Re: [asterisk-users] call forwarding number from outside.

[Mike]

That`s the normal behavior of assisted transfers.  Try a blind/non-assisted
transfer, that should show the original callerid.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 2:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call forwarding number from outside.

 

Hi!

I need help regarding the following problem:

when I receive a phone call to the PBX from the number 01234567890
rings the number 100, get up the phone, I transfer (assisted) to the number
100.
When the 100 number rings, the display shows the number of those who
transferred the call and not the number 01234567890.

How can you solve this problem?

Thanks and sorry for my English 

 

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[asterisk-users] call forwarding number from outside.

[Alessio]

Hi!

I need help regarding the following problem:

when I receive a phone call to the PBX from the number 01234567890
rings the number 100, get up the phone, I transfer (assisted) to the number 100.
When the 100 number rings, the display shows the number of those who 
transferred the call and not the number 01234567890.

How can you solve this problem?

Thanks and sorry for my English 
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[DHAVAL INDRODIYA]

Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that
you will receive in that ,

also read this for better implementation.

http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause

regards
Dhaval

On Fri, Jul 29, 2011 at 11:58 AM, Nikhil  wrote:

> **
> find the inline comment...
>
>
> On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
>
> The dialplan is very simple. When the call comes in, we hand the call over
> to adhearsion.
> This is how the dialplan looks:
>
> ;group 0 will be used for incoming calls
> EXOIN = DAHDI/g0
>
> ;group 11 for outgoing
> EXOOUT = DAHDI/G11
>
> ;This will be used by adhearsion
> EXOCID=
>
> [general]
> autofallthrough = yes ;really?
> clearglobalvars = no
>
> [frompstn]
> ;Send everything to adhearsion
> exten => _X.,1,Ringing
> exten => _X.,n,AGI(agi://127.0.0.1)
>
> exten => _X.,n,Hangup() ; Please try this.
>
>
> ; End dialplan
>
> The rest of the logic happens in adhearsion.
>
> --
> Thanks,
> Ishwar.
>
>
> On Thu, Jul 28, 2011 at 6:33 PM, Nikhil wrote:
>
>>  Can you share the dialplan ,where SIP call is dialing...
>> Thanks
>> Nikhil
>>
>>
>> On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
>>
>>  Hello everybody,
>>
>> We have an asterisk 1.8.4.1 setup, connected to a PRI line.
>>
>> We're currently facing an issue where asterisk does not recognise the
>> event when the called party declines/cuts the call. This happens
>> specifically over calls on a PRI line. For calls over SIP, call decline
>> event is captured properly.
>>
>> I wasn't able to find a solution on the asterisk-users mailing list
>> archive. Any suggestions/help would be much appreiciated :) I can share the
>> relevant parts of the configuration files, if needed.
>>
>> Here's an excerpt from asterisk logs for a SIP call.
>> -- SIP/x- requested special control 16, passing it to
>> SIP/x-0001
>> -- Started music on hold, class 'default', on SIP/x-0001
>> -- SIP/x- requested special control 20, passing it to
>> SIP/x-0001
>> -- Got SIP response 603 "Decline" back from 127.0.0.1:5063
>> -- SIP/x-0001 is busy
>> -- Stopped music on hold on SIP/x-0001
>>
>> As you can see, on a SIP call, a call reject event is identified.
>>
>> For a call over the PRI, on the other hand, this event is not recognised.
>> Here's an excerpt from asterisk log for a call over PRI.
>> Call from  to .
>> -- Requested transfer capability: 0x10 - 3K1AUDIO
>> -- Called G11/x
>> -- Started music on hold, class 'default', on DAHDI/i1/y
>> -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
>> -- DAHDI/i1/x-18f8 is ringing
>> # At this point in time, x rejects the call. The event that's logged
>> in asterisk is the following:
>> -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y
>> # And the call times out after the default 30s.
>> -- Nobody picked up in 3 ms
>>
>> Is there a reason why asterisk doesn't recognise the "call decline", and
>> does it need any configuration changes to enable this?
>>
>> Thanks for your help.
>>
>> --
>> Cheers,
>> Ishwar.
>>
>>
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Nikhil]

find the inline comment...

On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
The dialplan is very simple. When the call comes in, we hand the call 
over to adhearsion.

This is how the dialplan looks:

;group 0 will be used for incoming calls
EXOIN = DAHDI/g0

;group 11 for outgoing
EXOOUT = DAHDI/G11

;This will be used by adhearsion
EXOCID=

[general]
autofallthrough = yes ;really?
clearglobalvars = no

[frompstn]
;Send everything to adhearsion
exten => _X.,1,Ringing
exten => _X.,n,AGI(agi://127.0.0.1 )

   exten => _X.,n,Hangup() ; Please try this.


; End dialplan

The rest of the logic happens in adhearsion.

--
Thanks,
Ishwar.


On Thu, Jul 28, 2011 at 6:33 PM, Nikhil > wrote:


Can you share the dialplan ,where SIP call is dialing...
Thanks
Nikhil


On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:

Hello everybody,

We have an asterisk 1.8.4.1 setup, connected to a PRI line.

We're currently facing an issue where asterisk does not recognise
the event when the called party declines/cuts the call. This
happens specifically over calls on a PRI line. For calls over
SIP, call decline event is captured properly.

I wasn't able to find a solution on the asterisk-users mailing
list archive. Any suggestions/help would be much appreiciated :)
I can share the relevant parts of the configuration files, if needed.

Here's an excerpt from asterisk logs for a SIP call.
-- SIP/x- requested special control 16, passing
it to SIP/x-0001
-- Started music on hold, class 'default', on SIP/x-0001
-- SIP/x- requested special control 20, passing
it to SIP/x-0001
-- Got SIP response 603 "Decline" back from 127.0.0.1:5063

-- SIP/x-0001 is busy
-- Stopped music on hold on SIP/x-0001

As you can see, on a SIP call, a call reject event is identified.

For a call over the PRI, on the other hand, this event is not
recognised. Here's an excerpt from asterisk log for a call over PRI.
Call from  to .
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called G11/x
-- Started music on hold, class 'default', on DAHDI/i1/y
-- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
-- DAHDI/i1/x-18f8 is ringing
# At this point in time, x rejects the call. The event that's
logged in asterisk is the following:
-- DAHDI/i1/x-18f8 is making progress passing it to
DAHDI/i1/y
# And the call times out after the default 30s.
-- Nobody picked up in 3 ms

Is there a reason why asterisk doesn't recognise the "call
decline", and does it need any configuration changes to enable this?

Thanks for your help.

--
Cheers,
Ishwar.


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Re: [asterisk-users] Dialplan required for recording

[DHAVAL INDRODIYA]

Hi Vinod,

You Need to look in MIxmonitor application on asterisk.

http://www.voip-info.org/wiki/view/MixMonitor

http://www.the-asterisk-book.com/unstable/applikationen-mixmonitor.html

Where you can find easy dialplan

On Fri, Jul 29, 2011 at 4:35 AM, Vinod Dharashive wrote:

> Hi team,
>
> Can any one help me to implement dialplan in which conversation between
> a-party and b-party (call patch) needs to be recorded.
>
> Thanks
> Vinod Dharashive
> Sent from BlackBerry® on Airtel
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[asterisk-users] Dialplan required for recording

[Vinod Dharashive]

Hi team,

Can any one help me to implement dialplan in which conversation between a-party 
and b-party (call patch) needs to be recorded.

Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
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Re: [asterisk-users] Voicemail not acting as documented.

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I found what I believe to be a bug and have submitted it:

https://issues.asterisk.org/jira/browse/ASTERISK-18207

Please correct me if I'm wrong.

Barry
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niQbzxp+9BOOy0SyGZkerqU=
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Re: [asterisk-users] Securing Asterisk

[john millican]

On 7/28/2011 11:31 AM, Bruce B wrote:

Hmmm, if alwaysauthreject is already breaking RFC rules then why not
break another rule for the greater good? It would only add another layer
of security.

Maybe: *alwaysregreject=yes*
*
*
*To drop SIP packets for both unauthorized registers and anonymous
calls. Keep it off by default and then allow users to turn it on if they
want to.

