Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Kaushal Shriyan]

On Wed, Sep 14, 2011 at 7:04 AM, Steve Edwards
 wrote:
> On Wed, 14 Sep 2011, Kaushal Shriyan wrote:
>
>> I have carried out the below steps
>>
>> [root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1
>> obd-demo.alaw
>
>> sox: Output file obd-demo.alaw: using sample rate 8000
>>        size bytes, encoding u-law, 1 channel
>
> Sox v14.x complains about the '-b.'
>
> You are encoding as u-law, but naming the file alaw. Since [a|u]law are
> 'headerless' file formats, this will probably confuse Asterisk.
>
>> -rwxr-xr-x 1 root root  741459 Sep 14 06:32 obd-demo.mp3
>> -rw-r--r-- 1 root root 2041482 Sep 14 06:32 obd-demo.wav
>> -rw-r--r-- 1 root root  185165 Sep 14 06:32 obd-demo.alaw
>>
>> Am i doing it correctly ? Please comment
>
> I've never used alaw (I'm in the US).
>
> I don't think an MP3 needs 'execute' permissions :)
>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000

Hi Steve,

Please let me know the correct procedure to get .alaw file format
since I belong to India region.

Regards

Kaushal

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[asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call

[Lee, John (Sydney)]

I was trying to do a SIP call between two Asterisk servers (1.4.21.2)

1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);

2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends place calls and receive calls
context=incoming   ; Context for incoming calls from this
user
host=dynamic   ; This peer register with us
dtmfmode=rfc2833   ; Choices are inband, rfc2833, or info
qualify=yes; Monitor latency between Asterisk server
and phone
call-limit=99
username=1166  ; Username to use in INVITE until peer
registers
secret=password; Normally you do NOT need to set this
parameter
mailbox=1166@default   ; mailbox 5100 in voicemail context
.default.
callgroup=1
pickupgroup=1

The call was unsuccessful as follows.
 
1) On the caller machine, this is what we got from the console
-- Executing [1166@incoming:1] Dial("SIP/1166-09d81668",
"SIP/1166:password@asterisk-callee") in new stack
-- Called 1166:password@asterisk-callee
-- SIP/asterisk-callee is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

2) On the callee machine, this is what we got from the console,
[Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite:
Call from '2765' to extension '1166:password' rejected because extension
not found.

However, I found out that if I remove "secret=.." from the SIP entry and
call without the password, then I will be able to call.

Any thoughts?

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Re: [asterisk-users] Reporting for Asterisk Call Center

[Nicolás Gudiño]

Hi Tzafrir,

On Sat, Sep 10, 2011 at 4:28 AM, Tzafrir Cohen wrote:

> On Fri, Sep 09, 2011 at 01:28:28PM -0500, Gerardo Barajas wrote:
> > There are a lot of reporting tools.
> > I have used:
> >
> > Asternic: http://www.asternic.biz/
>
Non of those are Free (Open Source).
>
>
Clarification: Asternic Call Center Stats Lite is free (GPL3) and can be
downloaded from the above link. The PRO version is commercial.
Asternic CDR reports for FreePBX is also free and available for download on
the asternic.biz site.

--
Nicolás Gudiño
I'm speaking at ElastixWorld: http://www.elastixworld.com/2011/
I'm speaking at 4K Conference: http://www.4kconf.com
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Re: [asterisk-users] using variables in the shell function

[Dale Noll]

On 09/13/2011 07:49 PM, Israel Gottlieb wrote:

is it possible to pas variables to the shell function

Set(recordingavail=${SHELL("ls
/var/lib/asterisk/sounds/custom/${TOPMENU}")})

im trying to see if a file is available before playing the file

or does anybody have a different idea but not using agi

asterisk 1.6.2.20
thanks



You should check out the STAT function.

core show function STAT



This should evaluate to 1
 ${STAT(e,/var/lib/asterisk/sounds/en/vm-goodbye.gsm)})

This should evaluate to 0
 ${STAT(e,/var/lib/asterisk/sounds/en/xyzzy.gsm)}


Dale


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Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Steve Edwards]

On Wed, 14 Sep 2011, Kaushal Shriyan wrote:


I have carried out the below steps

[root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.alaw



sox: Output file obd-demo.alaw: using sample rate 8000
size bytes, encoding u-law, 1 channel


Sox v14.x complains about the '-b.'

You are encoding as u-law, but naming the file alaw. Since [a|u]law are 
'headerless' file formats, this will probably confuse Asterisk.



-rwxr-xr-x 1 root root  741459 Sep 14 06:32 obd-demo.mp3
-rw-r--r-- 1 root root 2041482 Sep 14 06:32 obd-demo.wav
-rw-r--r-- 1 root root  185165 Sep 14 06:32 obd-demo.alaw

Am i doing it correctly ? Please comment


I've never used alaw (I'm in the US).

I don't think an MP3 needs 'execute' permissions :)

--
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] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Kaushal Shriyan]

On Wed, Sep 14, 2011 at 6:42 AM, Steve Edwards
 wrote:
> On Wed, 14 Sep 2011, Kaushal Shriyan wrote:
>
>> Also please let me know the difference between .ulaw and .alaw format and
>> is there a way i can play this file formats.
>
> alaw = Europe, ulaw = US & Japan
>
> Wikipedia has articles on both algorithms if you are interested in the
> specifics.
>
> If you format your file correctly, Asterisk can play it.
>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
>
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>

Hi Steve,

I have carried out the below steps

[root@host0040 test]# lame --decode obd-demo.mp3 obd-demo.wav
input:  obd-demo.mp3  (44.1 kHz, 1 channel, MPEG-1 Layer III)
output: obd-demo.wav  (16 bit, Microsoft WAVE)
skipping initial 1105 samples (encoder+decoder delay)
Frame#   887/886256 kbps
[root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.alaw
sox: Detected file format type: wav

sox: WAV Chunk fmt
sox: WAV Chunk data
sox: Reading Wave file: Microsoft PCM format, 1 channel, 44100 samp/sec
sox: 88200 byte/sec, 2 block align, 16 bits/samp, 2041438 data bytes
sox: 1020719 Samps/chans
sox: Input file obd-demo.wav: using sample rate 44100
size shorts, encoding signed (2's complement), 1 channel
sox: Output file obd-demo.alaw: using sample rate 8000
size bytes, encoding u-law, 1 channel
sox: Output file: comment "Processed by SoX"

sox: resample opts: Kaiser window, cutoff 0.80, beta 16.00

[root@host0040 test]# ls -ltr
total 2932
-rwxr-xr-x 1 root root  741459 Sep 14 06:32 obd-demo.mp3
-rw-r--r-- 1 root root 2041482 Sep 14 06:32 obd-demo.wav
-rw-r--r-- 1 root root  185165 Sep 14 06:32 obd-demo.alaw

Am i doing it correctly ? Please comment

Regards,

Kaushal

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Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Steve Edwards]

On Wed, 14 Sep 2011, Kaushal Shriyan wrote:

Also please let me know the difference between .ulaw and .alaw format 
and is there a way i can play this file formats.


alaw = Europe, ulaw = US & Japan

Wikipedia has articles on both algorithms if you are interested in the 
specifics.


If you format your file correctly, Asterisk can play it.

--
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] using variables in the shell function

[Steve Edwards]

On Wed, 14 Sep 2011, Israel Gottlieb wrote:


is it possible to pas variables to the shell function

Set(recordingavail=${SHELL("ls /var/lib/asterisk/sounds/custom/${TOPMENU}")})

im trying to see if a file is available before playing the file

or does anybody have a different idea but not using agi


Why not AGI?

They both ('shelling out' or calling an AGI) have the same 'impact' on 
system resources.


You can even write an AGI in shell if you lack the skills for other 
languages like C, PHP, or Perl.


You should be able to cobble up an AGI in PHP (or Perl, but I'm not much 
of a Perl coder myself) just by cutting and pasting from some of the 
examples on voip-info.org.


This simple task would be a great way for you to 'get your feet wet.'

What will you do if the file is not available?

--
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] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Kaushal Shriyan]

On Tue, Sep 13, 2011 at 6:47 PM, Matthew J. Roth  wrote:
> Kaushal,
>
> Your version of SoX does not have MP3 support.  Since you have LAME
> installed, use it as a first step to produce an intermediate file
> that SoX supports.  Then use SoX to convert the intermediate file
> to the desired format.
>
> Step 1
> --
>
> # lame --decode obd-demo.mp3 obd-demo.wav
> input:  obd-demo.mp3  (8 kHz, 1 channel, MPEG-2.5 Layer III)
> output: obd-demo.wav  (16 bit, Microsoft WAVE)
> skipping initial 1105 samples (encoder+decoder delay)
> Frame# 16818/16818   16 kbps
> # file obd-demo.wav
> obd-demo.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, 
> mono 8000 Hz
>
>
> Step 2
> --
>
> # sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.ulaw
> sox: Detected file format type: wav
>
> sox: WAV Chunk fmt
> sox: WAV Chunk data
> sox: Reading Wave file: Microsoft PCM format, 1 channel, 8000 samp/sec
> sox:         16000 byte/sec, 2 block align, 16 bits/samp, 19372126 data bytes
> sox:         9686063 Samps/chans
> sox: Input file obd-demo.wav: using sample rate 8000
>        size shorts, encoding signed (2's complement), 1 channel
> sox: Output file obd-demo.ulaw: using sample rate 8000
>        size bytes, encoding u-law, 1 channel
> sox: Output file: comment "Processed by SoX"
>
>
> Regards,
>
> Matthew Roth

Thanks Matthew Roth. It worked. Also please let me know the difference
between .ulaw and .alaw format and is there a way i can play this file
formats.

Regards

Kaushal

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[asterisk-users] using variables in the shell function

[Israel Gottlieb]

is it possible to pas variables to the shell function

Set(recordingavail=${SHELL("ls
/var/lib/asterisk/sounds/custom/${TOPMENU}")})

im trying to see if a file is available before playing the file

or does anybody have a different idea but not using agi

asterisk 1.6.2.20
thanks
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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

I made the switch and everything seems to be working. It's hard to tell, since 
it never seems to fail for me, but fails once people get in. 

A question, though. When we moved the original box to our new office, they 
asked if we could support the dms100 setting, which worked. National seems to 
be working.  Is there a big difference between the two? 

I'm just a simple programmer who happens to be the only IT guy in the office. 

