sending an SIP
> INVITE
> >> message to your
> >> provider, but not getting any kind of
> response.  After
> >> a number of
> >> attempts at re-transmitting the message, it's
> giving up.
> >>
> >> You need to check your network configuration and
> find out
> >> why responses
> >> from the provider aren't getting back to your
> Asterisk
> >> system.  This is
> >> typically a problem with firewalls, either on the
> Asterisk
> >> system itself
> >> or between Asterisk and your VoIP provider.
> >>
> >>
> >>
> >> -- 
> >> Jared Smith
> >> Training Manager
> >> Digium, Inc.
> >>
> >>
> >> ___
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> >
> >
> > ___
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
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> 


  

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Re: [asterisk-users] can't call through voip provider

[Landy Landy]

According to the provider he says he doesn't see anything coming in on their 
side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new 
connections. I thought when asterisk starts a communication with a remote 
server using an unprivate port to port 5060 theres already an ESTABLISHED 
communication. I don't know if I'm having problems with my firewall script or 
what but, since there isn't any new connections coming form outside I think I'm 
ok to accept only ESTABLISHED,RELATED coming in.

I don't know but, I'm stuck with this problem and don't know what else to do.

--- On Wed, 11/18/09, Warren Selby  wrote:

> From: Warren Selby 
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Date: Wednesday, November 18, 2009, 5:03 PM
> What does your provider see when you
> attempt to call them?
> 
> 
> 
> Thanks,
> --Warren Selby
> 
> On Nov 18, 2009, at 3:38 PM, Landy Landy  
> 
> wrote:
> 
> > Thanks for replying.
> >
> > But how come I'm able to use a softphone to place
> calls from withing  
> > the lan? I really dont get it. What ports should I
> enable in the  
> > INPUT chain?
> >
> >
> >
> > --- On Wed, 11/18/09, Jared Smith 
> wrote:
> >
> >> From: Jared Smith 
> >> Subject: Re: [asterisk-users] can't call through
> voip provider
> >> To: "Asterisk Users Mailing List - Non-Commercial
> Discussion"  
> >> >
> >> Date: Wednesday, November 18, 2009, 9:28 AM
> >> On Wed, 2009-11-18 at 06:01 -0800,
> >> Landy Landy wrote:
> >>> Please help me with this, I can find any
> solution on
> >> this pls help. Your help will be very appreciated.
> Thanks.
> >>
> >> It appears that Asterisk keeps sending an SIP
> INVITE
> >> message to your
> >> provider, but not getting any kind of
> response.  After
> >> a number of
> >> attempts at re-transmitting the message, it's
> giving up.
> >>
> >> You need to check your network configuration and
> find out
> >> why responses
> >> from the provider aren't getting back to your
> Asterisk
> >> system.  This is
> >> typically a problem with firewalls, either on the
> Asterisk
> >> system itself
> >> or between Asterisk and your VoIP provider.
> >>
> >>
> >>
> >> -- 
> >> Jared Smith
> >> Training Manager
> >> Digium, Inc.
> >>
> >>
> >> ___
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
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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] can't call through voip provider

[Warren Selby]

What does your provider see when you attempt to call them?



Thanks,
--Warren Selby

On Nov 18, 2009, at 3:38 PM, Landy Landy   
wrote:

> Thanks for replying.
>
> But how come I'm able to use a softphone to place calls from withing  
> the lan? I really dont get it. What ports should I enable in the  
> INPUT chain?
>
>
>
> --- On Wed, 11/18/09, Jared Smith  wrote:
>
>> From: Jared Smith 
>> Subject: Re: [asterisk-users] can't call through voip provider
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> > >
>> Date: Wednesday, November 18, 2009, 9:28 AM
>> On Wed, 2009-11-18 at 06:01 -0800,
>> Landy Landy wrote:
>>> Please help me with this, I can find any solution on
>> this pls help. Your help will be very appreciated. Thanks.
>>
>> It appears that Asterisk keeps sending an SIP INVITE
>> message to your
>> provider, but not getting any kind of response.  After
>> a number of
>> attempts at re-transmitting the message, it's giving up.
>>
>> You need to check your network configuration and find out
>> why responses
>> from the provider aren't getting back to your Asterisk
>> system.  This is
>> typically a problem with firewalls, either on the Asterisk
>> system itself
>> or between Asterisk and your VoIP provider.
>>
>>
>>
>> -- 
>> Jared Smith
>> Training Manager
>> Digium, Inc.
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] can't call through voip provider

[Landy Landy]

Thanks for replying.

