Re: [asterisk-users] Not able to play wav files in asterisk

2011-12-26 Thread Tzafrir Cohen 
On Mon, Dec 26, 2011 at 10:48:36AM +0530, shalu dhamija wrote:
> 
> 
> 
> Hello all, 
> 
> 
> 
> I am trying to play .wav file using asterisk 1.8.7.1.  I tried playing the 
> wav files having different properties. 
> 
> The properties of the file can be seen using 'file' command or 'ffmpeg -i 
> ' 
> 
> 
> 
> 1.  Filname: miss_audio.wav 
> 
> 
> 
> [root@localhost en]# file miss_audio.wav 
> 
> miss_audio.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 
> 8000 Hz 

Asterisk prefers the payload to be signed-linear rather than mu-law.

> 
> 
> 
> [root@localhost en]# ffmpeg -i miss_audio.wav 
> 
> Stream #0:0: Audio: pcm_mulaw ([7][0][0][0] / 0x0007), 8000 Hz, 1 channels, 
> s16, 64 kb/s 

> 
> 
> 
> I am getting the following warning in this case. 
> 
> 
> 
> [Dec 24 15:08:47] WARNING[26513]: format_wav.c:92 check_header_fmt: Not a wav 
> file 7 

This message is, indeed, misleading.

> [Dec 24 15:08:47] WARNING[26513]: file.c:376 fn_wrapper: Unable to open 
> format wav 

But this one is correct.

I wonder if something along the lines of:

  sox orig.wav new.wav -s

would do.

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[asterisk-users] AEL2: How to get rid of "does not end with a return; I will insert one" warnings

2011-12-26 Thread Olivier 
Hi,

I'm converting to asterisk 1.8 an existing (and lengthy) dialplan
written in AEL2.

I'm using in many places things like

macro foo {
BlahBlah();
return;

catch h {
BlahBlah();
}

};

How can I safely get rid of "does not end with a return; I will insert
one" warnings with such constructs ?

Regards

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[asterisk-users] Which language and tool to write diaplans ?

2011-12-26 Thread Olivier 
Hi,

Today, I mostly use AEL2 when writing diaplans but I'm wondering if I
should move to something else.

Which language and tool would you recommend to write diaplans ?

For instance, is Asterisk-LUA a good alternative ?
Or shall I completly move to AGI and write most things in python, Java
or anything else ?

For instance, I used to think I needed some syntax highlighting
support by text editor but I'm not so sure anymore.

Regards

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[asterisk-users] Function TESTTIME example

2011-12-26 Thread Olivier 
Hi,

Has someone a dialplan example using TESTTIME function (see core show
function TESTTIME) ?
I'm only getting replies such as "Function TESTTIME cannot be read".

Regards

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Re: [asterisk-users] Not able to play wav files in asterisk

2011-12-26 Thread amit anand 
Hi

File format is incorrect. First correct the format to 8000Hz, 16 bit mono
PCM uncompressed. you can use wav pad sound editor to do the conversion.

On Mon, Dec 26, 2011 at 15:22, Tzafrir Cohen wrote:

> On Mon, Dec 26, 2011 at 10:48:36AM +0530, shalu dhamija wrote:
> >
> >
> >
> > Hello all,
> >
> >
> >
> > I am trying to play .wav file using asterisk 1.8.7.1.  I tried playing
> the wav files having different properties.
> >
> > The properties of the file can be seen using 'file' command or 'ffmpeg
> -i '
> >
> >
> >
> > 1.  Filname: miss_audio.wav
> >
> >
> >
> > [root@localhost en]# file miss_audio.wav
> >
> > miss_audio.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law,
> mono 8000 Hz
>
> Asterisk prefers the payload to be signed-linear rather than mu-law.
>
> >
> >
> >
> > [root@localhost en]# ffmpeg -i miss_audio.wav
> >
> > Stream #0:0: Audio: pcm_mulaw ([7][0][0][0] / 0x0007), 8000 Hz, 1
> channels, s16, 64 kb/s
>
> >
> >
> >
> > I am getting the following warning in this case.
> >
> >
> >
> > [Dec 24 15:08:47] WARNING[26513]: format_wav.c:92 check_header_fmt: Not
> a wav file 7
>
> This message is, indeed, misleading.
>
> > [Dec 24 15:08:47] WARNING[26513]: file.c:376 fn_wrapper: Unable to open
> format wav
>
> But this one is correct.
>
> I wonder if something along the lines of:
>
>  sox orig.wav new.wav -s
>
> would do.
>
> --
>   Tzafrir Cohen
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>
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Re: [asterisk-users] Not able to play wav files in asterisk