- Stephen H. Gerstacker

On Sep 13, 2011, at 10:23, "Doug Lytle"  wrote:

> 
> Stephen H. Gerstacker wrote:
>> Anything else I can try?
> 
> Try switchtype=national just for testing.
> 
> 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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Re: [asterisk-users] Question about voip.ms service.

[John Novack]

naren wrote:


Ok that makes sense. I will take a look at my set up and see why it is not 
registering with voip.ms .


Understand that with IAX, voip.ms will not show you as registered.
Your Asterisk should show you as registered from the CLI
CLI> iax2 show registry
XX.XX.XX.XXX:4569 N   Y  ZZ.ZZ.ZZZ.ZZZ:4569 60  
Registered

X= Voip.ms server
Y=your account number
Z=Your IP address

In IAX general section:

register => Y:passw...@newyork.voip.ms ; change this to your server 
specified

this is shown in their example, with your data filled in

then your specific section for voipms:

[voipms];
;
type=friend
username=Y
secret=PASSWORD
context=from-voipms ; this points to your inbound context in extensions
host=newyork.voip.ms
disallow=all
allow=ulaw ;Codec 1 supported
allow=gsm  ; Codec 2 supported
insecure=port,invite ; from voip.ms example
requirecalltoken=no ; required after 1.4.26

Hope this helps

JN


I opened a ticket with voip.ms  as well about an hour ago. I do 
like their service as well, that is why I want to try and get it working with them.

Thanks John.

On Tue, Sep 13, 2011 at 5:29 PM, John Novack mailto:jnov...@stromberg-carlson.org>> wrote:

Voip.ms has excellent support if you need it, which many do not.
You log in to your account, then you can change from SIP to IAX, and if you 
click on the correct link they will give you your sample with your account 
information
You need to set up a registration line in IAX, then a context in IAX that 
points to a context in extensions.conf
Registration takes care of voip.ms  finding you
their web site setup is about as complete a site as I have seen, with many 
more options than I would ever need
The only somewhat confusing issue is when using IAX they will not show you 
as registered
Your Asterisk will, though.

John Novack


naren wrote:


That's what I am hoping to do as well. Could you share some insight on how you set up 
the DID on the voip.ms  web site to forward to Asterisk using IAX? In 
particular I am trying to find out where you set the url / ip address of your asterisk 
installation on the voip.ms  web site.

Thanks!

On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone mailto:rhuddles...@gmail.com>> wrote:

I'm using them for inbound and outbound on Asterisk and FreeSwitch

Sent from my iPhone



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[asterisk-users] realtime goto/gotoif/dial

[Hans Witvliet]

Hi all,

I presume i made a silly mistake while filling a database

But while googling on the results, i came across a lot of messages about
the layout of app_data in case of goto and dial statements.
(mostly about using the old "|" seperator instead of the "," separator.

So i was wondering, is this issue been solved? (I presume so, but can
not find any confirmation about it)

Hans

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Re: [asterisk-users] Question about voip.ms service.

[naren]

Ok that makes sense. I will take a look at my set up and see why it is not
registering with voip.ms.

I opened a ticket with voip.ms as well about an hour ago. I do like their
service as well, that is why I want to try and get it working with them.

Thanks John.

On Tue, Sep 13, 2011 at 5:29 PM, John Novack
wrote:

>  Voip.ms has excellent support if you need it, which many do not.
> You log in to your account, then you can change from SIP to IAX, and if you
> click on the correct link they will give you your sample with your account
> information
> You need to set up a registration line in IAX, then a context in IAX that
> points to a context in extensions.conf
> Registration takes care of voip.ms finding you
> their web site setup is about as complete a site as I have seen, with many
> more options than I would ever need
> The only somewhat confusing issue is when using IAX they will not show you
> as registered
> Your Asterisk will, though.
>
> John Novack
>
>
> naren wrote:
>
>
>  That's what I am hoping to do as well. Could you share some insight on how
> you set up the DID on the voip.ms web site to forward to Asterisk using
> IAX? In particular I am trying to find out where you set the url / ip
> address of your asterisk installation on the voip.ms web site.
>
>  Thanks!
>
> On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone wrote:
>
>>  I'm using them for inbound and outbound on Asterisk and FreeSwitch
>>
>> Sent from my iPhone
>>
>>
>
> --
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>
> --
>
> Dog is my Co-pilot
>
>
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Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?

[Tarek Sawah]

try to look for N82 nokia mobile devices.. you get the benefits of a Mobile 
device with it's phone book and mobility features (games when you are bored :P) 
.. and other features.. and the native SIP client works fluently with no 
problems at all supporting almost commercial codecs like (G729).. and it works 
with WIFI.. i use it at home.





Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993




> Date: Sat, 27 Aug 2011 10:14:24 +0100
> From: gordon+aster...@drogon.net
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
>
> On Sat, 27 Aug 2011, Alan Lord (News) wrote:
>
> > On 26/08/11 19:02, linux guy wrote:
> >> I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home
> >> asterisk system.
> >
> > We've been using the Siemens Gigaset 685IP range for over three years and 
> > I'm
> > (still) very pleased with them:
>
> +1
>
> The base station is separate from the handsets - which is typically
> different from most DECT setups - the plus point is that you can position
> the base in a good location - ie. high on a wall, rather than anywhere
> else. Another plus is that the base has a single built-in ATA, so it can
> connect to the home PSTN line. The base also has an Ethernet socket to
> connect to the LAN and it can have up to 6 SIP accounts - each handset (up
> to 6) can be configured to ring on a particular SIP account or many SIP
> accounts and/or the PSTN line. Each handset has a default SIP account (or
> PSTN) to make outgoing calls on, but you can select any other SIP account
> or the PSTN by appending a code to the number you dial.
>
> They are very flexible - and being DECT, have superb range.
>
> I've installed many of these for my customers - typically the home office
> types - where they only want one phone on their desk - so the same handset
> can answer their home phone or their office SIP account, while providing
> wireless handsets throughout the rest of the house.
>
> A limitation is that one base can only handle 2 simultaneous SIP calls
> (plus a call via the PSTN), so if 2 phones are in-use, then the system
> can't take a 3rd call, however that's rarely a limitation in a domestic
> environment.
>
> Gordon
>
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Re: [asterisk-users] Anyone using Asterisk on VirtualBox ?

[Tarek Sawah]

i did do some Asterisk tests on SUN VBOX .. works like a charm but you need to 
dedicate some good resources to the virtual box!



Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993




> From: zhulizh...@live.com 
> To: asterisk-users@lists.digium.com 
> Date: Fri, 2 Sep 2011 08:37:55 + 
> Subject: Re: [asterisk-users] Anyone using Asterisk on VirtualBox ? 
>  
> hi: 
> please check the redfone solution. 
>  
> Best regards, 
> James.zhu 
> Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,  
> gateway(fxs/fxo/pri<->SIP). 
> website: www.voipviews.com 
>  
>  
>  > From: aster...@a-domani.nl 
>  > To: asterisk-users@lists.digium.com 
>  > Date: Thu, 1 Sep 2011 23:48:46 +0200 
>  > Subject: Re: [asterisk-users] Anyone using Asterisk on VirtualBox ? 
>  > 
>  > On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote: 
>  > 
>  > > 
>  > > 
>  > > My main interest of being on Virtual platform is portability / Backup. 
>  > > In case of any h/w issues, or crashes, simply copy the VM on to 
>  > > another box and you are up in minutes. 
>  > > 
>  > > 
>  > > Sanjay 
>  > > -- 
>  > Doing that right now, although in my case i use XEN. 
>  > Besides being hw independant, it is easier to play with a different 
>  > version for a while (1.4 / 1.6.0 / 1.6.1 / 1.6.2 / 1.8.0) and being able 
>  > to switch back in minutes. 
>  > 
>  > hw 
>  > 
>  > -- 
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Re: [asterisk-users] Question about voip.ms service.

[John Novack]

Voip.ms has excellent support if you need it, which many do not.
You log in to your account, then you can change from SIP to IAX, and if you 
click on the correct link they will give you your sample with your account 
information
You need to set up a registration line in IAX, then a context in IAX that 
points to a context in extensions.conf
Registration takes care of voip.ms finding you
their web site setup is about as complete a site as I have seen, with many more 
options than I would ever need
The only somewhat confusing issue is when using IAX they will not show you as 
registered
Your Asterisk will, though.

John Novack


naren wrote:


That's what I am hoping to do as well. Could you share some insight on how you set up the 
DID on the voip.ms  web site to forward to Asterisk using IAX? In 
particular I am trying to find out where you set the url / ip address of your asterisk 
installation on the voip.ms  web site.

Thanks!

On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone mailto:rhuddles...@gmail.com>> wrote:

I'm using them for inbound and outbound on Asterisk and FreeSwitch

Sent from my iPhone



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Re: [asterisk-users] Asterisk 1.8 not accepting call from DID

[Tarek Sawah]

you didn't provide your "dialplan" for the incoming call context from_poland? 
nor registration string?
could be a dial plan problem .. or codec issue.. as long as you register 
"properly" the server has no problem with NAT.. it's a routing or codec issue i 
think.

Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993




> Date: Mon, 5 Sep 2011 19:50:34 -0600
> From: syscon...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 1.8 not accepting call from DID
>
> It seems to me "nat=yes" is not working correctly in asterisk 1.8.5
> rtp set debug on
>
> shows:
> Got RTP packet from 10.0.0.110:6000 (type 00, seq 029667, ts 2129095321, len 
> 000160)
> Sent RTP packet to 10.0.0.110:6010 (type 00, seq 065112, ts 2129095320, len 
> 000160)
>
> I've tried 'nat=yes' 'nat=comedia' it makes no differece.
>
> --
> Joseph
>
> On 09/05/11 15:00, Joseph wrote:
> >I have DID, it registers OK with the provider, but when I try to call this 
> >number (it suppose to ring my Asterisk) asterisk 1.8 does not respond.
> >
> >sip show peers
> >Name/username Host Dyn Forcerport ACL Port Status
> >actio-out/48746612254 81.15.150.20 N 5060 OK (201ms)
> >
> >sip.conf part:
> >[general]
> >context=default
> >allowguest=no allowoverlap=no
> >udpbindaddr=0.0.0.0
> >useragent = Centrala
> >
> >[actio-out]
> >type=friend
> >secret=
> >user=48746612254
> >username=48746612254
> >fromuser=48746612254
> >authname=48746612254
> >callerpage=48746612254
> >fromdomain=sip.actio.pl
> >host=sip.actio.pl
> >insecure=port,invite
> >nat=yes
> >qualify=yes
> >dtmfmode=inband
> >disallow=all
> >allow=ulaw
> >allow=alaw
> >context=from_poland
> >canreinvite=no
> >
> >The setting above worked OK with Asteriks 1.4
> >
> >Here is debug info, which I don't know how to interpret.
> >
> >-- Executing [901148746612254@internal:1] Dial("SIP/11-0002", 
> >"SIP/901148746612254@pstn-1270,60,tr") in new stack
> >[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:25695 sip_request_call: Asked to 
> >create a SIP channel with formats: 0x4 (ulaw)
> > == Using UDPTL CoS mark 5
> >[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:7496 sip_alloc: Allocating new SIP 
> >dialog for 5a2cdf8339e0ad2911ad393036c05165@127.0.0.1:0 - INVITE (No RTP)
> >[Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:347 ast_rtp_instance_new: Using 
> >engine 'asterisk' for RTP instance '0x88c3b10'
> >[Sep 5 14:04:35] DEBUG[26209]: res_rtp_asterisk.c:474 ast_rtp_new: Allocated 
> >port 16690 for RTP instance '0x88c3b10'
> >[Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:356 ast_rtp_instance_new: RTP 
> >instance '0x88c3b10' is setup and ready to go
> >[Sep 5 14:04:35] DEBUG[26209]: res_rtp_asterisk.c:2372 ast_rtp_prop_set: 
> >Setup RTCP on RTP instance '0x88c3b10'
> > == Using SIP RTP CoS mark 5
> >[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:4928 do_setnat: Setting NAT on RTP 
> >to Off
> >[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:4936 do_setnat: Setting NAT on 
> >UDPTL to Off
> >[Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:1459 
> >ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 
> >'SIP/pstn-1270-0003' with that of
> >'SIP/11-0002'
> >[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: 
> >Not copying variable DIALEDTIME.
> >[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: 
> >Not copying variable ANSWEREDTIME.
> >[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: 
> >Not copying variable DIALEDPEERNAME.
> >[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: 
> >Not copying variable DIALEDPEERNUMBER.
> >[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: 
> >Not copying variable DIALSTATUS.
> >[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: 
> >Not copying variable SIPCALLID.
> >[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: 
> >Not copying variable SIPDOMAIN.
> >[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: 
> >Not copying variable SIPURI.
> >[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:5463 sip_call: Outgoing Call for 
> >901148746612254
> >[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:10989 add_sdp: ** Our capability: 
> >0xc (ulaw|alaw) Video flag: False Text flag: False
> >[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:10990 add_sdp: ** Our prefcodec: 
> >0x4 (ulaw)
> >[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:3054 initialize_initreq: 
> >Initializing initreq for method INVITE - callid
> >770d283f78ef7d00782d2dd043212ed2@10.0.0.103:5060
> > -- Called SIP/901148746612254@pstn-1270
> >[Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional) 
> >Stopping retransmission (but retaining packet) on
> >'770d283f78ef7d00782d2dd043212ed2@10.0.0.103:5060' Request 102: Found
> >[Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 

Re: [asterisk-users] Voicemail config

[Bryant Zimmerman]

I would not use voicemail for this.

I would do the following.


Have the call come in. 

Ring the on call tech with a dial.

If they don't pickup and press an accept key then.

Answer the call.

Record a message from the caller.

Use a script to e-mail the message to the tech. 

  (I would recommend you use some kind of ticketing system to e-mail to 
that way you could have a progression if the on call tech does not respond 
in a timely manner. ) 


This is just one possible way to address this. problem I can think of about 
10 others depending on the actual requirements.


Thanks


Bryant Zimmerman (ZK Tech Inc.)

616-855-1030 Ext. 2003



From: "Kelly opal" 

Sent: Tuesday, September 13, 2011 5:35 PM

To: asterisk-users@lists.digium.com

Subject: [asterisk-users] Voicemail config

Hi
Is there a way to use variables in voicemail.conf.
I want to have an oncall tech system. The tech oncall has his number and 
email set in astdb. When a tech call comes in the dial plan checks astdb 
and sets 2 variable ${oc} for the number and ${ocem} for the email. I can 
easily dial the number using the variable, but if the call should go to 
voicemail I am not getting the email. Below is my voicemail.conf entry for 
the mailbox. I can hard code my email and it works so I know there is 
nothing wrong on the system.
 
121 => 121,Tech Support,${ocem},1113334...@vtext.net
 
Any help would be greatly appreciated.
 
Kelly


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Re: [asterisk-users] Reporting for Asterisk Call Center

[Tarek Sawah]

actually Bilal, 
the Asteirsk CDR reports are placed on a different Database than the 
configurations .. you will need to install asterisk-addons which includes a 
module for cdr reporting to MYSQL DB. so you don't have to do the configs from 
the DB at all 
second.. in regards to the Flash Operator Panel you can have a look at a demo 
here: 
http://www.asternic.org/demo.php
its a nice web interface gives you a live look at your call center .. who is 
active who is idle .. how many in Queue .. who is online and who is offline.. 
what trunks are busy ... etc

third: theoretically you can set all Asterisk boxes to load from one database 
server (never done it myself).. actually it's one of the methods used for 
redundancy (somebody correct me if i'm wrong?). my only concern is with DB 
Management systems there is what we call it "LOCK" where a process locks the 
whole db or a part or it in order to do it's manipulation.. so i'm not sure if 
the database will be "locked" by one of the asterisk boxes when writing to it? 
which prevents the rest from writing to it at the same time?
regards



Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993



> Date: Tue, 13 Sep 2011 02:43:05 -0700
> From: bilmar...@yahoo.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Reporting for Asterisk Call Center
> 
> 
> Dear Tareq;
> 
> I am not using mysql, the configuration on the text configuratoin files and 
> the logs are existed under the directory (/var/log/asterisk).
> 
> Well, to use mysql: then it means the configuration will be also in the 
> database or I can use mysql only for reporting?
> 
> What is the Flash Operator?
> 
> By the way, I have another question if you can help me if you used the 
> database with sql, actually I was facing one time a case and maybe the 
> Database usage will help me if you can advise me:
> 
> If I have multiple Asterisk servers are running, and I need them to work 
> centralized (I mean from one configuration) so to work as one system, then if 
> I have database for configuration, I can acheive this by making all the 
> servers read and write the configuration from the database server? 
> 
> Thanks for your help Tareq.
> 
> Regards
> Bilal
> 
> 
> ---
> > 
> > those reports can be easiely extracted from the MYSQL
> > database my friend.. and you can add the Flash Operator
> > Panel if you want to monitor live activities like how many
> > in queue and how many ON CALL .. etc
> > 
> > anyway the Elastix is a stand alone distribution you can
> > find more info and downloads at : http://elastix.org/
> > regards
> > 
> > 
> > Tarek Sawah
> > 
> > Information Technology  Adviser
> > 
> > Integrated Digital Systems
> > 
> > CCNP, MCSE, RHCE, TELECOM
> > 
> > USA: +1 386 492 9993
> > 
> > 
> > 
> 
> 
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Re: [asterisk-users] Voicemail config

[Danny Nicholas]

If you change ${ocem} to ${ocem}@default, this will probably work as you
want it to.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelly opal
Sent: Tuesday, September 13, 2011 4:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voicemail config

 

Hi

Is there a way to use variables in voicemail.conf.

I want to have an oncall tech system. The tech oncall has his number and
email set in astdb. When a tech call comes in the dial plan checks astdb and
sets 2 variable ${oc} for the number and ${ocem} for the email. I can easily
dial the number using the variable, but if the call should go to voicemail I
am not getting the email. Below is my voicemail.conf entry for the mailbox.
I can hard code my email and it works so I know there is nothing wrong on
the system.

 

121 => 121,Tech Support,${ocem},1113334...@vtext.net

 

Any help would be greatly appreciated.

 

Kelly

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[asterisk-users] Voicemail config

[Kelly opal]

Hi
Is there a way to use variables in voicemail.conf.
I want to have an oncall tech system. The tech oncall has his number and email 
set in astdb. When a tech call comes in the dial plan checks astdb and sets 2 
variable ${oc} for the number and ${ocem} for the email. I can easily dial the 
number using the variable, but if the call should go to voicemail I am not 
getting the email. Below is my voicemail.conf entry for the mailbox. I can hard 
code my email and it works so I know there is nothing wrong on the system.

121 => 121,Tech Support,${ocem},1113334...@vtext.net

Any help would be greatly appreciated.

Kelly--
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Re: [asterisk-users] Question about voip.ms service.

[naren]

That's what I am hoping to do as well. Could you share some insight on how
you set up the DID on the voip.ms web site to forward to Asterisk using IAX?
In particular I am trying to find out where you set the url / ip address of
your asterisk installation on the voip.ms web site.

Thanks!

On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone wrote:

> I'm using them for inbound and outbound on Asterisk and FreeSwitch
>
> Sent from my iPhone
>
>
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Re: [asterisk-users] Determine negotiated codec in script

[Danny Nicholas]

"Sip show channels" will give you the active codec.  You can get the
information using an AGI or a system command.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Browning
Sent: Tuesday, September 13, 2011 4:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Determine negotiated codec in script

Sorry if this is an obvious question and perhaps my Google foo isn't right
on this one:

I have calls coming into an Asterisk server that may be using 2 different
codecs.  I am recording audio in both cases but the challenge is knowing
which codec was negotiated at call setup.  I need to pass the proper format
to the record command as the codecs cannot be transcoded and are only
supported for playback/record/passthru etc.

Is there some global variable present that I can look at for codec
identification?

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Re: [asterisk-users] Question about voip.ms service.