But how come I'm able to use a softphone to place calls from withing the lan? I 
really dont get it. What ports should I enable in the INPUT chain?



--- On Wed, 11/18/09, Jared Smith  wrote:

> From: Jared Smith 
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Date: Wednesday, November 18, 2009, 9:28 AM
> On Wed, 2009-11-18 at 06:01 -0800,
> Landy Landy wrote:
> > Please help me with this, I can find any solution on
> this pls help. Your help will be very appreciated. Thanks.
> 
> It appears that Asterisk keeps sending an SIP INVITE
> message to your
> provider, but not getting any kind of response.  After
> a number of
> attempts at re-transmitting the message, it's giving up.
> 
> You need to check your network configuration and find out
> why responses
> from the provider aren't getting back to your Asterisk
> system.  This is
> typically a problem with firewalls, either on the Asterisk
> system itself
> or between Asterisk and your VoIP provider.
> 
> 
> 
> -- 
> Jared Smith
> Training Manager
> Digium, Inc.
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


  

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[asterisk-users] Off Topic

[Gary Reuter]

Please forgive this off-topic post... I've been on this list since
2005 (over 45k messages in my archive) and this is obviously really
not something I normally do.
If you have a minute and are feeling generous, please visit
http://bailout.chipin.com/ and consider helping me out.
Sorry if I've offended or wasted your time, but believe me that you
don't feel as bad as I do.

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Re: [asterisk-users] Security Against brute force attack

[Ioan Indreias]

Hello Xavier,

Unfortunately we are not aware of any Asterisk configuration which will
protect against of a brute force attack on SIP.

We use BFD - http://www.rfxn.com/projects/brute-force-detection/ .

We have found first details here: http://engineertim.com/?cat=15 and we are
currently maintaining 4 rules (SIP and IAX) . All of them could be
downloaded from here:
http://www.modulo.ro/Modulo/downloads/tools/tenora.bfd.tar.gz

We have tried to document the installation of BFD on an Asterisk server
here:
http://www.modulo.ro/Modulo/ro/Articole/Securitate_pentru_servere_Asterisk.html
(in
Romanian)


HTH,
Ioan (Nini) Indreias
www.modulo.ro


On Mon, Nov 16, 2009 at 7:24 PM, TDF  wrote:

> fail2ban
>
>
> http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
>
>
> 2009/11/16 Xavier Mesquida 
>
>  Has Asterisk any protection against brute force attack for SIP
>> authentication?
>> Something like a maximum login attempt limit
>> Thanks
>>
>>
>>
>
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[asterisk-users] Problem install wctdm24xxxp [resolved]

[Sylvain MEYNELLY (NEWTEK)]

Title:  




Ok thank you for giving to me a direction

I copy all the driver from a working  server with same hardware and
paste into this machine.

Everything working

What I don't is why same distribution and same hardware give me this
problem.

Thank you very much for your help

Bye

-- 



  

 Sylvain
MEYNELLY 
NEWTEK 
BP 1496 Port-Gentil 
GABON 
Cell : +241 05 08 20 20 



  




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Re: [asterisk-users] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!

[Shaun Ruffell]

On 11/16/2009 09:42 AM, Ex Vito wrote:
>   Shaun,
>
>   Thanks for your feedback. See my inline comments.
>
>
> On Fri, Nov 13, 2009 at 7:18 PM, Shaun Ruffell  wrote:
>>
>> It appears there may be a regression in dahdi-linux 2.2.0 with regards to
>> the wcte12xp driver and the VPMADT032 module (as discussed
>> https://issues.asterisk.org/view.php?id=15724).  Would you be willing to try
>> at least revision 7584 of
>> http://svn.asterisk.org/svn/dahdi/linux/branches/2.2 and report your results
>> on that issue?
>>
>
>   We will, either on 15724 or on 15798, both, I'd say, closely related
>   to what we're experiencing.
>
>   First, however, we will test 2.1.0.4 for at least 10 days to
>   effectively confirm are experiencing the regression you refer. We
>   defined 10 stable days because we've been experiencing 1 - 3
>   failures per week with 2.2.0.2.
>
>   So, we will move to 2.1.0.4 later today and, if there are no incidents,
>   we will move to SVN revision 7584 or later on Nov 27th.
>
>
>>
>> The idle load you're seeing can be a little misleading, but essentially,
>> once you load the drivers for both the wctdm24xxp and wcte12xp, there is a
>> fixed cost associated with continuously moving the TDM data to/from the
>> card.  The load imposed by the drivers would only go up after this point if
>> a) software echocan is enabled, or b) you're conferencing many calls in the
>> kernel.  Otherwiseit's fixed.
>>
>
>   I agree the load can be misleading however, consider:
>
>   - The system is really idle - zero calls, zero activity, no other software.
>   - 0.3 to 0.4 load on a modern CPU is a lot of processing, is there
> that much data to move around ?! :)

yes :)  All the TDM data is still moved between the host and the card as 
if there were 24 calls up.  The driver doesn't make any distinction 
between calls being up or calls not being up unless you're using 
software echocan.