To be fair to OP, using Asterisk with open ports to the world is a legit
use of Asterisk even if most of us don't employ it that way or use it
solely with closed networks (VPN, etc...). There are many people who
would benefit from a security feature that would simply ignore
unauthorized registers and anonymous calls.

OP is suggesting an improvement to Asterisk; maybe people should weigh
options and see if it's time to act more on the security side or not.
There is no question that if a hacker knows there is a SIP server then
they will keep the IP on the list for later use or share it
with colleagues even if it seems secure right now. A DDoS is always a
possibility and that you can't save yourself from at all.

Right now the situation is more like this:

*Knock Knock:*
*Owner: *Whose there?
*Thief:* This is Mr. X from China, and I am here to steal your TV.
*Owner: *Hi, I am James Smith, 45, 190lbs and I have a nice laptop as
well but I am home now and I can't let you in.
*Thief (laughing):* No problem, I will come back at midnight when you
are sleeping :-)

- Bruce




What I didn't tell you Mr thief is I sleep very lightly, Have a shotgun, 
a shovel and 20 acres of back yard and I know how to use all three!


Why is there such an aversion to using the right tool for the job? 
Asterisk is not the security tool it is the voice tool!


JohnM


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[asterisk-users] Questions about FMFM with linked servers

[Dovey Forman]

All;

In a linked server environment, running Asterisk 1.6 I am noticing that when
a call is placed from server A to server B (via 4 digit extension) and
server B ext has a FMFM to call their mobile, the mobile phone shows the
default caller ID setting on server B instead of the actual caller id of the
person who initiated the call on server A.

This scenario, of course, works in the event a call in placed via the PSTN
into Server A (or B) and rings the FMFM extension. In this case, the mobile
phones sees the correct (initial) caller ID on the mobile.

Thanks!

--Dovey
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Re: [asterisk-users] Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions

[Mike]

> 
> > I'm looking to disable rejecting calls from my call center employees.
> > They are using Polycom phones. Is there a way to either disable the
> > reject/DND features on the Polycom phones (don`t think so) or have the
> > Asterisk PBX ignore "Got SIP response 486 "Busy Here" back from
> > 12.23.34.45" response from specific phones/SIP registrations and just
> > keep on ringing?
> 
> It is impossible for a SIP UAS to 'ignore' a '486 Busy Here' response.
> SIP phones are not dumb devices like traditional PBX phones, they are
> completely in control of their end of the call. If the phone sends a
> '486 Busy Here' response, it has already dropped the call.
> 
> It *is* possible to disable quite a lot of Polycom phone features via the
> configuration file; I would suggest spending some quality time with the
> Administrator's Guide (and ensure you are running 3.x software on the
> phones) to find out if those features can be disabled.
> 

Thanks Kevin.  The Polycom admin guide isn't new to me, but I have to admit
I missed the relevant part, just found how to do this :
voIpProt.SIP.use486forReject parameter.

Unfortunately, that still leaves DND and the volume keys that cannot from
what I can tell cannot be disabled, so call center agents can still pretend
to work ;-)


Regards,

Mike


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Re: [asterisk-users] Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions

[Eric Wieling]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Mike
> Sent: Thursday, July 28, 2011 3:47 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Disabling Polycom "reject" and "DND" or disable
> Asterisk 486 "Busy Here" actions
>
>
> I'm looking to disable rejecting calls from my call center employees. They
> are using Polycom phones.  Is there a way to either disable the reject/DND
> features on the Polycom phones (don`t think so) or have the Asterisk PBX
> ignore  "Got SIP response 486 "Busy Here" back from 12.23.34.45" response
> from specific phones/SIP registrations and just keep on ringing?

For Firmware 3.3.0 or later.

voIpProt.SIP.serverFeatureControl.localProcessing.dnd="0"
You may also have to add voIpProt.SIP.serverFeatureControl.dnd="1"   Since 
Asterisk doesn't support the Polycom "server based DND" that should effectively 
disable DND. 

If you just want to disable the reject response use call.rejectBusyOnDnd="0"


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Re: [asterisk-users] Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions

[Kevin P. Fleming]

On 07/28/2011 03:47 PM, Mike wrote:


I’m looking to disable rejecting calls from my call center employees.
They are using Polycom phones. Is there a way to either disable the
reject/DND features on the Polycom phones (don`t think so) or have the
Asterisk PBX ignore “Got SIP response 486 "Busy Here" back from
12.23.34.45” response from specific phones/SIP registrations and just
keep on ringing?


It is impossible for a SIP UAS to 'ignore' a '486 Busy Here' response. 
SIP phones are not dumb devices like traditional PBX phones, they are 
completely in control of their end of the call. If the phone sends a 
'486 Busy Here' response, it has already dropped the call.


It *is* possible to disable quite a lot of Polycom phone features via 
the configuration file; I would suggest spending some quality time with 
the Administrator's Guide (and ensure you are running 3.x software on 
the phones) to find out if those features can be disabled.


--
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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] MoH - conversion command

[Eric Wieling]

In order to get the proper encoding for Asterisk, you must provide the 
correct values for each of these characteristics.  In your case, they 
are as follows:

rate = 8000
data size = 8-bit (byte)
data encoding = gsm
channels = 1 (mono)

Therefore, the command you would use to create your native MOH files is:

sox in.wav -t gsm -r 8000 -b -c 1 out.gsm

As G729 is a patented codec and so not available in Free Software, you'll have 
to figure out how to translate those values for whatever software you are using 
to create G729 files.

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Bryant Zimmerman
> Sent: Thursday, July 28, 2011 3:44 PM
> To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] MoH - conversion command
> 
> 
> 
> 
> 
> From: "Mike" 
> Sent: Thursday, July 28, 2011 1:29 PM
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"  us...@lists.digium.com>
> Subject: Re: [asterisk-users] MoH - conversion command
> 
> >
> > > I should have said I am trying this both from a landline using ulaw,
> > > and from a Polycom phone using g729 codec. G729 is noticeablty
> > > worst, as you`d expect, maybe this is what is reported by my customers.
> > >
> > > Is there any way to have a "decent" g729 file, or should I just give
> > > up and change everyone to ulaw ?
> >
> > G.729 is a *speech* codec, and as such it does not handle non-speech
> > (music, tones, etc.) very well at all.
> >
> 
> Hi Kevin,
> 
> I understand that perfectly, I was just wondering if there was a "less bad"
> way of converting music to g729 format so people find the quality bad, but
> don`t feel like cutting their ears off.
> 
> Regards,
> 
> Mike
> 
> 
> 
> Kevin
> 
> I have used the asterisk cli to do my conversion to g729. I have gotten the
> best results there.
> 
> cli   file convert in_file out_file
> asterisk -x "file convert in_file  out_file"
> 
> 
> Thanks
> 
> Bryant Zimmerman (ZK Tech Inc.)


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Re: [asterisk-users] Voicemail not acting as documented.

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 07/28/2011 02:42 PM, Barry L. Kline wrote:

> Is there some other parameter required to get this to fire or am I 
> reading more into that sentence from the CHANGES document than is 
> actually there?

Sorry for replying to my own post, but I've done some more
investigating. I glanced through the source for app_voicemail and am
beginning to wonder if there need be a physical SIP device configured to
use that mailbox for the mailbox to be polled.  Is that the case?

This Asterisk installation is acting as a VM server for a legacy phone
system and none of the VMboxes are actually connected to a SIP phone on
this box.  Can this be the source of my problem?

Barry
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[asterisk-users] Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions

[Mike]

Hi,

 

I'm looking to disable rejecting calls from my call center employees. They
are using Polycom phones.  Is there a way to either disable the reject/DND
features on the Polycom phones (don`t think so) or have the Asterisk PBX
ignore  "Got SIP response 486 "Busy Here" back from 12.23.34.45" response
from specific phones/SIP registrations and just keep on ringing?

 

Mike

 

 

 

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Re: [asterisk-users] MoH - conversion command

[Bryant Zimmerman]

  


From: "Mike" 

Sent: Thursday, July 28, 2011 1:29 PM

To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Subject: Re: [asterisk-users] MoH - conversion command


> 

> > I should have said I am trying this both from a landline using ulaw,

> > and from a Polycom phone using g729 codec. G729 is noticeablty worst,

> > as you`d expect, maybe this is what is reported by my customers.