2011-12-26 Thread Israel Gottlieb 
On Mon, Dec 26, 2011 at 9:03 AM, Steve Edwards wrote:

> On Mon, 26 Dec 2011, isr...@gmail.com wrote:
>
>  Rename the wav to ulaw
>> Miss_audio.ulaw
>>
>
> Very bad advice.
>

that might be but if you take a pcm "ulaw" encoded file and name it  ".wav"
asterisk will throw that error
I think asterisk should get smarter and read the header to get the format
or whatever else is needed and not only the extension of the filename

> --
> Thanks in advance,
> --**--**
> -
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> Newline  Fax: +1-760-731-3000
>
>
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[asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread sean darcy 
I'm trying to allow access to the office from home. But the ip provider 
(cablevision) blocks udp 5060. I can see the register packets leaving on 
wireshark, but nothing received by office. Changed to port to 6111 and 
now the packets show up.


In the server I've set port=6111 in the device in sip.conf, but * is NOT 
listening for 6111:


netstat -an | grep 5060
tcp0  0 0.0.0.0:50600.0.0.0:* 
LISTEN

udp0  0 0.0.0.0:50600.0.0.0:*

netstat -an | grep 6111
[root@PBX ~]#

I can't see how to use bindaddr. I need * to also listen to 5060: that's 
what our sip providers use.


So how do I get * to listen to two different ports?

sean


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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread Kevin P. Fleming 

On 12/26/2011 08:55 AM, sean darcy wrote:

I'm trying to allow access to the office from home. But the ip provider
(cablevision) blocks udp 5060. I can see the register packets leaving on
wireshark, but nothing received by office. Changed to port to 6111 and
now the packets show up.

In the server I've set port=6111 in the device in sip.conf, but * is NOT
listening for 6111:

netstat -an | grep 5060
tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN
udp 0 0 0.0.0.0:5060 0.0.0.0:*

netstat -an | grep 6111
[root@PBX ~]#

I can't see how to use bindaddr. I need * to also listen to 5060: that's
what our sip providers use.

So how do I get * to listen to two different ports?


The 'port' setting for a peer has nothing at all to do with where 
Asterisk would listen for traffic; it defines the port number Asterisk 
should *send* traffic to for a non-dynamic peer.


You didn't mention which version of Asterisk you are using, so it's hard 
to give you concrete answers. All current versions of Asterisk can only 
listen on one port for SIP over UDP traffic, but the versions that 
support SIP over TCP can listen on a different port for TCP. If you are 
using a TCP-capable version of Asterisk and a TCP-capable endpoint, you 
can try that.


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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread Doug Lytle 


Kevin P. Fleming wrote:
If you are using a TCP-capable version of Asterisk and a TCP-capable 
endpoint, you can try that. 


Or,

You can setup OpenVPN, works very well.

Doug


--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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[asterisk-users] Debugging call quality issues in Asterisk

2011-12-26 Thread Ishwar Sridharan 
Hola Asterisk Gurus,

We have a telephony server in India which runs CentOS release 5.7 (Final)
version with four-span Digium Card, one of which has two E1 PRI line
terminating on the server.

$ dahdi_hardware
pci::07:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen)

For the last couple of weeks on some calls, we've been experiencing call
quality issues, specifically on outbound calls. We haven't had any
complaints on inbound call quality, and I'm forced to conclude that the
issue exists only for outbound calls.

We haven't seen a pattern on call quality drops, except that it happens
during daytime, when the call volumes are good. A quick look at the system
stats(sar) do not report an increase in system activity coinciding with the
bad quality calls.

$ cat /proc/interrupts; sleep 10; cat /proc/interrupts
reports that there are 10k interrupts generated over a period of 10s. So, I
guess the issue is probably not with the resources. I might also be looking
at all the wrong parameters, being an asterisk/telephony n00b.

Would you folks give me pointers on the best way to figure out if the issue
is with the PRI line or with the server. Also, why does it happen
specifically for outbound calls? Greatly appreciate the help.