>
>   We tested some variations and here is what we found:
>   (recall, AEX410 with no VPM is physically installed)
>
>   DAHDI 2.2.0.2
>   - Removed TE121 - Idle load is 0
>   - TE121 without VPM - Idle load is 0.3 - 0.4
>   - TE121 with VPM - Idle load is 0.3 - 0.4
>
>   DAHDI 2.1.0.4
>   - TE121 without VPM - Idle load is 0.01 - 0.02
>   - TE121 with VPM - Idle load is 0.01 - 0.05
>
>   DAHDI 2.2.0.1
>   - TE121 with VPM - Idle load is 0.26 - 0.32
>
>   DAHDI 2.2.0
>   - TE121 with VPM - Idle load is 0.31 - 0.36
>
>   DAHDI SVN-branch-2.2-r7584
>   - TE121 with VPM - Idle load is 0.69 - 0.82
>
>   Under all cases, we stopped asterisk, unloaded DAHDI,
>   rebuilt DAHDI (+Asterisk if needed), loaded DAHDI, started
>   Asterisk, waited 120s, tested one inbound call + one outbound
>   call.
>
>   In short:
>
>   - DAHDI 2.1, Idle load is 0
>   - DAHDI 2.2, Idle load is 0.3 - 0.4
>   - DAHDI 2.2. SVN, Idle load is roughly twice as in 2.2 releases
>
>   So, while your explanation regarding system load makes sense,
>   I find it odd that 2.2, which apparently is not working ok for us
>   in this case, is showing such behaviour when 2.1 does not and
>   neither Zaptel 1.2/1.4 used many other systems I've installed.
>
>   What would you say the explanation for these observations is ?

The change from 2.1 to 2.2 I believe is a timing anomaly.  The work 
related to checking the alarm conditions from the framer on the TE122 
was moved from the interrupt handler and into a workqueue, and therefore 
the system is more accurately able to account for it's time.  Between 
2.2.0.2 and the current head of the 2.2 branch, the frequency that that 
timers are checked was doubled (from once every 200ms to once every 
100ms) and that is a likely candidate for the doubling of the load. 
I'll need to run my own tests to confirm this.

>
>   Would you say that there is a correlation between the load and
>   the suspected not so good behaviour we're getting from DAHDI
>   2.2.0.2 on our case ?
>
>   Or is it just a coincidence ?

My belief is that this is just a coincidence.  At least from the driver 
standpoint, if you're not receiving any "latency increases" or IRQ 
misses isn't constantly rolling up during normal operation, then the 
data is consistent coming off the card.  There can still be problems 
moving that data up to user space if there is something that is 
preventing asterisk from being scheduled in a timely enough fashion to 
keep the kernel buffers non-empty (for transmit) or non-full (for receive).

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Problem install wctdm24xxxp

[Shaun Ruffell]

On 11/18/2009 12:08 PM, Sylvain MEYNELLY (NEWTEK) wrote:
> I have the following error but can't find any response on google.
>
> Hope you can help
>
> what is failed with error -5
>
> wctdm24xxp0: BAR0 is not IO Memory.
> wctdm24xxp: probe of :00:08.0 failed with error -5

The "probe failed with error -5" means there was an IO error that 
prevented the driver from attaching to the physical device.  In your 
case, one of the memory regions advertised by the cards is not what the 
driver expects it to be.  I would contact Digium technical support for 
assistance.

Cheers,
Shaun
-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Problem install wctdm24xxxp

[Danny Nicholas]

It's a driver problem.  See this link.

http://lists.digium.com/pipermail/dahdi-commits/2009-October/001348.html

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sylvain
MEYNELLY (NEWTEK)
Sent: Wednesday, November 18, 2009 12:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem install wctdm24xxxp

 

Hi everybody

I have a big problem for installing my card

I have the following error but can't find any response on google.