> >

> > Is there any way to have a "decent" g729 file, or should I just give

> > up and change everyone to ulaw ?

> 

> G.729 is a *speech* codec, and as such it does not handle non-speech

> (music, tones, etc.) very well at all.

>


Hi Kevin,


I understand that perfectly, I was just wondering if there was a "less 
bad"

way of converting music to g729 format so people find the quality bad, but

don`t feel like cutting their ears off. 


Regards,


Mike 


Kevin


I have used the asterisk cli to do my conversion to g729. I have gotten the 
best results there.


cli   file convert in_file out_file

asterisk -x "file convert in_file  out_file"


Thanks


Bryant Zimmerman (ZK Tech Inc.)


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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Eric Wieling]

1) You have to have channels configured for your PRI SOMEWHERE in the Asterisk 
DAHDI configs.
2) Can't troubleshoot when everything important is masked by an AGI script.  
Reproduce the problem using standard dialplan stuff.

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ishwar Sridharan
> Sent: Thursday, July 28, 2011 2:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI
> line
> 
> Hi Eric,
> 
> There weren't any lines with "PRI channel =>" in the chan_dahdi.conf
> 
> However, I added the lines you'd mentioned, near the top of the file. Still,
> no difference in either the behaviour or the asterisk output.
> 
> Please note that as soon as the call lands on asterisk, we pass the control
> over to adhearsion. Does that affect how events are handled in asterisk?
> 
> --
> Thanks,
> Ishwar.
> 
> 
> 
> On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling  wrote:
> 
> 
> 
> 
>   > -Original Message-
>   > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-
> users-
>   > boun...@lists.digium.com] On Behalf Of Nikhil
>   > Sent: Thursday, July 28, 2011 9:03 AM
>   > To: asterisk-users@lists.digium.com
>   > Subject: Re: [asterisk-users] Capturing call Reject/Decline events
> on a PRI
>   > line
> 
>   >
>   > Can you share the dialplan ,where SIP call is dialing...
>   > Thanks
>   > Nikhil
>   >
>   > On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
>   >
>   >   Hello everybody,
>   >
>   >   We have an asterisk 1.8.4.1 setup, connected to a PRI line.
>   >
>   >   We're currently facing an issue where asterisk does not
> recognise
>   > the event when the called party declines/cuts the call. This
> happens
>   > specifically over calls on a PRI line. For calls over SIP, call 
> decline
> event is
>   > captured properly.
>   >
>   >   I wasn't able to find a solution on the asterisk-users mailing 
> list
>   > archive. Any suggestions/help would be much appreiciated :) I can
> share the
>   > relevant parts of the configuration files, if needed.
>   >
>   >   Here's an excerpt from asterisk logs for a SIP call.
>   >   -- SIP/x- requested special control 16, passing 
> it
> to
>   > SIP/x-0001
>   >   -- Started music on hold, class 'default', on SIP/x-
> 0001
>   >   -- SIP/x- requested special control 20, passing 
> it
> to
>   > SIP/x-0001
>   >   -- Got SIP response 603 "Decline" back from 127.0.0.1:5063
> 
>   > 
> 
>   >   -- SIP/x-0001 is busy
>   >   -- Stopped music on hold on SIP/x-0001
>   >
>   >   As you can see, on a SIP call, a call reject event is 
> identified.
>   >
>   >   For a call over the PRI, on the other hand, this event is not
>   > recognised. Here's an excerpt from asterisk log for a call over PRI.
>   >   Call from  to .
>   >   -- Requested transfer capability: 0x10 - 3K1AUDIO
>   >   -- Called G11/x
>   >   -- Started music on hold, class 'default', on DAHDI/i1/y
>   >   -- DAHDI/i1/x-18f8 is proceeding passing it to
> DAHDI/i1/y
>   >   -- DAHDI/i1/x-18f8 is ringing
>   >   # At this point in time, x rejects the call. The event 
> that's
> logged
>   > in asterisk is the following:
>   >   -- DAHDI/i1/x-18f8 is making progress passing it to
>   > DAHDI/i1/y
>   >   # And the call times out after the default 30s.
>   >   -- Nobody picked up in 3 ms
>   >
>   >   Is there a reason why asterisk doesn't recognise the "call
> decline",
>   > and does it need any configuration changes to enable this?
>   >
>   >   Thanks for your help.
> 
> 
> 
>   Try adding the following before your PRI channel => lines in your
> chan_dahdi.conf.  If you are using a GUI like FreePBX, you will have place
> the info where you need to for FreePBX.
> 
>   facilityenable=yes
>   priindication=outofband
> 
> 
> 
> 
>   --
>   
> _
>   -- Bandwidth and Colocation Provided by http://www.api-
> digital.com --
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> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 


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Re: [asterisk-users] MoH - conversion command

[Bryant Zimmerman]

From: "Mike" 

Sent: Thursday, July 28, 2011 1:29 PM

To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Subject: Re: [asterisk-users] MoH - conversion command


> 

> > I should have said I am trying this both from a landline using ulaw,

> > and from a Polycom phone using g729 codec. G729 is noticeablty worst,

> > as you`d expect, maybe this is what is reported by my customers.

> >

> > Is there any way to have a "decent" g729 file, or should I just give

> > up and change everyone to ulaw ?

> 

> G.729 is a *speech* codec, and as such it does not handle non-speech

> (music, tones, etc.) very well at all.

>


Hi Kevin,


I understand that perfectly, I was just wondering if there was a "less 
bad"

way of converting music to g729 format so people find the quality bad, but

don`t feel like cutting their ears off. 


Regards,


Mike 


Kevin


I have used the asterisk cli to do my conversion to g729. I have gotten the 
best results there.


Thanks


Bryant Zimmerman (ZK Tech Inc.)
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Ishwar Sridharan]

Hi Eric,

There weren't any lines with "PRI channel =>" in the chan_dahdi.conf

However, I added the lines you'd mentioned, near the top of the file. Still,
no difference in either the behaviour or the asterisk output.

Please note that as soon as the call lands on asterisk, we pass the control
over to adhearsion. Does that affect how events are handled in asterisk?

--
Thanks,
Ishwar.


On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling  wrote:

>
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Nikhil
> > Sent: Thursday, July 28, 2011 9:03 AM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a
> PRI
> > line
> >
> > Can you share the dialplan ,where SIP call is dialing...
> > Thanks
> > Nikhil
> >
> > On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
> >
> >   Hello everybody,
> >
> >   We have an asterisk 1.8.4.1 setup, connected to a PRI line.
> >
> >   We're currently facing an issue where asterisk does not recognise
> > the event when the called party declines/cuts the call. This happens
> > specifically over calls on a PRI line. For calls over SIP, call decline
> event is
> > captured properly.
> >
> >   I wasn't able to find a solution on the asterisk-users mailing list
> > archive. Any suggestions/help would be much appreiciated :) I can share
> the
> > relevant parts of the configuration files, if needed.
> >
> >   Here's an excerpt from asterisk logs for a SIP call.
> >   -- SIP/x- requested special control 16, passing it
> to
> > SIP/x-0001
> >   -- Started music on hold, class 'default', on
> SIP/x-0001
> >   -- SIP/x- requested special control 20, passing it
> to
> > SIP/x-0001
> >   -- Got SIP response 603 "Decline" back from 127.0.0.1:5063
> > 
> >   -- SIP/x-0001 is busy
> >   -- Stopped music on hold on SIP/x-0001
> >
> >   As you can see, on a SIP call, a call reject event is identified.
> >
> >   For a call over the PRI, on the other hand, this event is not
> > recognised. Here's an excerpt from asterisk log for a call over PRI.
> >   Call from  to .
> >   -- Requested transfer capability: 0x10 - 3K1AUDIO
> >   -- Called G11/x
> >   -- Started music on hold, class 'default', on DAHDI/i1/y
> >   -- DAHDI/i1/x-18f8 is proceeding passing it to
> DAHDI/i1/y
> >   -- DAHDI/i1/x-18f8 is ringing
> >   # At this point in time, x rejects the call. The event that's
> logged
> > in asterisk is the following:
> >   -- DAHDI/i1/x-18f8 is making progress passing it to
> > DAHDI/i1/y
> >   # And the call times out after the default 30s.
> >   -- Nobody picked up in 3 ms
> >
> >   Is there a reason why asterisk doesn't recognise the "call
> decline",
> > and does it need any configuration changes to enable this?
> >
> >   Thanks for your help.
>
>
> Try adding the following before your PRI channel => lines in your
> chan_dahdi.conf.  If you are using a GUI like FreePBX, you will have place
> the info where you need to for FreePBX.
>
> facilityenable=yes
> priindication=outofband
>
>
>
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>
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Ishwar Sridharan]

Hi AJS,

Our dialplan doesn't have a Dial() statement as that's taken care of by
adhearsion.
However, I added "exten => y, n, NoOp(Hangup cause was ${HANGUPCAUSE})"
at the end of our context, restarted asterisk.
The log doesn't have anything new.