[Robert-iPhone]

I'm using them for inbound and outbound on Asterisk and FreeSwitch

Sent from my iPhone

On Sep 13, 2011, at 5:14 PM, "Danny Nicholas"  wrote:

> That’s what this part of extensions.conf should do:
> 
> ; inbound context example for your DID numbers, do not add the number 1 in 
> front
> 
>  
> 
> [voipms-inbound]
> 
> exten => 7863643011,1,Answer() ;your DID
> 
>  
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
> Sent: Tuesday, September 13, 2011 4:09 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question about voip.ms service.
> 
>  
> 
> Yup, that part I got. What I am not clear about is how to set up the DID to 
> go to my URI. When I select "manage DIDs" and click on the one I want to 
> change, I see the following options for routing the DID
> 
>  
> 
> x SIP/IAX - [main account] IAX2/10 <- with my account number
> 
> x SIP URI - SIP:mysi...@myuri.com:5060
> 
> x System - Hangup
> 
>  
> 
> There are several other options but they are not selectable for me because I 
> have not set up to use them.
> 
>  
> 
> I used to have the routing set to SIP URI where I was able to specify my URI 
> where the call was routed to. But with the SIP/IAX option I do not have that 
> ability. 
> 
>  
> 
> I am missing something fundamental here. My asterisk has the iax.conf and 
> extensions.conf entries ready to receive calls from voip.ms, but I don't know 
> how to tel voip.ms to send the calls to my asterisk with the IAX protocol. 
> 
>  
> 
> I understand this is probably a question for the voip.ms folks, but since a 
> couple of people mentioned earlier that they were rocking with IAX, I thought 
> it would be an easy question for them to point me in the right direction.
> 
>  
> 
> Thanks. 
> 
> On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel  
> wrote:
> 
> I was lurking in this conversation and I went to look more carefully
> at the voip.ms site. I found sample files at
> http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29
> 
> Hope that helps.
> 
> 
> 
> On Tue, Sep 13, 2011 at 3:59 PM, naren  wrote:
> > I see the section you are talking about. It is on the home page if I am not
> > logged in. I see the Authentication section and the text "IAX/SIP
> > registration", but it doesn't seem to be a link. I am not sure how I can
> > find the page that has the details about the IAX/SIP registration. I see in
> > the wiki there is a page that has the configuration info for iax.conf and
> > extensions.conf.
> > Thanks for your help.
> > naren
> >
> > On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas  wrote:
> >>
> >> Did you read the “IAX/SIP registration” section (under Authentication) on
> >> voip.ms?
> >>
> >>
> >>
> >> From: asterisk-users-boun...@lists.digium.com
> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
> >> Sent: Tuesday, September 13, 2011 2:22 PM
> >> To: John Novack
> >> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] Question about voip.ms service.
> >>
> >>
> >>
> >> Ok... this is probably a dumb question but I can't figure out how to set
> >> voip.ms to use IAX for my DID... with SIP I was able to specify the URI so 
> >> I
> >> pointed it to my asterisk installation, but with IAX I don't have that
> >> option. Is that supposed to work some other way?
> >>
> >>
> >>
> >> Thanks a bunch!
> >>
> >> On Mon, Sep 12, 2011 at 11:18 PM, naren  wrote:
> >>
> >> I am novice with Asterisk, I had to piece together a lot of bits of info
> >> from lots of internet searches to get my very basic setup working. I
> >> probably shouldn't say that because it seems like Nat is not a very basic
> >> setup with Asterisk.
> >>
> >>
> >>
> >> The reason for wanting to stay with SIP is because I have my setup working
> >> with that protocol with an incoming and an outgoing line. I just wanted to
> >> add a second outgoing with voip.ms.
> >>
> >>
> >>
> >> But, I have come so far, so well why not... I will give IAX a shot, and
> >> see what traps I need to wade through :)
> >>
> >>
> >>
> >> Thanks
> >>
> >>
> >>
> >> On Mon, Sep 12, 2011 at 11:09 AM, John Novack
> >>  wrote:
> >>
> >> Never have had a problem with their IAX service.
> >>
> >> And ( for now ) a little hedge against the hackers.
> >>
> >> Since Asterisk is involved, why not use IAX anyway?
> >>
> >>
> >> John Novack
> >>
> >>
> >> naren wrote:
> >>
> >>
> >>
> >> I also found this... seems like voip.ms outbound is broken for now!
> >>
> >>
> >>
> >> http://pbxinaflash.com/forum/showthread.php?t=10735
> >>
> >>
> >>
> >>
> >>
> >> On Sun, Sep 11, 2011 at 10:34 PM, naren  wrote:
> >>
> >> Hi,
> >>
> >>
> >>
> >> I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem
> >> with the incoming, but my outgoing is not working. If at all possible, I
> >> would like to stick with SIP. Since the original poster (Glen) had 
> >> mentioned
> >> that he had

[asterisk-users] Determine negotiated codec in script

[Tom Browning]

Sorry if this is an obvious question and perhaps my Google foo isn't
right on this one:

I have calls coming into an Asterisk server that may be using 2
different codecs.  I am recording audio in both cases but the
challenge is knowing which codec was negotiated at call setup.  I need
to pass the proper format to the record command as the codecs cannot
be transcoded and are only supported for playback/record/passthru etc.

Is there some global variable present that I can look at for codec
identification?

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Re: [asterisk-users] Question about voip.ms service.

[Danny Nicholas]

That’s what this part of extensions.conf should do:

; inbound context example for your DID numbers, do not add the number 1 in front

 

[voipms-inbound]

exten => 7863643011,1,Answer() ;your DID

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
Sent: Tuesday, September 13, 2011 4:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about voip.ms service.

 

Yup, that part I got. What I am not clear about is how to set up the DID to go 
to my URI. When I select "manage DIDs" and click on the one I want to change, I 
see the following options for routing the DID

 

x SIP/IAX - [main account] IAX2/10 <- with my account number

x SIP URI - SIP:mysi...@myuri.com:5060

x System - Hangup

 

There are several other options but they are not selectable for me because I 
have not set up to use them.

 

I used to have the routing set to SIP URI where I was able to specify my URI 
where the call was routed to. But with the SIP/IAX option I do not have that 
ability. 

 

I am missing something fundamental here. My asterisk has the iax.conf and 
extensions.conf entries ready to receive calls from voip.ms, but I don't know 
how to tel voip.ms to send the calls to my asterisk with the IAX protocol. 

 

I understand this is probably a question for the voip.ms folks, but since a 
couple of people mentioned earlier that they were rocking with IAX, I thought 
it would be an easy question for them to point me in the right direction.

 

Thanks. 

On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel  wrote:

I was lurking in this conversation and I went to look more carefully
at the voip.ms site. I found sample files at
http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29

Hope that helps.



On Tue, Sep 13, 2011 at 3:59 PM, naren  wrote:
> I see the section you are talking about. It is on the home page if I am not
> logged in. I see the Authentication section and the text "IAX/SIP
> registration", but it doesn't seem to be a link. I am not sure how I can
> find the page that has the details about the IAX/SIP registration. I see in
> the wiki there is a page that has the configuration info for iax.conf and
> extensions.conf.
> Thanks for your help.
> naren
>
> On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas  wrote:
>>
>> Did you read the “IAX/SIP registration” section (under Authentication) on
>> voip.ms?
>>
>>
>>
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
>> Sent: Tuesday, September 13, 2011 2:22 PM
>> To: John Novack
>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Question about voip.ms service.
>>
>>
>>
>> Ok... this is probably a dumb question but I can't figure out how to set
>> voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I
>> pointed it to my asterisk installation, but with IAX I don't have that
>> option. Is that supposed to work some other way?
>>
>>
>>
>> Thanks a bunch!
>>
>> On Mon, Sep 12, 2011 at 11:18 PM, naren  wrote:
>>
>> I am novice with Asterisk, I had to piece together a lot of bits of info
>> from lots of internet searches to get my very basic setup working. I
>> probably shouldn't say that because it seems like Nat is not a very basic
>> setup with Asterisk.
>>
>>
>>
>> The reason for wanting to stay with SIP is because I have my setup working
>> with that protocol with an incoming and an outgoing line. I just wanted to
>> add a second outgoing with voip.ms.
>>
>>
>>
>> But, I have come so far, so well why not... I will give IAX a shot, and
>> see what traps I need to wade through :)
>>
>>
>>
>> Thanks
>>
>>
>>
>> On Mon, Sep 12, 2011 at 11:09 AM, John Novack
>>  wrote:
>>
>> Never have had a problem with their IAX service.
>>
>> And ( for now ) a little hedge against the hackers.
>>
>> Since Asterisk is involved, why not use IAX anyway?
>>
>>
>> John Novack
>>
>>
>> naren wrote:
>>
>>
>>
>> I also found this... seems like voip.ms outbound is broken for now!
>>
>>
>>
>> http://pbxinaflash.com/forum/showthread.php?t=10735
>>
>>
>>
>>
>>
>> On Sun, Sep 11, 2011 at 10:34 PM, naren  wrote:
>>
>> Hi,
>>
>>
>>
>> I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem
>> with the incoming, but my outgoing is not working. If at all possible, I
>> would like to stick with SIP. Since the original poster (Glen) had mentioned
>> that he had gotten outgoing working, I was wondering if you would be kind
>> enough to post some thoughts on that. Were you able to get it working with
>> just the default example sip.conf / extensions.conf settings that they have
>> on their website?
>>
>>
>>
>> I have pretty much the same settings. When I dial out, the destination
>> rings, but I can't hear a ringback tone from on the source side ( I am using
>> a PAP2T router with a phone). I have set up outgoing with actionvoip before
>> and that i

Re: [asterisk-users] Question about voip.ms service.

[naren]

Yup, that part I got. What I am not clear about is how to set up the DID to
go to my URI. When I select "manage DIDs" and click on the one I want to
change, I see the following options for routing the DID

x SIP/IAX - [main account] IAX2/10 <- with my account number
x SIP URI - SIP:mysi...@myuri.com:5060
x System - Hangup

There are several other options but they are not selectable for me because I
have not set up to use them.

I used to have the routing set to SIP URI where I was able to specify my URI
where the call was routed to. But with the SIP/IAX option I do not have that
ability.

I am missing something fundamental here. My asterisk has the iax.conf and
extensions.conf entries ready to receive calls from voip.ms, but I don't
know how to tel voip.ms to send the calls to my asterisk with the IAX
protocol.

I understand this is probably a question for the voip.ms folks, but since a
couple of people mentioned earlier that they were rocking with IAX, I
thought it would be an easy question for them to point me in the right
direction.

Thanks.