Hope you can help

what is failed with error -5 

wctdm24xxp0: BAR0 is not IO Memory.
wctdm24xxp: probe of :00:08.0 failed with error -5   

System is debian lenny with zaptel driver from repository
I have tried 4 card with same error

Thank you 

best regards

-- 


Sylvain MEYNELLY 
NEWTEK 
BP 1496 Port-Gentil 
GABON 
Cell : +241 05 08 20 20 

newtek

 

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Re: [asterisk-users] clever ways to "share" an extension between sip and fxs

[Ira]

At 07:06 AM 11/18/2009, you wrote:
>I know that I'm not looking for Dial(SIP/x&SIP/y) - as documented, this
>handles nothing like what I'm looking for.

It's not the answer you're looking for, but that feature is built 
into a Aastra 480i-CT and I think a 57i-CT.

Ira 


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[asterisk-users] Problem install wctdm24xxxp

[Sylvain MEYNELLY (NEWTEK)]

Title:  




Hi everybody

I have a big problem for installing my card

I have the following error but can't find any response on google.

Hope you can help

what is failed with error -5 

wctdm24xxp0: BAR0 is not IO Memory.
wctdm24xxp: probe of :00:08.0 failed with error -5   

System is debian lenny with zaptel driver from repository
I have tried 4 card with same error

Thank you 

best regards

-- 



  

 Sylvain
MEYNELLY 
NEWTEK 
BP 1496 Port-Gentil 
GABON 
Cell : +241 05 08 20 20 



  




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Re: [asterisk-users] asterisk 1.4.26.3 makes kernel panic

[Vieri]

--- On Wed, 11/18/09, Robert Grignon  wrote:

> BTW, the last "log entry" I have in /var/log/asterisk/full

And the last log entries in /var/log/messages are:

Nov 18 10:08:20 voip2 asterisk[25572]: rc_avpair_new: unknown attribute 
1490026597
Nov 18 10:08:20 voip2 asterisk[25572]: rc_avpair_new: unknown attribute 
1490026597



  

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[asterisk-users] Asterisk 1.2.18 and meetme causing Audio bleeds

[Jon Thomas]

Lookin for anyone who has experienced an issue similar to this.  It's quite
baffling as I'm unable to locate much help when it comes to debugging such
an audio oddity.

I'm currently running Asterisk 1.2.18 with a T1/E1 PRI.

To cause the audio bleed (is audio bleed actually what I should even call
this?): I create a meetme conference and have a few people call in to it.
Once established I then use another phone and dial out a non-conference
extension.

As soon as Asterisk picks up the call for routing - the audio from the
meetme application suddenly broadcasts into my call.
The users of the conference can NOT hear me but I can hear everything
they're saying.  It's like Asterisk is treating my line as a call monitor -
though asterisk indicates it is NOT (via cli show channels verbose and
meetme list conf#)

Any thoughts as to possible areas to focus on for such an issue?  Asterisk
full logs indicate no errors or anomalies...and based on results from
google, this is not an issue that seems to be reported often, if ever.

Thanks!
Jon.
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[asterisk-users] AGI and paging

[Jeff LaCoursiere]

Hello,

I have an AGI (in C) on 1.4.26.3 that puts a caller on hold, does a few 
things, then blind transfers the call (with EXEC Dial...) to a parking 
space.  This is working fine.

Now I want to add an overhead page AFTER the transfer has happened, 
basically announcing that there is a caller waiting.  Trouble is the 
channel is gone, and my "EXEC Page..." is returning -1.

I had to trap SIGALRM to keep the channel teardown from killing the AGI 
program in the first place (though I tried DeadAGI first - it still gets a 
SIGALRM when the Dial is complete).

It seems that I should be able to do the "EXEC Page" as a new channel... 
why is it not allowing it?

Also open to any other implementation ideas.  The AGI has an AMI 
connection open and I suppose I could try something via AMI for the page?

Thanks for any clues!

j

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Re: [asterisk-users] asterisk 1.4.26.3 makes kernel panic

[Vieri]

--- On Wed, 11/18/09, Robert Grignon  wrote:

> What Hardware are you using?
> What OS are you running?
> If your getting a kernel panic you can install a
> crashkernel (kdump) and
> upon receiving a kernel panic it will reboot to a
> crashkernel, capture
> the crashinfo and safely reboot the system. You can then
> use the "crash"
> utility to analyse the information.
> 
> However, first thing I would try is to change the slot that
> your
> hardware card is plugged into... The kernel is complaining
> about an
> inturrupt issue and its possible its conflicting with
> something else... 