--
Thanks,
Ishwar.


On Thu, Jul 28, 2011 at 6:38 PM, A J Stiles
wrote:

> On Thursday 28 Jul 2011, Ishwar Sridharan wrote:
>
> > Is there a reason why asterisk doesn't recognise the "call decline", and
> > does it need any configuration changes to enable this?
>
> What are you seeing for ${HANGUPCAUSE} when this happens ?  (Put a line
> such
> as
>
> exten => y, n, NoOp(Hangup cause was ${HANGUPCAUSE})
>
> in your extensions.conf after the Dial() statement.)
>
> Note that with traditional phones, there is actually no way to decline a
> call
> besides answering it and asking the caller politely to go away  :)
>
> --
> AJS
>
> Answers come *after* questions.
>
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[asterisk-users] Voicemail not acting as documented.

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

In
http://svnview.digium.com/svn/asterisk/branches/1.8/CHANGES?view=markup,
line 180 states:

> Voicemail now runs the externnotify script when pollmailboxes is
> activated and notices a change.

My voicemail.conf configuration for my LDAP vm storage is thus:

externnotify = /opt/asterisk/bin/mwi.pl
pollmailboxes = yes
pollfreq = 30

The script is called whenever I leave a voice mail as well as when I
listen to the voicemail via the voicemail() and voicemailmain()
applications.  When I listen to a voicemail using an email client the
script is not called.  My impression from that line in the CHANGES
document is that it should.

Is there some other parameter required to get this to fire or am I
reading more into that sentence from the CHANGES document than is
actually there?

Thanks.

Barry






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Version: GnuPG v1.4.5 (GNU/Linux)

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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Ishwar Sridharan]

The dialplan is very simple. When the call comes in, we hand the call over
to adhearsion.
This is how the dialplan looks:

;group 0 will be used for incoming calls
EXOIN = DAHDI/g0

;group 11 for outgoing
EXOOUT = DAHDI/G11

;This will be used by adhearsion
EXOCID=

[general]
autofallthrough = yes ;really?
clearglobalvars = no

[frompstn]
;Send everything to adhearsion
exten => _X.,1,Ringing
exten => _X.,n,AGI(agi://127.0.0.1)

; End dialplan

The rest of the logic happens in adhearsion.

--
Thanks,
Ishwar.


On Thu, Jul 28, 2011 at 6:33 PM, Nikhil  wrote:

> **
> Can you share the dialplan ,where SIP call is dialing...
> Thanks
> Nikhil
>
>
> On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
>
> Hello everybody,
>
> We have an asterisk 1.8.4.1 setup, connected to a PRI line.
>
> We're currently facing an issue where asterisk does not recognise the event
> when the called party declines/cuts the call. This happens specifically over
> calls on a PRI line. For calls over SIP, call decline event is captured
> properly.
>
> I wasn't able to find a solution on the asterisk-users mailing list
> archive. Any suggestions/help would be much appreiciated :) I can share the
> relevant parts of the configuration files, if needed.
>
> Here's an excerpt from asterisk logs for a SIP call.
> -- SIP/x- requested special control 16, passing it to
> SIP/x-0001
> -- Started music on hold, class 'default', on SIP/x-0001
> -- SIP/x- requested special control 20, passing it to
> SIP/x-0001
> -- Got SIP response 603 "Decline" back from 127.0.0.1:5063
> -- SIP/x-0001 is busy
> -- Stopped music on hold on SIP/x-0001
>
> As you can see, on a SIP call, a call reject event is identified.
>
> For a call over the PRI, on the other hand, this event is not recognised.
> Here's an excerpt from asterisk log for a call over PRI.
> Call from  to .
> -- Requested transfer capability: 0x10 - 3K1AUDIO
> -- Called G11/x
> -- Started music on hold, class 'default', on DAHDI/i1/y
> -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
> -- DAHDI/i1/x-18f8 is ringing
> # At this point in time, x rejects the call. The event that's logged in
> asterisk is the following:
> -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y
> # And the call times out after the default 30s.
> -- Nobody picked up in 3 ms
>
> Is there a reason why asterisk doesn't recognise the "call decline", and
> does it need any configuration changes to enable this?
>
> Thanks for your help.
>
> --
> Cheers,
> Ishwar.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
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Re: [asterisk-users] MoH - conversion command

[Mike]

you have to select musical compositions that are 'less incompatible'
> with G.729's compression methods. When we chose the current MOH selections
> included with Asterisk, our initial list was much larger, but we had to
> remove some because they sound terrible when compressed with G.729.
> 

Thanks Kevin.  That makes a lot of sense.

Mike


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Re: [asterisk-users] Why no traction for Windows version?

[Gilles]

On Thu, 28 Jul 2011 13:08:33 -0500, "Danny Nicholas"
 wrote:
>If they have, it would probably be on www.nerdvittles.com 

It looks like The Incredible PBX runs on CentOS

www.nerdvittles.com/index.php?p=740


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Re: [asterisk-users] Why no traction for Windows version?

[Tim Nelson]

- Original Message -
> On Thu, 28 Jul 2011 12:46:03 -0500, "Danny Nicholas"
>  wrote:
> >Interrupting - you have to not use DAHDI (SIP Only) and make sure you
> >have
> >the necessary libs downloaded in your Cygwin install.
> 
> It's OK, I don't mind using a VoIP gateway instead of a PCI card. Has
> someone written an HOWTO to compile 1.4 or 1.6 for Windows?
> 
> Does it require patching to Asterisk and/or libraries, or does Cygwin
> handles the whole thing?

Here's a question I must ask...

Why do people want to run Asterisk on Windows? Typically, the person knows the 
Windows environment better. However, that doesn't mean that simply by running 
on Windows Asterisk will be easier to manage. It is still run via console, 
editing text files, etc. In reality, it seems more work and headache to run 
this on a non-supported platform when you're really not gaining anything. AND, 
this doesn't even include the whole Linux vs. Windows argument.

Yes, yes I understand the whole 'its open source I want to do whatever I want 
with it' argument. Fine. But why take that attitude about Asterisk, only to 
amputate it by running on a non-open-source platform?

Kids these days... :)

--Tim

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Re: [asterisk-users] Why no traction for Windows version?

[Danny Nicholas]

If they have, it would probably be on www.nerdvittles.com 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, July 28, 2011 1:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Why no traction for Windows version?

On Thu, 28 Jul 2011 12:46:03 -0500, "Danny Nicholas"
 wrote:
>Interrupting - you have to not use DAHDI (SIP Only) and make sure you 
>have the necessary libs downloaded in your Cygwin install.

It's OK, I don't mind using a VoIP gateway instead of a PCI card. Has
someone written an HOWTO to compile 1.4 or 1.6 for Windows?

Does it require patching to Asterisk and/or libraries, or does Cygwin
handles the whole thing?


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Re: [asterisk-users] Why no traction for Windows version?

[Gilles]

On Thu, 28 Jul 2011 12:46:03 -0500, "Danny Nicholas"
 wrote:
>Interrupting - you have to not use DAHDI (SIP Only) and make sure you have
>the necessary libs downloaded in your Cygwin install.