--
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[asterisk-users] Working on web based IVR Designer for asterisk and Freeswitch

2011-12-26 Thread Nasir Iqbal 
We are working to develop a web based IVR Designer that will work with
Asterisk as well as Freeswitch using Raphaejs library, Click following link
for detail

http://sourcecodemania.com/ivr-designer-using-raphaeljs-for-asterisk/

Looking for your valuable suggestions


Regards
Nasir Iqbal

ICTBroadcast
SMS, Fax and Voice broadcasting solution
http://www.ictbroadcast.com/
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Re: [asterisk-users] Codec warnings after upgrade to 1.8

2011-12-26 Thread Joseph 

On 12/23/11 10:40, Eric Wieling wrote:

I'm getting various codec related warnings after upgrading to 1.8.  Did I miss 
something in the UPGRADE file?  Does Asterisk no longer transcode 8-)?

WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel 
DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native formats 
0x4 (ulaw)

And

WARNING[11120]: channel.c:4909 ast_write: Codec mismatch on channel 
SIP/interglobe-sip-01e6 setting write format to g722 from ulaw native 
formats 0x4 (ulaw)

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This is similar error I'm getting on asterisk 1.8.7 when trying to communicate 
with codec: ulaw with asterisk 1.4

When I try to dialin on asterisk-1.4.39 I get an error:
NOTICE[2414]: chan_iax2.c:9541 socket_process: Rejected connect
attempt from 192.168.141.8, requested/capability 0x2/0x703
incompatible with our capability 0xc.
NOTICE[2417]: chan_iax2.c:9541 socket_process: Rejected connect
attempt from 192.168.141.8, requested/capability 0x2/0x703
incompatible with our capability 0xc.

On asterisk-1.8.7 I get:
 WARNING[4277]: chan_iax2.c:10666 socket_process: Call rejected by 
192.168.141.1: Unable to negotiate codec

If you find a solution please let me know.





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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread sean darcy 

On 12/26/2011 10:39 AM, Kevin P. Fleming wrote:

On 12/26/2011 08:55 AM, sean darcy wrote:

I'm trying to allow access to the office from home. But the ip provider
(cablevision) blocks udp 5060. I can see the register packets leaving on
wireshark, but nothing received by office. Changed to port to 6111 and
now the packets show up.

In the server I've set port=6111 in the device in sip.conf, but * is NOT
listening for 6111:

netstat -an | grep 5060
tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN
udp 0 0 0.0.0.0:5060 0.0.0.0:*

netstat -an | grep 6111
[root@PBX ~]#

I can't see how to use bindaddr. I need * to also listen to 5060: that's
what our sip providers use.

So how do I get * to listen to two different ports?


The 'port' setting for a peer has nothing at all to do with where
Asterisk would listen for traffic; it defines the port number Asterisk
should *send* traffic to for a non-dynamic peer.

You didn't mention which version of Asterisk you are using, so it's hard
to give you concrete answers. All current versions of Asterisk can only
listen on one port for SIP over UDP traffic, but the versions that
support SIP over TCP can listen on a different port for TCP. If you are
using a TCP-capable version of Asterisk and a TCP-capable endpoint, you
can try that.



Thanks for the response. Home asterisk : 10.0.0  - Office: 1.8.8.0

So I thought I'd leave all the sip providers on udp, and move the 
home-office to tcp.


And registration just work Just Worked over the default tcp registry 
port - which I was surprised to learn was 5060, and not 5061.


So it appears cablevision only blocks udp 5060.

sean





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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread Yaroslav Panych 
2011/12/26 sean darcy :
> So how do I get * to listen to two different ports?
sip.conf
section [general]
bindport=whatever-port-you-want

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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread James Sharp 

On 12/26/2011 04:15 PM, sean darcy wrote:


Thanks for the response. Home asterisk : 10.0.0 - Office: 1.8.8.0

So I thought I'd leave all the sip providers on udp, and move the
home-office to tcp.

And registration just work Just Worked over the default tcp registry
port - which I was surprised to learn was 5060, and not 5061.

So it appears cablevision only blocks udp 5060.

sean


How about using multiple IP tables entries to redirect several ports 
down to 5060.  Dunno if it would work, just a wild-hair guess.  I should 
test it later.



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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread Kevin P. Fleming 

On 12/26/2011 05:06 PM, James Sharp wrote:

On 12/26/2011 04:15 PM, sean darcy wrote:


Thanks for the response. Home asterisk : 10.0.0 - Office: 1.8.8.0

So I thought I'd leave all the sip providers on udp, and move the
home-office to tcp.

And registration just work Just Worked over the default tcp registry
port - which I was surprised to learn was 5060, and not 5061.

So it appears cablevision only blocks udp 5060.

sean


How about using multiple IP tables entries to redirect several ports
down to 5060. Dunno if it would work, just a wild-hair guess. I should
test it later.