Hi,

My hardware as seen by lspci:

00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface 
Bridge (rev 01)
Subsystem: ASUSTeK Computer Inc. Unknown device 8179
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- 
Stepping- SERR- FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- 
SERR- TAbort- 
SERR- TAbort- 
SERR- TAbort- 
SERR- TAbort- 
SERR- TAbort- 
SERR- TAbort- 
SERR- TAbort- SERR- TAbort- SERR- TAbort- SERR- http://lse.sourceforge.net/kdump/ ?

I'll try to move my PCI cards to different slots but how can I tell which one 
is the culprit?
(I think I have just one free slot and I *need* all the telephony cards so I 
don't have much choice)

My PCI cards are (as seen with lspci):
1 4-port ISDN card
1 PRI single-span card
1 NIC

Any pointers?

Thanks!

Vieri

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Vieri
> Sent: Wednesday, November 18, 2009 4:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] asterisk 1.4.26.3 makes kernel
> panic
> 
> Hi,
> 
> I'm experiencing "frequent" kernel panics on a system with
> Asterisk
> 1.4.26.3.
> There is no core dump, "just" a kernel panic.
> This is the only data I could copy from the screen:
> 
> EIP: 0060: [] Tainted: P VLI
> EFLAGS: 00210297 (2.6.23-gentoo-r8 #1)
> eax: 0130 ebx:  ecx: 00220028 edx: 0978
> esi: 346e5802 edi:  ebp: c3b45500 esp: caf05bfc
> ds: 007b es: 007b fs: 00d8
> gs: 0033 ss: 0068
> Process asterisk (pid: 7631, ti: caf04000 task: f7987000
> task.ti:
> caf04000)
> Stack: ...
> Call Trace: ...
> Code: ...
> EIP: [] ss:esp 0068:caf05bfc
> Kernel panic - not syncing: Fatal exception in interrupt
> 
> As you can see, kernel is 2.6.23 but it's the same as on
> another server
> I have but with asterisk 1.2. Kernel 2.6.23 does not panic
> with asterisk
> 1.2. The hardware on both servers is identical (isdn cards,
> pri card,
> etc.) except for the motherboard.
> 
> How can I further diagnose the problem (besides, I think I
> can easily
> reproduce it as soon as I set it up as the "production"
> server)?
> What can be causing the kernel panic?
> 
> BTW, the last "log entry" I have in /var/log/asterisk/full
> is:
> [Nov 18 10:08:20] VERBOSE[6627]
> logger.c:   -- dialparties.agi: Checking
> CW and CFB status for extension 6169
> 
> (this system runs freepbx)
> 
> Help greatly appreciated.
> 
> Vieri
> 
> 
> 
>       
> 
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[asterisk-users] Bug CDR report - dst "s" ?

[Diana Lopez]

Hello everybody,

I have a question about value dst of cdr table in asteriskcdrdb,  so in
my db I see many registers with letter "s" in dst field, I found a
opinion that "s" is the default extension and the all calls enter the
system as "s" and are then changed as they pass through the dial plan,
so, if you have an IVR and they hang up during the IVR, the call will be
"s".   Is it true?

And if it's true there's a solution to the dst put a number or is normal
this case, I want to change "s" by a value.

Thanks a lot

-- 


Diana López



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Re: [asterisk-users] Queues without agent login

[Tarek Sawah]

Simply use 
member=SIP/Tarek
member=IAX2/JONAS
member=LOCAL/whatever

simple and good.. 
with member=SIP/extension  i'm facing a CALL WAITING issue.. the agent hears a 
callwaiting signal whenever the queue tries to call .. so i woul dsuggest using 
call-limit and busy limite with all your Agents

--
AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308






From: jonas.kell...@telenet.be
To: asterisk-users@lists.digium.com
Date: Wed, 18 Nov 2009 16:21:12 +0100
Subject: [asterisk-users] Queues without agent login






  
  


Is it possible to make use of queues for incoming calls but to have agents that 
do not need to log in ?



If I create a queue and make certain SIP-users member of the queue, do these 
SIP-users always need to log in to the queue to be able to receive calls that 
are in the queue ??



Can't a member be just available when the phone is registered to the 
Asterisk-server ? In stead of also having to call an extension to log in (and 
having to give some PIN).



I just want a queue (with MoH) to collect multiple incoming calls and then one 
at a time transfer them to an available SIP-phone.



Is this possible ?



Thanks you.