It's OK, I don't mind using a VoIP gateway instead of a PCI card. Has
someone written an HOWTO to compile 1.4 or 1.6 for Windows?

Does it require patching to Asterisk and/or libraries, or does Cygwin
handles the whole thing?


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Re: [asterisk-users] Why no traction for Windows version?

[Danny Nicholas]

Interrupting - you have to not use DAHDI (SIP Only) and make sure you have
the necessary libs downloaded in your Cygwin install.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, July 28, 2011 12:42 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Why no traction for Windows version?

On Thu, 28 Jul 2011 12:04:38 +0500, "Faisal Hanif" 
wrote:
>I have tried asterisk on windows XP using Cygwin and it worked fine.

Would you mind explaining how to do this?

Thank you.


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Re: [asterisk-users] Why no traction for Windows version?

[Gilles]

On Thu, 28 Jul 2011 12:04:38 +0500, "Faisal Hanif" 
wrote:
>I have tried asterisk on windows XP using Cygwin and it worked fine.

Would you mind explaining how to do this?

Thank you.


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Re: [asterisk-users] MoH - conversion command

[Kevin P. Fleming]

On 07/28/2011 01:29 PM, Mike wrote:



I should have said I am trying this both from a landline using ulaw,
and from a Polycom phone using g729 codec.  G729 is noticeablty worst,
as you`d expect, maybe this is what is reported by my customers.

Is there any way to have a "decent" g729 file, or should I just give
up and change everyone to ulaw ?


G.729 is a *speech* codec, and as such it does not handle non-speech
(music, tones, etc.) very well at all.



Hi Kevin,

I understand that perfectly, I was just wondering if there was a "less bad"
way of converting music to g729 format so people find the quality bad, but
don`t feel like cutting their ears off.


Nope; you have to select musical compositions that are 'less 
incompatible' with G.729's compression methods. When we chose the 
current MOH selections included with Asterisk, our initial list was much 
larger, but we had to remove some because they sound terrible when 
compressed with G.729.


--
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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] MoH - conversion command

[Mike]

> 
> > I should have said I am trying this both from a landline using ulaw,
> > and from a Polycom phone using g729 codec.  G729 is noticeablty worst,
> > as you`d expect, maybe this is what is reported by my customers.
> >
> > Is there any way to have a "decent" g729 file, or should I just give
> > up and change everyone to ulaw ?
> 
> G.729 is a *speech* codec, and as such it does not handle non-speech
> (music, tones, etc.) very well at all.
>

Hi Kevin,

I understand that perfectly, I was just wondering if there was a "less bad"
way of converting music to g729 format so people find the quality bad, but
don`t feel like cutting their ears off. 

Regards,

Mike 
 


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Re: [asterisk-users] MoH - conversion command

[Kevin P. Fleming]

On 07/28/2011 11:57 AM, Mike wrote:


I should have said I am trying this both from a landline using ulaw, and
from a Polycom phone using g729 codec.  G729 is noticeablty worst, as you`d
expect, maybe this is what is reported by my customers.

Is there any way to have a "decent" g729 file, or should I just give up and
change everyone to ulaw ?


G.729 is a *speech* codec, and as such it does not handle non-speech 
(music, tones, etc.) very well at all.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] hide google voice number

[Terry Brummell]

Yes, they used to allow it.  Like CallWithUs and Voip.ms (and I'm sure other 
VTSP's) do.



From: A.H. Jos
Sent: Thu 7/28/2011 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hide google voice number


Do you mean that was possible to set the CID in the early days of GVoice?


On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell  wrote:

Google Voice will show your number no matter what, there was a problem with 
abuse when they let you send the CID in the early days.  Pretty sure there is 
nothing you can do about it.



From: A.H. Jos
Sent: Thu 7/28/2011 9:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] hide google voice number


Hi list,
I have Asterisk speaking with google talk, is there any way to set or at least 
hide my google voice number when I call others?
thanks for your help,
AHJos


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Re: [asterisk-users] hide google voice number

[A.H. Jos]

Do you mean that was possible to set the CID in the early days of GVoice?

On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell  wrote:

>  Google Voice will show your number no matter what, there was a problem
> with abuse when they let you send the CID in the early days.  Pretty sure
> there is nothing you can do about it.
>
> --
> *From:* A.H. Jos
> *Sent:* Thu 7/28/2011 9:22 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] hide google voice number
>
> Hi list,
> I have Asterisk speaking with google talk, is there any way to set or at
> least hide my google voice number when I call others?
> thanks for your help,
> AHJos
>
> --
> _
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] MoH - conversion command

[Mike]

> On 07/28/2011 10:53 AM, Mike wrote:
> > Hi,
> >
> > I've been trying to get MoH files to sound decent. I've got a hold of
> > Royalty-free Classical music (a safe choice for most of my customers)
> > and I`ve been trying to convert them to the normal telephony/Asterisk
> > format using sox. Unfortunately, it sounds really bad. I don't expect
> > concert hall quality of course, 8000KHz being what it is, but is there
> > a better way to convert from good quality .wav files to 8000Khz ? Am I
> > using the wrong tool?
> 
> Can you elaborate as to what you mean by "really bad"?  What acoustic
> artifacts are you encountering?
> 
> Are you testing from a mobile phone?  Cell phones use variable bit rate
> codecs and at times, vicious compression, depending on signal strength and
> other factors.  Anything is going to sound like crap on them regardless.
> Make sure you are testing from a reasonable endpoint.
> 

Alex,

I should have said I am trying this both from a landline using ulaw, and
from a Polycom phone using g729 codec.  G729 is noticeablty worst, as you`d
expect, maybe this is what is reported by my customers.

Is there any way to have a "decent" g729 file, or should I just give up and
change everyone to ulaw ?

Mike



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Re: [asterisk-users] MoH - conversion command

[Robert Huddleston]

Personally I like to just hook up an old ghetto blaster / boombox to the
line in port on my sound card :)

Kidding aside - I think audio quality for MoH is not always going to sound
as nice as you might want.

I mostly stream online radio over my MoH and the quality is not the
greatest.

Maybe it's my SIP provider - or maybe just the notion of streaming audio
from an internet stream.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, July 28, 2011 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MoH - conversion command

On Thu, 28 Jul 2011, Mike wrote:

> I?ve got a hold of Royalty-free Classical music (a safe choice for 
> most of my customers) and I`ve been trying to convert them to the 
> normal telephony/Asterisk format using sox.  Unfortunately, it sounds 
> really bad.

I convert files using:

 sox "${INPUT}" -c 1 -s -w -r 8000 /tmp/$$.wav

What does your sox command line look like?

Can you post a link to 'before' and 'after' files?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000


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Re: [asterisk-users] Securing Asterisk

[Bruce B]

Hmmm, if alwaysauthreject is already breaking RFC rules then why not break
another rule for the greater good? It would only add another layer of
security.

Maybe: *alwaysregreject=yes*
*
*
*To drop SIP packets for both unauthorized registers and anonymous calls.
Keep it off by default and then allow users to turn it on if they want to.

To be fair to OP, using Asterisk with open ports to the world is a legit use
of Asterisk even if most of us don't employ it that way or use it solely
with closed networks (VPN, etc...). There are many people who would benefit
from a security feature that would simply ignore unauthorized registers and
anonymous calls.

OP is suggesting an improvement to Asterisk; maybe people should weigh
options and see if it's time to act more on the security side or not. There
is no question that if a hacker knows there is a SIP server then they will
keep the IP on the list for later use or share it with colleagues even if it
seems secure right now. A DDoS is always a possibility and that you can't
save yourself from at all.