On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel wrote:

> I was lurking in this conversation and I went to look more carefully
> at the voip.ms site. I found sample files at
> http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29
>
> Hope that helps.
>
>
> On Tue, Sep 13, 2011 at 3:59 PM, naren  wrote:
> > I see the section you are talking about. It is on the home page if I am
> not
> > logged in. I see the Authentication section and the text "IAX/SIP
> > registration", but it doesn't seem to be a link. I am not sure how I can
> > find the page that has the details about the IAX/SIP registration. I see
> in
> > the wiki there is a page that has the configuration info for iax.conf and
> > extensions.conf.
> > Thanks for your help.
> > naren
> >
> > On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas 
> wrote:
> >>
> >> Did you read the “IAX/SIP registration” section (under Authentication)
> on
> >> voip.ms?
> >>
> >>
> >>
> >> From: asterisk-users-boun...@lists.digium.com
> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
> >> Sent: Tuesday, September 13, 2011 2:22 PM
> >> To: John Novack
> >> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] Question about voip.ms service.
> >>
> >>
> >>
> >> Ok... this is probably a dumb question but I can't figure out how to set
> >> voip.ms to use IAX for my DID... with SIP I was able to specify the URI
> so I
> >> pointed it to my asterisk installation, but with IAX I don't have that
> >> option. Is that supposed to work some other way?
> >>
> >>
> >>
> >> Thanks a bunch!
> >>
> >> On Mon, Sep 12, 2011 at 11:18 PM, naren  wrote:
> >>
> >> I am novice with Asterisk, I had to piece together a lot of bits of info
> >> from lots of internet searches to get my very basic setup working. I
> >> probably shouldn't say that because it seems like Nat is not a very
> basic
> >> setup with Asterisk.
> >>
> >>
> >>
> >> The reason for wanting to stay with SIP is because I have my setup
> working
> >> with that protocol with an incoming and an outgoing line. I just wanted
> to
> >> add a second outgoing with voip.ms.
> >>
> >>
> >>
> >> But, I have come so far, so well why not... I will give IAX a shot, and
> >> see what traps I need to wade through :)
> >>
> >>
> >>
> >> Thanks
> >>
> >>
> >>
> >> On Mon, Sep 12, 2011 at 11:09 AM, John Novack
> >>  wrote:
> >>
> >> Never have had a problem with their IAX service.
> >>
> >> And ( for now ) a little hedge against the hackers.
> >>
> >> Since Asterisk is involved, why not use IAX anyway?
> >>
> >>
> >> John Novack
> >>
> >>
> >> naren wrote:
> >>
> >>
> >>
> >> I also found this... seems like voip.ms outbound is broken for now!
> >>
> >>
> >>
> >> http://pbxinaflash.com/forum/showthread.php?t=10735
> >>
> >>
> >>
> >>
> >>
> >> On Sun, Sep 11, 2011 at 10:34 PM, naren  wrote:
> >>
> >> Hi,
> >>
> >>
> >>
> >> I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem
> >> with the incoming, but my outgoing is not working. If at all possible, I
> >> would like to stick with SIP. Since the original poster (Glen) had
> mentioned
> >> that he had gotten outgoing working, I was wondering if you would be
> kind
> >> enough to post some thoughts on that. Were you able to get it working
> with
> >> just the default example sip.conf / extensions.conf settings that they
> have
> >> on their website?
> >>
> >>
> >>
> >> I have pretty much the same settings. When I dial out, the destination
> >> rings, but I can't hear a ringback tone from on the source side ( I am
> using
> >> a PAP2T router with a phone). I have set up outgoing with actionvoip
> before
> >> and that is working fine, so I am thinking my router settings for my
> ports
> >> are correct - but I am no expert.
> >>
> >>
> >>
> >> I would really appreciate it if you could post the relevant section of
> >> your sip.conf for me.
> >>
> >>
> >>
> >> Thanks!
> >>
> >> Naren
> 

Re: [asterisk-users] Send DTMF

[Danny Nicholas]

http://www.voip-info.org/wiki/view/Asterisk+cmd+SendDTMF

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ezequiel Lovelle
Sent: Tuesday, September 13, 2011 3:37 PM
To: Asterisk Users
Subject: [asterisk-users] Send DTMF

 

¿How can i could to Send DTMF digits on a current call by scripting or similar 
methods?

 

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[asterisk-users] Send DTMF

[Ezequiel Lovelle]

  

¿How can i could to Send DTMF digits on a current call by scripting
or similar methods? 

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Re: [asterisk-users] Question about voip.ms service.

[Dave Aibel]

I was lurking in this conversation and I went to look more carefully
at the voip.ms site. I found sample files at
http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29

Hope that helps.


On Tue, Sep 13, 2011 at 3:59 PM, naren  wrote:
> I see the section you are talking about. It is on the home page if I am not
> logged in. I see the Authentication section and the text "IAX/SIP
> registration", but it doesn't seem to be a link. I am not sure how I can
> find the page that has the details about the IAX/SIP registration. I see in
> the wiki there is a page that has the configuration info for iax.conf and
> extensions.conf.
> Thanks for your help.
> naren
>
> On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas  wrote:
>>
>> Did you read the “IAX/SIP registration” section (under Authentication) on
>> voip.ms?
>>
>>
>>
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
>> Sent: Tuesday, September 13, 2011 2:22 PM
>> To: John Novack
>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Question about voip.ms service.
>>
>>
>>
>> Ok... this is probably a dumb question but I can't figure out how to set
>> voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I
>> pointed it to my asterisk installation, but with IAX I don't have that
>> option. Is that supposed to work some other way?
>>
>>
>>
>> Thanks a bunch!
>>
>> On Mon, Sep 12, 2011 at 11:18 PM, naren  wrote:
>>
>> I am novice with Asterisk, I had to piece together a lot of bits of info
>> from lots of internet searches to get my very basic setup working. I
>> probably shouldn't say that because it seems like Nat is not a very basic
>> setup with Asterisk.
>>
>>
>>
>> The reason for wanting to stay with SIP is because I have my setup working
>> with that protocol with an incoming and an outgoing line. I just wanted to
>> add a second outgoing with voip.ms.
>>
>>
>>
>> But, I have come so far, so well why not... I will give IAX a shot, and
>> see what traps I need to wade through :)
>>
>>
>>
>> Thanks
>>
>>
>>
>> On Mon, Sep 12, 2011 at 11:09 AM, John Novack
>>  wrote:
>>
>> Never have had a problem with their IAX service.
>>
>> And ( for now ) a little hedge against the hackers.
>>
>> Since Asterisk is involved, why not use IAX anyway?
>>
>>
>> John Novack
>>
>>
>> naren wrote:
>>
>>
>>
>> I also found this... seems like voip.ms outbound is broken for now!
>>
>>
>>
>> http://pbxinaflash.com/forum/showthread.php?t=10735
>>
>>
>>
>>
>>
>> On Sun, Sep 11, 2011 at 10:34 PM, naren  wrote:
>>
>> Hi,
>>
>>
>>
>> I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem
>> with the incoming, but my outgoing is not working. If at all possible, I
>> would like to stick with SIP. Since the original poster (Glen) had mentioned
>> that he had gotten outgoing working, I was wondering if you would be kind
>> enough to post some thoughts on that. Were you able to get it working with
>> just the default example sip.conf / extensions.conf settings that they have
>> on their website?
>>
>>
>>
>> I have pretty much the same settings. When I dial out, the destination
>> rings, but I can't hear a ringback tone from on the source side ( I am using
>> a PAP2T router with a phone). I have set up outgoing with actionvoip before
>> and that is working fine, so I am thinking my router settings for my ports
>> are correct - but I am no expert.
>>
>>
>>
>> I would really appreciate it if you could post the relevant section of
>> your sip.conf for me.
>>
>>
>>
>> Thanks!
>>
>> Naren
>>
>>
>>
>> On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards 
>> wrote:
>>
>> On Thu, 9 Jun 2011, John Novack wrote:
>>
>> I use voip.ms and have no issues using IAX and Asterisk 1.4.xx
>>
>>
>>
>> 'slam-dunk.'
>>
>>
>>
>> Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall
>>
>> a
>>
>> Their on line config samples just work!
>>
>>
>>
>> is
>>
>>
>>
>> Suggest you check your firewall and your configs, and above all post some
>> more information
>>
>>
>>
>> IAX
>>
>>
>>
>> If you really want to upset some, top post as I have just done!
>>
>>
>>
>> Agreed.
>>
>>
>>
>> The real issue is communication, top bottom or in the middle
>>
>>
>>
>> Sometimes, it's just about being considerate to 'the next guy.'
>>
>> --
>> Thanks in advance,
>> -
>> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
>> Newline                                              Fax: +1-760-731-3000
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>              http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk

Re: [asterisk-users] Question about voip.ms service.

[Danny Nicholas]

I see what you mean.  Maybe if you call their support they can tell you what 
you need to know. If not, voicepulse is a pretty good provider.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
Sent: Tuesday, September 13, 2011 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about voip.ms service.

 

I see the section you are talking about. It is on the home page if I am not 
logged in. I see the Authentication section and the text "IAX/SIP 
registration", but it doesn't seem to be a link. I am not sure how I can find 
the page that has the details about the IAX/SIP registration. I see in the wiki 
there is a page that has the configuration info for iax.conf and 
extensions.conf. 

 

Thanks for your help.

naren

 

On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas  wrote:

Did you read the “IAX/SIP registration” section (under Authentication) on 
voip.ms? 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
Sent: Tuesday, September 13, 2011 2:22 PM
To: John Novack
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about voip.ms service.

 

Ok... this is probably a dumb question but I can't figure out how to set 
voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I 
pointed it to my asterisk installation, but with IAX I don't have that option. 
Is that supposed to work some other way?

 

Thanks a bunch!

On Mon, Sep 12, 2011 at 11:18 PM, naren  wrote:

I am novice with Asterisk, I had to piece together a lot of bits of info from 
lots of internet searches to get my very basic setup working. I probably 
shouldn't say that because it seems like Nat is not a very basic setup with 
Asterisk.

 

The reason for wanting to stay with SIP is because I have my setup working with 
that protocol with an incoming and an outgoing line. I just wanted to add a 
second outgoing with voip.ms. 

 

But, I have come so far, so well why not... I will give IAX a shot, and see 
what traps I need to wade through :)

 

Thanks

 

On Mon, Sep 12, 2011 at 11:09 AM, John Novack  
wrote:

Never have had a problem with their IAX service.

And ( for now ) a little hedge against the hackers.

Since Asterisk is involved, why not use IAX anyway?


John Novack




naren wrote: 

 

I also found this... seems like voip.ms outbound is broken for now!