Jonas.
_
Windows 7: I wanted simpler, now it's simpler. I'm a rock star.
http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009___
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Re: [asterisk-users] Queues without agent login

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

jonas kellens wrote:
> Is it possible to make use of queues for incoming calls but to have
> agents that do not need to log in ?
> 

Make the phones members of the queue.  In queues.conf:


[MY_QUEUE]

member => SIP/1234
member => SIP/5678

etc.

Barry

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFLBBYACFu3bIiwtTARAg5oAKClAtJ98LaSXnjCDBx4xlRcLQ9l/wCgoeI+
BAi7wu2nQ6vNPZSaLCDB4DA=
=egpe
-END PGP SIGNATURE-

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[asterisk-users] Queues without agent login

[jonas kellens]

Is it possible to make use of queues for incoming calls but to have
agents that do not need to log in ?

If I create a queue and make certain SIP-users member of the queue, do
these SIP-users always need to log in to the queue to be able to receive
calls that are in the queue ??

Can't a member be just available when the phone is registered to the
Asterisk-server ? In stead of also having to call an extension to log in
(and having to give some PIN).

I just want a queue (with MoH) to collect multiple incoming calls and
then one at a time transfer them to an available SIP-phone.

Is this possible ?

Thanks you.

Jonas.
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[asterisk-users] clever ways to "share" an extension between sip and fxs

[Jeremy Kister]

Using Asterisk 1.6.1.9, I'm looking for a way to "share an extension" 
between a SIP phone (Cisco 7940) and a SLT on a FXS port of a Cisco 1760 
(via sip) -- at any given time I want to be able to pick up either phone 
and it should be "bridged" to the other - just like having two SLTs on the 
same copper pair.

The goal is to have a cheap cordless telephone sit right next to a SIP 
phone and have technology-ignorant folk use the SIP phone as their primary 
phone and pick up the cordless when they are on a call and want to roam. 
Asking these people to transfer to 700, then wait to hear what extension 
the call is parked on, then dial that extension on their other phone is too 
much to ask :D

I've been thinking about having each on a separate extension and both 
auto-joining a meetme bridge when going off-hook, but it seems rather 
infeasible and eliminates features like call parking/transferring for those 
who would use it.

I know that I'm not looking for Dial(SIP/x&SIP/y) - as documented, this 
handles nothing like what I'm looking for.

Ideas?

-- 

Jeremy Kister
http://jeremy.kister.net./

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Re: [asterisk-users] softphone/debug panel with BLF

[Philipp Kempgen]

Leif Neland schrieb:
> Mostly to debug/test BLF, is there a softphone or another app. which can 
> subscribe to hints on Asterisk?

X-Lite?
http://www.counterpath.com/x-lite.html


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] can't call through voip provider

[Jared Smith]

On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote:
> Please help me with this, I can find any solution on this pls help. Your help 
> will be very appreciated. Thanks.

It appears that Asterisk keeps sending an SIP INVITE message to your
provider, but not getting any kind of response.  After a number of
attempts at re-transmitting the message, it's giving up.

You need to check your network configuration and find out why responses
from the provider aren't getting back to your Asterisk system.  This is
typically a problem with firewalls, either on the Asterisk system itself
or between Asterisk and your VoIP provider.



-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] asterisk 1.4.26.3 makes kernel panic

[Robert Grignon]

What Hardware are you using?
What OS are you running?
If your getting a kernel panic you can install a crashkernel (kdump) and
upon receiving a kernel panic it will reboot to a crashkernel, capture
the crashinfo and safely reboot the system. You can then use the "crash"
utility to analyse the information.

However, first thing I would try is to change the slot that your
hardware card is plugged into... The kernel is complaining about an
inturrupt issue and its possible its conflicting with something else... 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Wednesday, November 18, 2009 4:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk 1.4.26.3 makes kernel panic

Hi,

I'm experiencing "frequent" kernel panics on a system with Asterisk
1.4.26.3.
There is no core dump, "just" a kernel panic.
This is the only data I could copy from the screen:

EIP: 0060: [] Tainted: P VLI
EFLAGS: 00210297 (2.6.23-gentoo-r8 #1)
eax: 0130 ebx:  ecx: 00220028 edx: 0978
esi: 346e5802 edi:  ebp: c3b45500 esp: caf05bfc
ds: 007b es: 007b fs: 00d8
gs: 0033 ss: 0068
Process asterisk (pid: 7631, ti: caf04000 task: f7987000 task.ti:
caf04000)
Stack: ...
Call Trace: ...
Code: ...
EIP: [] ss:esp 0068:caf05bfc
Kernel panic - not syncing: Fatal exception in interrupt

As you can see, kernel is 2.6.23 but it's the same as on another server
I have but with asterisk 1.2. Kernel 2.6.23 does not panic with asterisk
1.2. The hardware on both servers is identical (isdn cards, pri card,
etc.) except for the motherboard.