Right now the situation is more like this:

*Knock Knock:*
*Owner: *Whose there?
*Thief:* This is Mr. X from China, and I am here to steal your TV.
*Owner: *Hi, I am James Smith, 45, 190lbs and I have a nice laptop as well
but I am home now and I can't let you in.
*Thief (laughing):* No problem, I will come back at midnight when you are
sleeping :-)

- Bruce



On Wed, Jul 27, 2011 at 2:20 PM, Matthew J. Roth  wrote:

> Kevin P. Fleming wrote:
> >
> > 'alwaysauthreject' in not imcompliant with any RFCs; the RFCs define
> > response codes that *can* be used to indicate (for example) that the
> > Request URI does not represent a target known to the receiver (404 Not
> > Found), but does not mandate that the server respond with that code in
> > that situation.
>
>
> Kevin,
>
> Thanks for the correction and I apologize if I'm propagating a
> misconception.  Am I misunderstanding this Asterisk Security Advisory?
>
> http://lists.digium.com/pipermail/asterisk-announce/2009-April/000177.html
>
>   In 2006, the Asterisk maintainers made it more difficult
>   to scan for valid SIP usernames by implementing an
>   option called "alwaysauthreject"...
>
>   ...What we have done is to carefully emulate exactly the
>   same responses throughout possible dialogs, which should
>   prevent attackers from gleaning this information. All
>   invalid users, if this option is turned on, will receive
>   the same response throughout the dialog, as if a
>   username was valid, but the password was incorrect.
>
>   It is important to note several things. First, this
>   vulnerability is derived directly from the SIP
>   specification, and it is a technical violation of RFC
>   3261 (and subsequent RFCs, as of this date), for us to
>   return these responses...
>
> I am asking out of genuine curiosity, because I trust your assessment
> more than my interpretation of the advisory.
>
> Thank you,
>
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
>
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Re: [asterisk-users] MoH - conversion command

[Steve Edwards]

On Thu, 28 Jul 2011, Mike wrote:

I’ve got a hold of Royalty-free Classical music (a safe choice for most 
of my customers) and I`ve been trying to convert them to the normal 
telephony/Asterisk format using sox.  Unfortunately, it sounds really 
bad.


I convert files using:

sox "${INPUT}" -c 1 -s -w -r 8000 /tmp/$$.wav

What does your sox command line look like?

Can you post a link to 'before' and 'after' files?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000--
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Re: [asterisk-users] MoH - conversion command

[Alex Balashov]

On 07/28/2011 10:53 AM, Mike wrote:

Hi,

I’ve been trying to get MoH files to sound decent. I’ve got a hold of
Royalty-free Classical music (a safe choice for most of my customers)
and I`ve been trying to convert them to the normal telephony/Asterisk
format using sox. Unfortunately, it sounds really bad. I don’t expect
concert hall quality of course, 8000KHz being what it is, but is there
a better way to convert from good quality .wav files to 8000Khz ? Am I
using the wrong tool?


Can you elaborate as to what you mean by "really bad"?  What acoustic 
artifacts are you encountering?


Are you testing from a mobile phone?  Cell phones use variable bit 
rate codecs and at times, vicious compression, depending on signal 
strength and other factors.  Anything is going to sound like crap on 
them regardless.  Make sure you are testing from a reasonable endpoint.


--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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[asterisk-users] MoH - conversion command

[Mike]

Hi,

 

I've been trying to get MoH files to sound decent.  I've got a hold of
Royalty-free Classical music (a safe choice for most of my customers) and
I`ve been trying to convert them to the normal telephony/Asterisk format
using sox.  Unfortunately, it sounds really bad. I don't expect concert hall
quality of course, 8000KHz being what it is, but is there a better way to
convert from good quality .wav files to 8000Khz ? Am I using the wrong tool?

 

Mike

 

 

 

 

 

 

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Re: [asterisk-users] hide google voice number

[Terry Brummell]

Google Voice will show your number no matter what, there was a problem with 
abuse when they let you send the CID in the early days.  Pretty sure there is 
nothing you can do about it.



From: A.H. Jos
Sent: Thu 7/28/2011 9:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] hide google voice number


Hi list,
I have Asterisk speaking with google talk, is there any way to set or at least 
hide my google voice number when I call others?
thanks for your help,
AHJos
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Re: [asterisk-users] Problem H323 asterisk 1.6.2.19

[troxlinux]

2011/7/27 Vladimir Mikhelson :
> Do you have any network devices or VPN tunnels in between the Asterisk
> and Avaya?
>

Hi , the server does not have connections vpn I have and it in the
same LAN that avaya


> The reason I am asking it looks like a potential networking issue.

ok,  but I do ping to him to avaya perfectly, without lost of packages

--- 172.16.8.5 ping statistics ---
16 packets transmitted, 16 received, 0% packet loss, time 15007ms
rtt min/avg/max/mdev = 0.513/1.506/6.393/1.306 ms


>
> Has this setup ever worked before?

I only have it between internal calls of asterisk and works, but h323 no  :(

>
> -Vladimir
>

regardss



-- 
rickygm

http://gnuforever.homelinux.com

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Re: [asterisk-users] Strange network issue

[Mark Deneen]

On Thu, Jul 28, 2011 at 4:46 AM, Duncan Turnbull wrote:

> On 28/07/2011, at 8:41 PM, Paul Hayes  wrote:
>
> > On 28/07/11 02:58, Mike Diehl wrote:
> >>
> >> Any ideas?
> >>
> >> Mike.
> >
> > I'd go on site if possible and see what actually happens at 19:00.  Set
> up a wireshark trace capturing all traffic through their router.
> >
> > --
> I am picking a cleaner plugging a powerful vacuum cleaner in ;-)
>
>
That's what I mentioned earlier, but thinking about it they must have a
German cleaning service to get such precise vacuum timing.

-M
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Re: [asterisk-users] Connect asterisk to normal telephone PBX

[michael k]

Thanks for the reply. I am using an analog phone in normal PBX. I have an
extension called 199 in asterisk and an extension 264 in analog PBX. So how
do i create an inbound or outbound routes for call between these two
extentions ?



On Thu, Jul 28, 2011 at 1:39 PM, Carlos M Cruz wrote:

> Hi,
>
> Did you created your normal Inbound and Outbound routes in freepbx? For use
> with your zap channels?
>
> You'll problably have to change your routes on your pbx too...
>
> Regards,
>
> Carlos M Cruz
>
> 2011/7/28 michael k 
>
>> Hello All,
>>
>> I don't even know the relevancy of my question. Please answer me if my
>> question have some sense.
>>
>> I have recently implemented an asterisk server with freepbx. I have
>> created 100 extentions and i can make successful calls between extensions
>> from anywhere. But my office have three different land-line numbers and
>> three of them are terminating into an internal PBX ( normal matrix telephone
>> PBX)  with more than 60 extensions. This internal PBX is the live PBX where
>> we can call local, STD and ISD from extensions.
>>
>> At present i have some practical difficulties to configure telephone lines
>> at the end of asterisk PBX. So i am trying to connect my asterisk PBX to the
>> normal telephone PBX.
>>
>> I have installed 1 port x100p FXO card  in my asterisk PBX and detected by
>> my freepbx. Then i removed my normal telephone extension cable from phone
>> and connected to the FXO  port of my asterisk PBX.
>>
>> Ultimately my intention is that
>>
>> 1) if somebody call to my normal telephone extension, that should reach to
>> my asterisk server, and asterisk server should send this call to my asterisk
>> extension.
>> 2) if i am calling from my asterisk extension, call should go to the
>> normal telephone PBX via FXO card in my asterisk server and ultimately the
>> call should send outside via the telephone PBX.
>>
>>
>> Is my approach is correct ? If it is wrong please somebody assist me to
>> connect my asterisk PBX to normal telephone PBX.
>>
>> Michael.K
>>
>>
>> --
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>>
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>>
>
>
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Re: [asterisk-users] Why no traction for Windows version?

[Paul Belanger]

On 11-07-26 03:21 AM, Gilles wrote:

Are there just not enough interest and too many, deep, Linux-specific
assumptions in the code, that would explain why Asterisk was never
officially ported to Windows?


Cost?

For me to run Asterisk under Windows requires me to purchase a Microsoft 
license for the OS, with Linux it is not required.