It won't, because SIP signaling requires that the UAs include their own 
addresses and ports in the signaling, which requires that they know what 
they are :-) There are some tricks that can be used to get around it, 
but they aren't trivial and they require that the mapping be done 
bidirectionally by whatever is doing the SNAT/DNAT stuff.


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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread sean darcy 

On 12/26/2011 05:43 PM, Yaroslav Panych wrote:

2011/12/26 sean darcy:

So how do I get * to listen to two different ports?

sip.conf
section [general]
bindport=whatever-port-you-want



Thanks, but the problem is to get more than 1 port, 5060 and (at least) 
one other.


sean


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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread Jim Dickenson 
Why not use IAX trunk instead of SIP. This would make it very easy to talk 
between the two * systems.
-- 
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mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Dec 26, 2011, at 4:07 PM, sean darcy wrote:

> On 12/26/2011 05:43 PM, Yaroslav Panych wrote:
>> 2011/12/26 sean darcy:
>>> So how do I get * to listen to two different ports?
>> sip.conf
>> section [general]
>> bindport=whatever-port-you-want
>> 
> 
> Thanks, but the problem is to get more than 1 port, 5060 and (at least) one 
> other.
> 
> sean
> 
> 
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[asterisk-users] odd "secret" problem

2011-12-26 Thread sean darcy 

I've now set up tcp to connect for some home-office connections.

Home is 10.0.0, office is 1.8.8.0.

The home sip device is home-going-to-office, office device: 
office-coming-from-home - home ip is 10.10.11.180


-- Called SIP/home-going-to-office/166
[Dec 26 18:42:31] NOTICE[4387]: chan_sip.c:20408 handle_response_invite: 
Failed to authenticate on INVITE to '"HOME" 
;tag=as3d6c8c47'


Nothing on the office cli.

home:
[home-going-to-office] ;  places calls
type=peer ;; we only call out
transport=tcp
fromuser=office-coming-from-home
remotesecret=password
.

office:
[office-coming-from-home] ; receives  calls
type=friend
transport=tcp
secret=password
..

What's odd is that if I remove secret from [office-coming-from-home] it 
all works!!


sean


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Re: [asterisk-users] how to listen on different sip port for a device?

2011-12-26 Thread sean darcy 

On 12/26/2011 08:17 PM, Jim Dickenson wrote:

Why not use IAX trunk instead of SIP. This would make it very easy to talk 
between the two * systems.


I've tried iax. I found the voice quality was better with sip.

YMMV.

sean


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Re: [asterisk-users] Codec warnings after upgrade to 1.8

2011-12-26 Thread Olivier 
Could you try with 1.8.8.0 ?
I think this one includes a fix for that error.

2011/12/26, Joseph :
> On 12/23/11 10:40, Eric Wieling wrote:
>>I'm getting various codec related warnings after upgrading to 1.8.  Did I
>> miss something in the UPGRADE file?  Does Asterisk no longer transcode
>> 8-)?
>>
>>WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel
>> DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native
>> formats 0x4 (ulaw)
>>
>>And
>>
>>WARNING[11120]: channel.c:4909 ast_write: Codec mismatch on channel
>> SIP/interglobe-sip-01e6 setting write format to g722 from ulaw native
>> formats 0x4 (ulaw)
>>
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>
> This is similar error I'm getting on asterisk 1.8.7 when trying to
> communicate with codec: ulaw with asterisk 1.4
>
> When I try to dialin on asterisk-1.4.39 I get an error:
> NOTICE[2414]: chan_iax2.c:9541 socket_process: Rejected connect
> attempt from 192.168.141.8, requested/capability 0x2/0x703
> incompatible with our capability 0xc.
> NOTICE[2417]: chan_iax2.c:9541 socket_process: Rejected connect
> attempt from 192.168.141.8, requested/capability 0x2/0x703
> incompatible with our capability 0xc.
>
> On asterisk-1.8.7 I get:
>   WARNING[4277]: chan_iax2.c:10666 socket_process: Call rejected by
> 192.168.141.1: Unable to negotiate codec
>
> If you find a solution please let me know.
>
>
>
>
>
> --
> Joseph
>
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[asterisk-users] how to stop hacking of my server

2011-12-26 Thread virendra bhati 
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except iptables ? I want to stop on the basis
of sip.conf account only. bcoz I can't apply iptables rules on server it's
remote server of server provider and we used it for making voip call for
customers.

for the time been i have close all sip accounts. but can't stop for more
then 1 days. I need your help 

*CLI log:- *
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.