How can I further diagnose the problem (besides, I think I can easily
reproduce it as soon as I set it up as the "production" server)?
What can be causing the kernel panic?

BTW, the last "log entry" I have in /var/log/asterisk/full is:
[Nov 18 10:08:20] VERBOSE[6627] logger.c:   -- dialparties.agi: Checking
CW and CFB status for extension 6169

(this system runs freepbx)

Help greatly appreciated.

Vieri



  

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Re: [asterisk-users] can't call through voip provider

[Landy Landy]

Hello.

Please help me with this, I can find any solution on this pls help. Your help 
will be very appreciated. Thanks.

--- On Tue, 11/17/09, Landy Landy  wrote:

> From: Landy Landy 
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Date: Tuesday, November 17, 2009, 7:33 AM
> Thanks for replying.
> 
> Here is the output of sip set debug peer voipprovider:
> 
> -- Called 1829257x...@voipprovider
> Retransmitting #1 (NAT) to myextip:5060:
> INVITE sip:18292574...@myextip SIP/2.0
> Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad
> Max-Forwards: 70
> From: "102" ;tag=as78863882
> To: 
> Contact: 
> Call-ID: 2908dd00500059761cc66bd81553e...@190.80.152.7
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Date: Tue, 17 Nov 2009 12:28:48 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 473
> 
> v=0
> o=root 1332315330 1332315330 IN IP4 190.80.152.7
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 190.80.152.7
> t=0 0
> m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:112 AAL2-G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:10 L16/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---
> Retransmitting #2 (NAT) to myextip:5060:
> INVITE sip:1829257x...@myextip SIP/2.0
> Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad
> Max-Forwards: 70
> From: "102" ;tag=as78863882
> To: 
> Contact: 
> Call-ID: 2908dd00500059761cc66bd81553e...@myextip
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Date: Tue, 17 Nov 2009 12:28:48 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 473
> 
> v=0
> o=root 1332315330 1332315330 IN IP4 myextip
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 190.80.152.7
> t=0 0
> m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:112 AAL2-G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:10 L16/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---
> Retransmitting #3 (NAT) to myextip:5060:
> INVITE sip:1829257x...@myextip SIP/2.0
> Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad
> Max-Forwards: 70
> From: "102" ;tag=as78863882
> To: 
> Contact: 
> Call-ID: 2908dd00500059761cc66bd81553e...@myextip
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Date: Tue, 17 Nov 2009 12:28:48 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 473
> 
> v=0
> o=root 1332315330 1332315330 IN IP4 myextip
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 myextip
> t=0 0
> m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:112 AAL2-G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:10 L16/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> 
> Scheduling destruction of SIP dialog
> '2908dd00500059761cc66bd81553e...@myextip' in 32000 ms
> (Method: INVITE)
> 
> 
> 
> By looking at this trace I dont see my provider's ip
> address anywhere. I guess I'm doing something wrong in my
> conf.
> 
> 
> 
> --- On Mon, 11/16/09, Warren Selby 
> wrote:
> 
> > From: Warren Selby 
> > Subject: Re: [asterisk-users] can't call through voip
> provider
> > To: "Asterisk Users Mailing List - Non-Commercial
> Discussion" 
> > Date: Monday, November 16, 2009, 9:51 PM
> > On Mon, Nov 16,
> > 2009 at 2:40 PM, Landy Landy 
> > wrote:
> >  
> > 
> > 
> > I don't know what else to try. When I try to call I
> get
> > this at the cli:
> > 
> > 
> > 
> > == Using SIP RTP CoS mark 5
> > 
> > -- Executing [91xxx763x...@default:1]
> > Dial("SIP/102-b6a06a40",
> > "SIP/1xxx763x...@voipprovider") in new stack
> > 
> > == Using SIP RTP CoS mark 5
> > 
> > -- Called 1xxx763x...@voipprovider
> > 
> > 
> > 
> > We could really use a little more of the CLI output of
> a
> > failed call.  Maybe increase your verbosity to at
> least
> > 10.  Also, what does the SIP debug of a call to the
> VOIP
> > provider look like (from the cli, type "sip set debug
> > peer voipprovider")?
> > 
> > 
> > -- 
> > Thanks,
> > --Warren Selby
> > http://www.selbytech.com
> > 
> > 
> > -Inline Attachment Follows-
> > 
> > ___
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Re: [asterisk-users] Saving CDR on Different Databases