--
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Re: [asterisk-users] hide google voice number

[A.H. Jos]

I am new to asterisk but I think that with gtalk things are different!!!
In my extensions.conf I have:
exten => 2103,2,Dial(SIP/sip.generic,20); twinkle client
exten => _33X,1,Set(CALLERID(num)=01215063743); that doesn't work
exten => _33X,2,Dial(gtalk/asterisk/+${EXTEN}@voice.google.com)

in my sip.conf I have nothing in relation to gtalk:
[sip.generic]
context=google  ;context to be defined in extensions . conf
type=friend   ;use defined context for both inbound and outbound
calls
disallow=all   ;enable gsm audio codec
allow=gsm  ;host could be dynamic
host=dynamic
canreinvite=no

in my gtalk.conf:
[general]
context=google  ; Context to dump call into
bindaddr=0.0.0.0  ; Address to bind to
allowguest=yes  ; Allow calls from people not in list of peers
callerid=2121212
[guest]  ; special account for options on guest account
disallow=all
allow=ulaw
allow=gsm
context=google
connection=asterisk
callerid=2121212

in my jabber.conf:
[general]
; debug = yes; uncomment to Enable debugging ( disabled by
default ).
autoregister=yes; Auto register users from buddy list .
[asterisk]
type=client
serverhost=talk.google.com; Route to server
username=minustoplusinfin...@gmail.com/Talk  ; Username with Talk resource .
secret=*** ; Gmail Password
usetls=yes  ; Use tls
usesasl=yes ; Use sasl
statusmessage="I am available"; Google Talk status message
timeout=100 ; Timeout ( in seconds ),
default is 5



On Thu, Jul 28, 2011 at 3:27 PM, Alex Balashov wrote:

> On 07/28/2011 09:22 AM, A.H. Jos wrote:
>
>> Hi list,
>> I have Asterisk speaking with google talk, is there any way to set or
>> at least hide my google voice number when I call others?
>>
>
> Set a different 'callerid' on either your outgoing sip.conf peer?
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
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>  
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Re: [asterisk-users] hide google voice number

[Alex Balashov]

On 07/28/2011 09:22 AM, A.H. Jos wrote:

Hi list,
I have Asterisk speaking with google talk, is there any way to set or
at least hide my google voice number when I call others?


Set a different 'callerid' on either your outgoing sip.conf peer?

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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[asterisk-users] hide google voice number

[A.H. Jos]

Hi list,
I have Asterisk speaking with google talk, is there any way to set or at
least hide my google voice number when I call others?
thanks for your help,
AHJos
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Eric Wieling]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Nikhil
> Sent: Thursday, July 28, 2011 9:03 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI
> line
> 
> Can you share the dialplan ,where SIP call is dialing...
> Thanks
> Nikhil
> 
> On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
> 
>   Hello everybody,
> 
>   We have an asterisk 1.8.4.1 setup, connected to a PRI line.
> 
>   We're currently facing an issue where asterisk does not recognise
> the event when the called party declines/cuts the call. This happens
> specifically over calls on a PRI line. For calls over SIP, call decline event 
> is
> captured properly.
> 
>   I wasn't able to find a solution on the asterisk-users mailing list
> archive. Any suggestions/help would be much appreiciated :) I can share the
> relevant parts of the configuration files, if needed.
> 
>   Here's an excerpt from asterisk logs for a SIP call.
>   -- SIP/x- requested special control 16, passing it to
> SIP/x-0001
>   -- Started music on hold, class 'default', on SIP/x-0001
>   -- SIP/x- requested special control 20, passing it to
> SIP/x-0001
>   -- Got SIP response 603 "Decline" back from 127.0.0.1:5063
> 
>   -- SIP/x-0001 is busy
>   -- Stopped music on hold on SIP/x-0001
> 
>   As you can see, on a SIP call, a call reject event is identified.
> 
>   For a call over the PRI, on the other hand, this event is not
> recognised. Here's an excerpt from asterisk log for a call over PRI.
>   Call from  to .
>   -- Requested transfer capability: 0x10 - 3K1AUDIO
>   -- Called G11/x
>   -- Started music on hold, class 'default', on DAHDI/i1/y
>   -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
>   -- DAHDI/i1/x-18f8 is ringing
>   # At this point in time, x rejects the call. The event that's logged
> in asterisk is the following:
>   -- DAHDI/i1/x-18f8 is making progress passing it to
> DAHDI/i1/y
>   # And the call times out after the default 30s.
>   -- Nobody picked up in 3 ms
> 
>   Is there a reason why asterisk doesn't recognise the "call decline",
> and does it need any configuration changes to enable this?
> 
>   Thanks for your help.


Try adding the following before your PRI channel => lines in your 
chan_dahdi.conf.  If you are using a GUI like FreePBX, you will have place the 
info where you need to for FreePBX.

facilityenable=yes
priindication=outofband



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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[A J Stiles]

On Thursday 28 Jul 2011, Ishwar Sridharan wrote:

> Is there a reason why asterisk doesn't recognise the "call decline", and
> does it need any configuration changes to enable this?

What are you seeing for ${HANGUPCAUSE} when this happens ?  (Put a line such 
as

exten => y, n, NoOp(Hangup cause was ${HANGUPCAUSE})

in your extensions.conf after the Dial() statement.)

Note that with traditional phones, there is actually no way to decline a call 
besides answering it and asking the caller politely to go away  :)

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Nikhil]

Can you share the dialplan ,where SIP call is dialing...
Thanks
Nikhil

On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:

Hello everybody,

We have an asterisk 1.8.4.1 setup, connected to a PRI line.

We're currently facing an issue where asterisk does not recognise the 
event when the called party declines/cuts the call. This happens 
specifically over calls on a PRI line. For calls over SIP, call 
decline event is captured properly.


I wasn't able to find a solution on the asterisk-users mailing list 
archive. Any suggestions/help would be much appreiciated :) I can 
share the relevant parts of the configuration files, if needed.


Here's an excerpt from asterisk logs for a SIP call.
-- SIP/x- requested special control 16, passing it to 
SIP/x-0001

-- Started music on hold, class 'default', on SIP/x-0001
-- SIP/x- requested special control 20, passing it to 
SIP/x-0001
-- Got SIP response 603 "Decline" back from 127.0.0.1:5063 


-- SIP/x-0001 is busy
-- Stopped music on hold on SIP/x-0001

As you can see, on a SIP call, a call reject event is identified.

For a call over the PRI, on the other hand, this event is not 
recognised. Here's an excerpt from asterisk log for a call over PRI.

Call from  to .
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called G11/x
-- Started music on hold, class 'default', on DAHDI/i1/y
-- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
-- DAHDI/i1/x-18f8 is ringing
# At this point in time, x rejects the call. The event that's 
logged in asterisk is the following:

-- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y
# And the call times out after the default 30s.
-- Nobody picked up in 3 ms

Is there a reason why asterisk doesn't recognise the "call decline", 
and does it need any configuration changes to enable this?


Thanks for your help.

--
Cheers,
Ishwar.


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[asterisk-users] Capturing call Reject/Decline events on a PRI line

[Ishwar Sridharan]

Hello everybody,

We have an asterisk 1.8.4.1 setup, connected to a PRI line.

We're currently facing an issue where asterisk does not recognise the event
when the called party declines/cuts the call. This happens specifically over
calls on a PRI line. For calls over SIP, call decline event is captured
properly.

I wasn't able to find a solution on the asterisk-users mailing list archive.
Any suggestions/help would be much appreiciated :) I can share the relevant
parts of the configuration files, if needed.

Here's an excerpt from asterisk logs for a SIP call.
-- SIP/x- requested special control 16, passing it to
SIP/x-0001
-- Started music on hold, class 'default', on SIP/x-0001
-- SIP/x- requested special control 20, passing it to
SIP/x-0001
-- Got SIP response 603 "Decline" back from 127.0.0.1:5063
-- SIP/x-0001 is busy
-- Stopped music on hold on SIP/x-0001

As you can see, on a SIP call, a call reject event is identified.

For a call over the PRI, on the other hand, this event is not recognised.
Here's an excerpt from asterisk log for a call over PRI.
Call from  to .
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called G11/x
-- Started music on hold, class 'default', on DAHDI/i1/y
-- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
-- DAHDI/i1/x-18f8 is ringing
# At this point in time, x rejects the call. The event that's logged in
asterisk is the following:
-- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y
# And the call times out after the default 30s.
-- Nobody picked up in 3 ms

Is there a reason why asterisk doesn't recognise the "call decline", and
does it need any configuration changes to enable this?