 

http://pbxinaflash.com/forum/showthread.php?t=10735

 

 

On Sun, Sep 11, 2011 at 10:34 PM, naren  wrote:

Hi, 

 

I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the 
incoming, but my outgoing is not working. If at all possible, I would like to 
stick with SIP. Since the original poster (Glen) had mentioned that he had 
gotten outgoing working, I was wondering if you would be kind enough to post 
some thoughts on that. Were you able to get it working with just the default 
example sip.conf / extensions.conf settings that they have on their website?

 

I have pretty much the same settings. When I dial out, the destination rings, 
but I can't hear a ringback tone from on the source side ( I am using a PAP2T 
router with a phone). I have set up outgoing with actionvoip before and that is 
working fine, so I am thinking my router settings for my ports are correct - 
but I am no expert.

 

I would really appreciate it if you could post the relevant section of your 
sip.conf for me.

 

Thanks!

Naren

 

On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards  wrote:

On Thu, 9 Jun 2011, John Novack wrote:

I use voip.ms and have no issues using IAX and Asterisk 1.4.xx

 

'slam-dunk.' 

 

Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall


a

Their on line config samples just work!

 

is 

 

Suggest you check your firewall and your configs, and above all post some more 
information

 

IAX 

 

If you really want to upset some, top post as I have just done!

 

Agreed. 

 

The real issue is communication, top bottom or in the middle

 

Sometimes, it's just about being considerate to 'the next guy.'

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



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello

asterisk-users mailing list
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New to As

Re: [asterisk-users] Question about voip.ms service.

[naren]

I see the section you are talking about. It is on the home page if I am not
logged in. I see the Authentication section and the text "IAX/SIP
registration", but it doesn't seem to be a link. I am not sure how I can
find the page that has the details about the IAX/SIP registration. I see in
the wiki there is a page that has the configuration info for iax.conf and
extensions.conf.

Thanks for your help.
naren


On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas  wrote:

> Did you read the “IAX/SIP registration” section (under Authentication) on
> voip.ms? 
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *naren
> *Sent:* Tuesday, September 13, 2011 2:22 PM
> *To:* John Novack
> *Cc:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Question about voip.ms service.
>
> ** **
>
> Ok... this is probably a dumb question but I can't figure out how to set
> voip.ms to use IAX for my DID... with SIP I was able to specify the URI so
> I pointed it to my asterisk installation, but with IAX I don't have that
> option. Is that supposed to work some other way?
>
> ** **
>
> Thanks a bunch!
>
> On Mon, Sep 12, 2011 at 11:18 PM, naren  wrote:
>
> I am novice with Asterisk, I had to piece together a lot of bits of info
> from lots of internet searches to get my very basic setup working. I
> probably shouldn't say that because it seems like Nat is not a very basic
> setup with Asterisk.
>
> ** **
>
> The reason for wanting to stay with SIP is because I have my setup working
> with that protocol with an incoming and an outgoing line. I just wanted to
> add a second outgoing with voip.ms. 
>
> ** **
>
> But, I have come so far, so well why not... I will give IAX a shot, and see
> what traps I need to wade through :)
>
> ** **
>
> Thanks
>
> ** **
>
> On Mon, Sep 12, 2011 at 11:09 AM, John Novack <
> jnov...@stromberg-carlson.org> wrote:
>
> Never have had a problem with their IAX service.
>
> And ( for now ) a little hedge against the hackers.
>
> Since Asterisk is involved, why not use IAX anyway?
>
>
> John Novack
>
>
>
>
> naren wrote: 
>
> ** **
>
> I also found this... seems like voip.ms outbound is broken for now!
>
> ** **
>
> http://pbxinaflash.com/forum/showthread.php?t=10735
>
> ** **
>
> ** **
>
> On Sun, Sep 11, 2011 at 10:34 PM, naren  wrote:
>
> Hi, 
>
> ** **
>
> I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem
> with the incoming, but my outgoing is not working. If at all possible, I
> would like to stick with SIP. Since the original poster (Glen) had mentioned
> that he had gotten outgoing working, I was wondering if you would be kind
> enough to post some thoughts on that. Were you able to get it working with
> just the default example sip.conf / extensions.conf settings that they have
> on their website?
>
> ** **
>
> I have pretty much the same settings. When I dial out, the destination
> rings, but I can't hear a ringback tone from on the source side ( I am using
> a PAP2T router with a phone). I have set up outgoing with actionvoip before
> and that is working fine, so I am thinking my router settings for my ports
> are correct - but I am no expert.
>
> ** **
>
> I would really appreciate it if you could post the relevant section of your
> sip.conf for me.
>
> ** **
>
> Thanks!
>
> Naren
>
> ** **
>
> On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards 
> wrote:
>
> On Thu, 9 Jun 2011, John Novack wrote:
>
> I use voip.ms and have no issues using IAX and Asterisk 1.4.xx
>
> ** **
>
> 'slam-dunk.' 
>
> ** **
>
> Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall
> 
>
>
> a
>
> Their on line config samples just work!
>
> ** **
>
> is 
>
> ** **
>
> Suggest you check your firewall and your configs, and above all post some
> more information
>
> ** **
>
> IAX 
>
> ** **
>
> If you really want to upset some, top post as I have just done!
>
> ** **
>
> Agreed. 
>
> ** **
>
> The real issue is communication, top bottom or in the middle
>
> ** **
>
> Sometimes, it's just about being considerate to 'the next guy.'
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
> 
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>  http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ** **
>
>
>
> 
>
> --
>
> 

Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Leif Madsen]

On 12/09/11 02:21 PM, linux guy wrote:

I'm about to start building my asterisk server and I can't seem to find
anything that discusses the pros and cons of installing the OS (Fedora
15) as console only or GUI, ie install KDE as well.


http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id291070

--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8

[Leif Madsen]

On 12/09/11 09:48 PM, Joseph wrote:

Was NAT problem fixed in 1.8.7 ? I'm using 1.4 but I've tried 1.6 and
1.8.5 and in both cases connection is not working with my provider with
SIP + NAT.
The connection is showing up as registered but the call is not coming IN
(congestion).


Can you define "NAT problem"? I'm unaware of any issues with Asterisk 
(or end points) behind NAT. It is mostly likely a configuration issue 
rather than a bug.


--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] Question about voip.ms service.

[Danny Nicholas]

Did you read the “IAX/SIP registration” section (under Authentication) on 
voip.ms? 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
Sent: Tuesday, September 13, 2011 2:22 PM
To: John Novack
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about voip.ms service.

 

Ok... this is probably a dumb question but I can't figure out how to set 
voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I 
pointed it to my asterisk installation, but with IAX I don't have that option. 
Is that supposed to work some other way?

 

Thanks a bunch!

On Mon, Sep 12, 2011 at 11:18 PM, naren  wrote:

I am novice with Asterisk, I had to piece together a lot of bits of info from 
lots of internet searches to get my very basic setup working. I probably 
shouldn't say that because it seems like Nat is not a very basic setup with 
Asterisk.

 

The reason for wanting to stay with SIP is because I have my setup working with 
that protocol with an incoming and an outgoing line. I just wanted to add a 
second outgoing with voip.ms. 

 

But, I have come so far, so well why not... I will give IAX a shot, and see 
what traps I need to wade through :)

 

Thanks

 

On Mon, Sep 12, 2011 at 11:09 AM, John Novack  
wrote:

Never have had a problem with their IAX service.

And ( for now ) a little hedge against the hackers.

Since Asterisk is involved, why not use IAX anyway?


John Novack




naren wrote: 

 

I also found this... seems like voip.ms outbound is broken for now!

 

http://pbxinaflash.com/forum/showthread.php?t=10735

 

 

On Sun, Sep 11, 2011 at 10:34 PM, naren  wrote:

Hi, 

 

I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the 
incoming, but my outgoing is not working. If at all possible, I would like to 
stick with SIP. Since the original poster (Glen) had mentioned that he had 
gotten outgoing working, I was wondering if you would be kind enough to post 
some thoughts on that. Were you able to get it working with just the default 
example sip.conf / extensions.conf settings that they have on their website?

 

I have pretty much the same settings. When I dial out, the destination rings, 
but I can't hear a ringback tone from on the source side ( I am using a PAP2T 
router with a phone). I have set up outgoing with actionvoip before and that is 
working fine, so I am thinking my router settings for my ports are correct - 
but I am no expert.

 

I would really appreciate it if you could post the relevant section of your 
sip.conf for me.

 

Thanks!

Naren

 

On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards  wrote:

On Thu, 9 Jun 2011, John Novack wrote:

I use voip.ms and have no issues using IAX and Asterisk 1.4.xx

 

'slam-dunk.' 

 

Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall


a

Their on line config samples just work!

 

is 

 

Suggest you check your firewall and your configs, and above all post some more 
information

 

IAX 

 

If you really want to upset some, top post as I have just done!

 

Agreed. 

 

The real issue is communication, top bottom or in the middle

 

Sometimes, it's just about being considerate to 'the next guy.'

-- 
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] Question about voip.ms service.

[naren]

Ok... this is probably a dumb question but I can't figure out how to set
voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I
pointed it to my asterisk installation, but with IAX I don't have that
option. Is that supposed to work some other way?

Thanks a bunch!

On Mon, Sep 12, 2011 at 11:18 PM, naren  wrote:

> I am novice with Asterisk, I had to piece together a lot of bits of info
> from lots of internet searches to get my very basic setup working. I
> probably shouldn't say that because it seems like Nat is not a very basic
> setup with Asterisk.
>
> The reason for wanting to stay with SIP is because I have my setup working
> with that protocol with an incoming and an outgoing line. I just wanted to
> add a second outgoing with voip.ms.
>
> But, I have come so far, so well why not... I will give IAX a shot, and see
> what traps I need to wade through :)
>
> Thanks
>
>
> On Mon, Sep 12, 2011 at 11:09 AM, John Novack <
> jnov...@stromberg-carlson.org> wrote:
>
>>  Never have had a problem with their IAX service.
>>
>> And ( for now ) a little hedge against the hackers.
>>
>> Since Asterisk is involved, why not use IAX anyway?
>>
>>
>> John Novack
>>
>>
>>
>> naren wrote:
>>
>>
>>  I also found this... seems like voip.ms outbound is broken for now!
>>
>>  http://pbxinaflash.com/forum/showthread.php?t=10735
>>
>>
>>
>> On Sun, Sep 11, 2011 at 10:34 PM, naren  wrote:
>>
>>> Hi,
>>>
>>>  I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem
>>> with the incoming, but my outgoing is not working. If at all possible, I
>>> would like to stick with SIP. Since the original poster (Glen) had mentioned
>>> that he had gotten outgoing working, I was wondering if you would be kind
>>> enough to post some thoughts on that. Were you able to get it working with
>>> just the default example sip.conf / extensions.conf settings that they have
>>> on their website?
>>>
>>>  I have pretty much the same settings. When I dial out, the destination
>>> rings, but I can't hear a ringback tone from on the source side ( I am using
>>> a PAP2T router with a phone). I have set up outgoing with actionvoip before
>>> and that is working fine, so I am thinking my router settings for my ports
>>> are correct - but I am no expert.
>>>
>>>  I would really appreciate it if you could post the relevant section of
>>> your sip.conf for me.
>>>
>>>  Thanks!
>>> Naren
>>>
>>>
>>>  On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards <
>>> asterisk@sedwards.com> wrote:
>>>
 On Thu, 9 Jun 2011, John Novack wrote:

  I use voip.ms and have no issues using IAX and Asterisk 1.4.xx
>

  'slam-dunk.'