[ABBAS SHAKEEL]

Ahan thats great thanks..(I was already doing this)


On Wed, Nov 18, 2009 at 3:56 PM, mickael ropars  wrote:

> Hi,
>
> I had the same problem as yours, and I am using a deadagi which copy the
> content of the table CDR is the mine.
>
> I am using the uniqueId value in CDR to know wich row I have to copy.
>
> regards
>
> Mickael
>
> 2009/11/18 ABBAS SHAKEEL 
>
>> Hello
>> If we need to save CDRs on different databases for same Asterisk server
>> ie
>> suppose for context [abcd] save to local:5432:abcd
>> and for context [wxyz] save to local:5432:wxyz
>>
>> Can we manage it ? or we need to do some thing in AGI
>>
>>
>> --
>> Kind Regards
>> Shakeel Abbas
>>
>>
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>
>
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Shakeel Abbas
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Re: [asterisk-users] Saving CDR on Different Databases

[mickael ropars]

Hi,

I had the same problem as yours, and I am using a deadagi which copy the
content of the table CDR is the mine.

I am using the uniqueId value in CDR to know wich row I have to copy.

regards

Mickael

2009/11/18 ABBAS SHAKEEL 

> Hello
> If we need to save CDRs on different databases for same Asterisk server ie
> suppose for context [abcd] save to local:5432:abcd
> and for context [wxyz] save to local:5432:wxyz
>
> Can we manage it ? or we need to do some thing in AGI
>
>
> --
> Kind Regards
> Shakeel Abbas
>
>
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] asterisk 1.4.26.3 makes kernel panic

[Vieri]

Hi,

I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3.
There is no core dump, "just" a kernel panic.
This is the only data I could copy from the screen:

EIP: 0060: [] Tainted: P VLI
EFLAGS: 00210297 (2.6.23-gentoo-r8 #1)
eax: 0130 ebx:  ecx: 00220028 edx: 0978
esi: 346e5802 edi:  ebp: c3b45500 esp: caf05bfc
ds: 007b es: 007b fs: 00d8
gs: 0033 ss: 0068
Process asterisk (pid: 7631, ti: caf04000 task: f7987000 task.ti: caf04000)
Stack: ...
Call Trace: ...
Code: ...
EIP: [] ss:esp 0068:caf05bfc
Kernel panic - not syncing: Fatal exception in interrupt

As you can see, kernel is 2.6.23 but it's the same as on another server I have 
but with asterisk 1.2. Kernel 2.6.23 does not panic with asterisk 1.2. The 
hardware on both servers is identical (isdn cards, pri card, etc.) except for 
the motherboard.

How can I further diagnose the problem (besides, I think I can easily reproduce 
it as soon as I set it up as the "production" server)?
What can be causing the kernel panic?

BTW, the last "log entry" I have in /var/log/asterisk/full is:
[Nov 18 10:08:20] VERBOSE[6627] logger.c:   -- dialparties.agi: Checking CW and 
CFB status for extension 6169

(this system runs freepbx)

Help greatly appreciated.

Vieri



  

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[asterisk-users] Saving CDR on Different Databases

[ABBAS SHAKEEL]

Hello
If we need to save CDRs on different databases for same Asterisk server ie
suppose for context [abcd] save to local:5432:abcd
and for context [wxyz] save to local:5432:wxyz

Can we manage it ? or we need to do some thing in AGI


-- 
Kind Regards
Shakeel Abbas
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[asterisk-users] question about call transfer

[Rilawich Ango]

Hi all,
 Refer to http://www.voip-info.org/wiki/view/Asterisk+config+features.conf.
 It shows we can use variable BLINDTRANSFER to call back the one who
transfer the call.  However, in my tests below.  The result is not as
expected.

case 1:
A calls B (dial(sip/B||Tt)
B answers and connects to A
B transfer to C
C doesn't answer the call and B ring again

case 2:
A calls B (dial(sip/B||Tt)
B answers and connects to A
A transfer to C
C doesn't answer the call but B ring instead of A

In case 2, the person who transfer the call can't get back the call.
Anyone can tell whether there is a way to correct in case 2?
Thanks, ango

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