Thanks for your help.

--
Cheers,
Ishwar.
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Re: [asterisk-users] Connect asterisk to normal telephone PBX

[Don Kelly]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, July 28, 2011 1:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Connect asterisk to normal telephone PBX

 

Hello All,

I don't even know the relevancy of my question. Please answer me if my
question have some sense. 

I have recently implemented an asterisk server with freepbx. I have created
100 extentions and i can make successful calls between extensions from
anywhere. But my office have three different land-line numbers and three of
them are terminating into an internal PBX ( normal matrix telephone PBX)
with more than 60 extensions. This internal PBX is the live PBX where we can
call local, STD and ISD from extensions. 

At present i have some practical difficulties to configure telephone lines
at the end of asterisk PBX. So i am trying to connect my asterisk PBX to the
normal telephone PBX.

I have installed 1 port x100p FXO card  in my asterisk PBX and detected by
my freepbx. Then i removed my normal telephone extension cable from phone
and connected to the FXO  port of my asterisk PBX. 

Ultimately my intention is that 

1) if somebody call to my normal telephone extension, that should reach to
my asterisk server, and asterisk server should send this call to my asterisk
extension. 
2) if i am calling from my asterisk extension, call should go to the normal
telephone PBX via FXO card in my asterisk server and ultimately the call
should send outside via the telephone PBX.


Is my approach is correct ? If it is wrong please somebody assist me to
connect my asterisk PBX to normal telephone PBX.

Michael.K

 

 

 

 

The approach sounds right IF your legacy PBX phone is a simple analog phone.
If it is a proprietary phone, you need to try a different approach.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax

 

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Re: [asterisk-users] Lightning and thunder (Claude Hayn

[William Kenworthy]

On Thu, 2011-07-28 at 09:57 +0100, Paul Hayes wrote:
> On 27/07/11 19:41, Claude Hayn wrote:
> >
> > The office manager freaks out each time and starts randomly rebooting
> > devices in no particular order including the UPS, PBX, Asterisk Gateway,
> > firewall and router.
> >
> 
> Ahh that old chestnut.  That's never a good thing, try to tell them not 
> to do this, although I know it's hard, I have customers who love to 
> reboot things too.
> 
> If it is the Asterisk system coming online too early causing problems on 
> the old PBX, does the UPS you are using have a power-on delay feature? 
> In some UPS you can set delays for various sockets on them.  Designed 
> for situations like this and also so everything doesn't try to power up 
> at once causing a power surge.
> 
> cheers,
> Paul.
> 
> --

Remove asterisk from the init process (not sure what distro you have)
and put it in the equivalent of gentoos local.start - the last file run
by the initscripts is for the user to put their own stuff, and because
its last, the system should then be ok when asterisk comes online - if
not its something else causing a problem.

BillK




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[asterisk-users] [chan_mobile addons] DTMF transfer from calling mobile to Asterisk through called mobile FAILED

[Trung Nguyen Dac]

Dear All

I've setup in lab a model include
a *handse*t (nokia 6021 in supported
list)
connect to an *bluetooth dongle* (Cambridge Silicon Radio, Ltd Bluetooth
Dongle (HCI mode)) attached to an *PC *(install Asterisk 1.6.2.19 and Bluez
3.7 with Asterisk addons ( asterisk-addons-1.6.2.3) on Centos 5.5)
Here is the model

[calling mobile ] <---calling to---> [called mobile]<---bluetooth--->[*asterisk
installed on PC*] <--sip-->[softphone]

after config bluetooth and mobile.conf, we could connected the fone to
asterisk, and make incoming / outgoing call to another Mobile.
With INCOMING Call from another mobile to attached mobile, we can hear and
talk to and SIP softphone registered to asterisk (VOICE 2 ways're ok)  but
we CANNOT using DTMF on calling mobile pass through called mobile to
Asterisk.

But in called (attached) mobile, when press keys, Asterisk can recognized
that. So i wonder that the dtmf discussion in some topic is dtmf from called
mobile to Asterisk.
Does any one has experienced with this issue can share your opinion . This
function is useful for demo

ps: sorry for verbose explanation.


Thank for your attention and looking for your rep ASAP
* *
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Re: [asterisk-users] Lightning and thunder (Claude Hayn

[Paul Hayes]

On 27/07/11 19:41, Claude Hayn wrote:


The office manager freaks out each time and starts randomly rebooting
devices in no particular order including the UPS, PBX, Asterisk Gateway,
firewall and router.



Ahh that old chestnut.  That's never a good thing, try to tell them not 
to do this, although I know it's hard, I have customers who love to 
reboot things too.


If it is the Asterisk system coming online too early causing problems on 
the old PBX, does the UPS you are using have a power-on delay feature? 
In some UPS you can set delays for various sockets on them.  Designed 
for situations like this and also so everything doesn't try to power up 
at once causing a power surge.


cheers,
Paul.

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Re: [asterisk-users] Strange network issue

[Duncan Turnbull]

On 28/07/2011, at 8:41 PM, Paul Hayes  wrote:

> On 28/07/11 02:58, Mike Diehl wrote:
>> 
>> Any ideas?
>> 
>> Mike.
> 
> I'd go on site if possible and see what actually happens at 19:00.  Set up a 
> wireshark trace capturing all traffic through their router.
> 
> --
I am picking a cleaner plugging a powerful vacuum cleaner in ;-)
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Re: [asterisk-users] Strange network issue

[Paul Hayes]

On 28/07/11 02:58, Mike Diehl wrote:


Any ideas?

Mike.


I'd go on site if possible and see what actually happens at 19:00.  Set 
up a wireshark trace capturing all traffic through their router.


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Re: [asterisk-users] Connect asterisk to normal telephone PBX

[Carlos M Cruz]

Hi,

Did you created your normal Inbound and Outbound routes in freepbx? For use
with your zap channels?

You'll problably have to change your routes on your pbx too...

Regards,

Carlos M Cruz

2011/7/28 michael k 

> Hello All,
>
> I don't even know the relevancy of my question. Please answer me if my
> question have some sense.
>
> I have recently implemented an asterisk server with freepbx. I have created
> 100 extentions and i can make successful calls between extensions from
> anywhere. But my office have three different land-line numbers and three of
> them are terminating into an internal PBX ( normal matrix telephone PBX)
> with more than 60 extensions. This internal PBX is the live PBX where we can
> call local, STD and ISD from extensions.
>
> At present i have some practical difficulties to configure telephone lines
> at the end of asterisk PBX. So i am trying to connect my asterisk PBX to the
> normal telephone PBX.
>
> I have installed 1 port x100p FXO card  in my asterisk PBX and detected by
> my freepbx. Then i removed my normal telephone extension cable from phone
> and connected to the FXO  port of my asterisk PBX.
>
> Ultimately my intention is that
>
> 1) if somebody call to my normal telephone extension, that should reach to
> my asterisk server, and asterisk server should send this call to my asterisk
> extension.
> 2) if i am calling from my asterisk extension, call should go to the normal
> telephone PBX via FXO card in my asterisk server and ultimately the call
> should send outside via the telephone PBX.
>
>
> Is my approach is correct ? If it is wrong please somebody assist me to
> connect my asterisk PBX to normal telephone PBX.
>
> Michael.K
>
>
> --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Radius billing for asterisk

[Nikhil]

Hi
Any company proving radius based billing for asterisk only for 
accounting ,not authenication and atherization.Please provide some links


Thanks
Nikhil

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Re: [asterisk-users] Why no traction for Windows version?

[Faisal Hanif]

I have tried asterisk on windows XP using Cygwin and it worked fine.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antonio Modesto
Sent: Thursday, July 28, 2011 1:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Why no traction for Windows version?

 

On Tue, 2011-07-26 at 09:45 +0200, Gilles wrote: 

 
On Tue, 26 Jul 2011 07:28:27 +, "Soeren Malchow (MCon)"
 wrote:
>And asterisk just runs fine on linux why bother ?
 
Because I, for one, would like to run Asterisk on my Windows
workstation at home as an enhanced answering machine :-)



Windows never was a good solution for these things, and i think it will never 
be.




 
 
 
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