  Though they suggest SIP, I chose IAX and have 4569 UDP open in my
> firewall
>

 a

  Their on line config samples just work!
>

  is


  Suggest you check your firewall and your configs, and above all post
> some more information
>

  IAX


  If you really want to upset some, top post as I have just done!
>

  Agreed.


  The real issue is communication, top bottom or in the middle
>

  Sometimes, it's just about being considerate to 'the next guy.'

 --
 Thanks in advance,

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


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 asterisk-users mailing list
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  http://lists.digium.com/mailman/listinfo/asterisk-users

>>>
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> --
>>
>> Dog is my Co-pilot
>>
>>
>
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Re: [asterisk-users] High delay from Asterisk as PSTN simulator

[Kevin P. Fleming]

On 09/13/2011 08:56 AM, Gustavo Santos wrote:

I'm trying to use Asterisk as a PSTN simulator to run performance tests
for echo cancellation algorithms. I'm using the following configuration:

SIP <-> Asterisk 1 <> Asterisk 2 <> Echo()

Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan
application.

The problem is the high delay using this configuration: 20 ms only in
Asterisk 2. I've read the source code of chan_dahdi, and I think the
channel has a 20 ms "buffer" (160 samples). Algorithms like mg2 and kb1
are configured to accept 128 taps (16 ms), so 20 ms is too high.

Someone knows how I can reduce the delay to at least 10 ms? Should I
change something in the source code?


20 milliseconds is far from a 'high' (long) delay. Asterisk handles 
audio in packets, it does not directly switch TDM streams. As a result, 
there is always going to be (at least) the delay of one packet time for 
audio passing into Asterisk and back out via the Echo() application. 
This is unavoidable.


An alternative solution would be to send a call into Asterisk2 and have 
it dial back to Asterisk1 (and then back to the originating endpoint) 
and bridge those two calls in Asterisk2; if both calls are on the same 
E1, then Asterisk will let the DAHDI hardware directly connect the two 
channels, resulting in a 1 or 2 millisecond delay.


But realistically... configuring an echo canceller with only a 16ms 
window of operation is not very practical. Sending a call through *any* 
network element that packetizes the audio will result in a delay longer 
than 16ms.


--
Kevin P. Fleming
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] Sip profiles per customer, behind a SIP proxy. How?

[Kevin P. Fleming]

On 09/10/2011 09:16 PM, Robert Thomas wrote:

Hello List,

I have been trying to configure  a sip profile ( peer / friend ) for
each of my customers behind a sip proxy for some time, but I have had no
success, so I would appreciate your help.

Customer  -> OpenSIPS -> Asterisk -> PSTN

The opensips is working as a sip proxy with record route, for billing,
load balancing and authentication purposes.

I would like to be able to define a particular context, or settings per
customer. Since the customer is behind the SIP proxy, all I see if the
packet comming from the SIP proxy. So I have created a peer profile with
the IP Address of my proxy. Problem been any setting I enable affect all
traffic coming throught the SIP proxy.

I was reading that Asterisk checks the SIP From: address username and
matches against names of devices with type=user-

However I have some problems with Asterisk 1.6.2, taking the caller id
either from the RPID so I manually parse the PAI header.

I was thinking about replacing the From with a customer ID, and for
those customers that use the FROM to signal caller id, to copy it over
the PAI header at the SIP proxy.

I don't know if overwriting the FROM would cause any problem with the
SIP clients behind the proxy.

Is there any better or different way to acomplish this?

I would say we should have some flexibility although we have a SIP proxy
as the source IP for all of our traffic.


Olle Johansson has a developer branch that includes a method to do 
exactly what you are looking for; I suggest you look him up and find out 
what state it is in, and see whether you can help test it.


--
Kevin P. Fleming
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] DMTF via rfc2833 and SIP-INFO simultaneously

[virendra bhati]

Hi ,

What was the solution of that problem ? Did provider change the setting at
there end or else ?

On Tue, Sep 13, 2011 at 7:37 PM, Kristijan Vrban
wrote:

> hello Virendra,
>
> thx for your response. but after i made clear to the carrier that i
> want the dmtf only via rfc2833
> and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed.
>
> Kristijan
>
> 2011/9/13 virendra bhati :
> > Hi
> > 1st check that how many manager is connected into the server. 1 or more
> then
> > you can say that 2 DTMF is capture by asterisk for same events.
> >
> >  manager show connected
> >   Username IP Address
> >   root 127.0.0.1
> >
> > it should be one only.
> >
> > I face the same case then I found that more then 1 manager was working
> into
> > the server.
> >
> >
> > On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban <
> vrban.l...@googlemail.com>
> > wrote:
> >>
> >> Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO
> >> simultaneously. That has the effect, that asterisk read every dtmf
> >> twice. and yes, it's mainly the carriers mistake. but is there a
> >> configure option, that asterisk accept only one DMTF method for
> >> inbound dtmf?
> >>
> >> Kristijan
> >>
> >> --
> >> _
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> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > --
> >
> >
> >
> > -
> > Thanks and regards
> >
> >  Virendra Bhati
> > +91-9172341457
> > Software Engineer
> >
> >
> > --
> > _
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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-- 



-
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] PRI Issues After Upgrade

[Doug Lytle]

Stephen H. Gerstacker wrote:

Anything else I can try?


Try switchtype=national just for testing.

Doug


--

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deserve neither Liberty nor Safety."


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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

I disabled the echo cancelled on the PRI and the same issues are still popping 
up:

PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/22 got hangup, cause 16

Anything else I can try?

Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext. 114
Fax: 866.528.3854
www.edpaymentsystems.com

On Sep 11, 2011, at 9:52 AM, Stephen H. Gerstacker wrote:

When we moved buildings, the PRI provider specifically asked to switch to 
dms100. That's how the old server was as well.

I'll try the echo canceller first.

- Stephen H. Gerstacker

On Sep 11, 2011, at 9:07, "Doug Lytle" 
mailto:supp...@drdos.info>> wrote:


Stephen H. Gerstacker wrote:

switchtype=dms100


Are you sure that your switchtype is correct and your provider has a dms100?  
If not, change this to national.

If so, the two things I'd try to narrow it down are:

1.)  Temporarily remove the AEX410 card and test again
2.)  Temporarily disable OSLEC for your echo canceller and test again.

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] High delay from Asterisk as PSTN simulator

[Gustavo Santos]

I'm trying to use Asterisk as a PSTN simulator to run performance tests for
echo cancellation algorithms. I'm using the following configuration:

SIP <-> Asterisk 1 <> Asterisk 2 <> Echo()

Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan
application.

The problem is the high delay using this configuration: 20 ms only in
Asterisk 2. I've read the source code of chan_dahdi, and I think the channel
has a 20 ms "buffer" (160 samples). Algorithms like mg2 and kb1 are
configured to accept 128 taps (16 ms), so 20 ms is too high.

Someone knows how I can reduce the delay to at least 10 ms? Should I change
something in the source code?

Thanks in advance,
Gustavo Santos.

-- 
Atenciosamente,
Gustavo Santos.
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Re: [asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously

[Kristijan Vrban]

hello Virendra,

thx for your response. but after i made clear to the carrier that i
want the dmtf only via rfc2833
and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed.

Kristijan

2011/9/13 virendra bhati :
> Hi
> 1st check that how many manager is connected into the server. 1 or more then
> you can say that 2 DTMF is capture by asterisk for same events.
>
>  manager show connected
>   Username IP Address
>   root 127.0.0.1
>
> it should be one only.
>
> I face the same case then I found that more then 1 manager was working into
> the server.
>
>
> On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban 
> wrote:
>>
>> Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO
>> simultaneously. That has the effect, that asterisk read every dtmf
>> twice. and yes, it's mainly the carriers mistake. but is there a
>> configure option, that asterisk accept only one DMTF method for
>> inbound dtmf?
>>
>> Kristijan
>>
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>
>
>
> --
>
>
>
> -
> Thanks and regards
>
>  Virendra Bhati
> +91-9172341457
> Software Engineer
>
>
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Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Matthew J. Roth]

Kaushal,

Your version of SoX does not have MP3 support.  Since you have LAME
installed, use it as a first step to produce an intermediate file
that SoX supports.  Then use SoX to convert the intermediate file
to the desired format.

Step 1
--

# lame --decode obd-demo.mp3 obd-demo.wav
input:  obd-demo.mp3  (8 kHz, 1 channel, MPEG-2.5 Layer III)
output: obd-demo.wav  (16 bit, Microsoft WAVE)
skipping initial 1105 samples (encoder+decoder delay)
Frame# 16818/16818   16 kbps
# file obd-demo.wav
obd-demo.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, 
mono 8000 Hz


Step 2
--

# sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.ulaw
sox: Detected file format type: wav

sox: WAV Chunk fmt
sox: WAV Chunk data
sox: Reading Wave file: Microsoft PCM format, 1 channel, 8000 samp/sec
sox: 16000 byte/sec, 2 block align, 16 bits/samp, 19372126 data bytes
sox: 9686063 Samps/chans
sox: Input file obd-demo.wav: using sample rate 8000
size shorts, encoding signed (2's complement), 1 channel
sox: Output file obd-demo.ulaw: using sample rate 8000
size bytes, encoding u-law, 1 channel
sox: Output file: comment "Processed by SoX"


Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468)

[bilal ghayyad]

Hi All;

Asterisk version is: 1.8.5.0

But I see at the consol the following warning and really I did google but did 
not understand if it is bug or related to settings:

[Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just 
did sched_add waitid(3429468) for sip_reinvite_retry for dialog 
3c581fa96f2b-53yysntgjmwb in handle_response_invite

But actually, we see some SNOM IP Phones has NR (Not Register) at the LCD, and 
it is able to receive and originate calls !!

I was think if this is bug or if it is related to session expire .. but I am 
not able to determine until now. Any help?

By the way: which paramter in the sip.conf can be used to determine the timeout 
of the sip registration (so the IP Phone should send the registartion packet to 
keepalive within this timeout, otherwise it will be considered not register)? 
Is it the defaultexpiry or something else?

Also, the above warning, to what it could be related? Is it a bug?

Regards
Bilal

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Re: [asterisk-users] SIP Realtime & Templates (!)

[Ishfaq Malik]

Hi

To the best of my knowledge there isn't.

But, if you're using realtime you can create a program to add your
extensions to the database and you can create the concept of templates
within that.

Regards

Ish

On Tue, 2011-09-13 at 12:27 +0200, Alexandru Oniciuc wrote:
> Hello,
> 
>  
> 
> Is it possible to assign templates defined in sip.conf to sip realtime
> peers? 
> 
> There was another mail in 2008 which asked the same question but never
> received a response.
> 
>  
> 
> Thanks,
> 
> Alex
> 
> 
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-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] SIP Realtime & Templates (!)

[Alexandru Oniciuc]

Hello,

Is it possible to assign templates defined in sip.conf to sip realtime peers?
There was another mail in 2008 which asked the same question but never received 
a response.

Thanks,
Alex
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Re: [asterisk-users] Reporting for Asterisk Call Center

[bilal ghayyad]

Dear Tareq;

I am not using mysql, the configuration on the text configuratoin files and the 
logs are existed under the directory (/var/log/asterisk).

Well, to use mysql: then it means the configuration will be also in the 
database or I can use mysql only for reporting?

What is the Flash Operator?

By the way, I have another question if you can help me if you used the database 
with sql, actually I was facing one time a case and maybe the Database usage 
will help me if you can advise me:

If I have multiple Asterisk servers are running, and I need them to work 
centralized (I mean from one configuration) so to work as one system, then if I 
have database for configuration, I can acheive this by making all the servers 
read and write the configuration from the database server? 

Thanks for your help Tareq.

Regards
Bilal


---
> 
> those reports can be easiely extracted from the MYSQL
> database my friend.. and you can add the Flash Operator
> Panel if you want to monitor live activities like how many
> in queue and how many ON CALL .. etc
> 
> anyway the Elastix is a stand alone distribution you can
> find more info and downloads at : http://elastix.org/
> regards
> 
> 
> Tarek Sawah
> 
> Information Technology  Adviser
> 
> Integrated Digital Systems
> 
> CCNP, MCSE, RHCE, TELECOM
> 
> USA: +1 386 492 9993
> 
> 
> 


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Re: [asterisk-users] broadcast

[Sam Govind]

I don't know why you are running into problems.
Once a call file is executed it creates two legs (according to call file
structure) A leg is "Channel: Local/1234@conference" and once it Answers the
call file the second leg is bridged which should be
Context->Extension->priority. So what I'm asking is make your conference
A-leg and your Playback/messages dial plan B-leg.

take a look at the changes I made to your dial-plan

[conference]
exten => 1234,1,Answer()
exten => 1234,n,Gotoif($[${FIRST-CALLER} > 1]?startmsg:pass)
exten => 1234,n(startmsg),System(echo -e
"Channel:local/1234@conference\\nContext: contest-call\\nExtension:
23\\nPriority: 1" > /tmp/${UNIQUEID}.call)
exten => 1234,n,system(mv /tmp/${UNIQUEID}.call
/var/spool/asterisk/outgoing)
exten => 1234,n(pass),Konference(43689956,ADMRSTVL)
exten => 1234,n,Hangup()

[contest-call]
exten => 23,1,Answer()
exten => 23,n,Set(p="/var/spool/asterisk/monitor/")
exten => 23,n,playback(${p}/LQA/12/Biology/Que3)
exten => 23,n,playback(${p}/LQA/12/Biology/Que4)
exten => 23,n,playback(${p}/LQA/12/Biology/Que5)
exten => 23,n,playback(${p}/LQA/12/Biology/Que6)
exten => 23,n,playback(${p}/LQA/12/Biology/Que7)
exten => 23,n,Wait(10)
exten => 23,n,Hangup()

Here, changed your script to what I'm thinking. use the above tweak
accordingly. make sure to find out FIRST-CALLER so your tapes start playing
into conference just for once.

-Sammy

On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati  wrote:

> Hi List,
>
> I make a script for .call file and then I started playback on local channel
> but nothing was hearing at another channles.
>
> exten => 1234,1,Answer()
> exten => 1234,n,System(echo -e "Channel: Channel: 
> local/23@contest-call\\nContext:
> contest-call\\nExtension: 23\\nPriority: 1" > /tmp/${UNIQUEID}.call)
> exten => 1234,n,Konference(43689956,ADMRSTVL)
>
> [contest-call]
>
> exten => _X!,1,Answer()
> exten => _X!,n,Set(p="/var/spool/asterisk/monitor/")
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que3)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que4)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que5)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que6)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que7)
> exten => _X!,n,Konference(43689956,ADMRSTV)
> exten => _X!,n,Wait(10)
> exten => _X!,n,Hangup()
>
> in it I am dialing 1234 from softphone then join to conf in mute mode,
> after it .call file start playback at it's own channels but I am not able to
> hear anything into conf.
>
> As i know localdial is not joining into the conf. but how I will do it so
> that I will be able to hear any played file into conference ?
>
>
>
> On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind  wrote:
>
>> Good to know,
>>
>> I think it'll be a feedback score or a poll from members of the
>> conference. So if you use the R option and collect DTMF from members, and an
>> AMI script listening to that particular DTMF event collects all. This way
>> your AMI listener script should be able to tell you at the end of poll what
>> user inserted with DTMF.
>>
>> So overall insertion of a broadcast message using Ahmed's method of .call
>> file and later on collecting DTMF events from AMI script
>> should theoretically work for you.
>>
>> On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati wrote:
>>
>>> Hi Sam,
>>>
>>> You are right. I am looking for the same
>>>
>>> On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind  wrote:
>>>
 IMHO, I think Bhaati is trying to get feedback from
 multiple conference users. See DTMF options in Konference module.
  'R' : enable DTMF relay: DTMF tones generate a manager event
  If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
 members in the conference

 While some file is played and users press any DTMF collect the AMI
 events from each user and use them as you require.

 Ref:
 http://main.voiptoday.org/index.php?option=com_content&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-available&catid=35:general&Itemid=173


 On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati wrote:

> Hi Ahmed,
>
> Konference is also an conferencing application.
>
> On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed 
> wrote:
>
>> Hhhmmm..I dunt have any experience with module Konference. Maybe
>> anyone else can help you on that. 
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra
>> bhati
>> *Sent:* Monday, September 12, 2011 1:28 PM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] broadcast
>>
>>  ** **
>>
>> Hi Ahmed,
>>
>> I did the same thing earlier to test the load of Digium card. But this
>> time I want to play file and want to get some DTMF from all the members 
>> of
>> conference.
>>
>> So in this case I need more

Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Eric Wieling]

"sox -h" will list the formats supported by your install of sox.  If mp3 is not 
listed, then your sox does not support mp3.  This is not uncommon.  Many Linux 
distros do not ship support for patent encumbered formats.   Either stop using 
mp3 (this is what I suggest) or compile and install sox with mp3 support.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan
Sent: Monday, September 12, 2011 8:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand 
format type: mp3

Hi,

Can someone please comment about the below issue

[root@host0040 kaushal]# file obd-demo.mp3
obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural
[root@host0040 kaushal]# sox obd-demo.mp3 -e stat
sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[root@host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw
sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[root@host0040 kaushal]# sox -v 0.125 -V  -t au -r 8000 -U -b -c 
1  resample -ql
-bash: obd-demo.ulaw: No such file or directory
[root@host0040 kaushal]# sox -V obd-demo.mp3 -t au -r 8000 -U -b -c 1 
obd-demo.ulaw resample -ql
sox: Failed reading obd-demo.mp3: Do not understand foReply rmat type: mp3

[root@host0040 kaushal]#

When i invoke the same obd-demo.mp3 it works perfectly fine

host0040*CLI> channel originate DAHDI/g0/xx Application MP3Player 
/home/kaushal/obd-demo.mp3 [Sep  9 16:44:56] DEBUG[12691]: sig_pri.c:985 
sig_pri_request: sig_pri_request 1 [Sep  9 16:44:56] DEBUG[12691]: 
sig_pri.c:6427 sig_pri_call: CALLER NAME:  NUM:
  -- Requested transfer capability: 0x00 - SPEECH
  -- Launching MP3Player(/home/kaushal/obd-demo.mp3) on DAHDI/i1/9833754756-1

[root@host0040 ~]# rpm -qa | grep sox
sox-12.18.1-1.el5_5.1
[root@host0040 ~]# rpm -qa | grep lame
lame-3.98.4-1.el5.rf
lame-devel-3.98.4-1.el5.rf
[root@host0040 ~]#


MP3 support in  SoX  is  optional
and requires access to either or both the external libmad and 
libmp3lame libraries.  To see if there is support for Mp3 run sox -h and
look for it under the list of supported file formats as "mp3".

[root@host0040 ~]# sox -h
sox: Version 12.18.1

Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ]

gopts: -e -h -p -q -S -V

fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x

effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax 
echo echos fade filter flanger highp highpass lowp lowpass mask mcompand 
noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb 
reverse silence speed stat stretch swap synth trim vibro vol

effopts: depends on effect

Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm hcom la lu 
maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis 
vox wav wve

Which package contains libmad and libmp3lame libraries available on CentOS 5.6

Regards,

Kaushal

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Re: [asterisk-users] Reporting for Asterisk Call Center

[A J Stiles]

On Friday 09 September 2011, bilal ghayyad wrote:
> Hi All;
> 
> Anyone advise for a free (open source) reporting to be used for asterisk
> call center?
> 
> Regards
> Bilal

Problem is, reporting is such a nebulous thing, about the only thing that will 
give you the required level of "generality of purpose" is a programming 
language.  So you might find it simplest just to roll your own.

Assuming you have set up CDR using some kind of database backend, then all you 
really need to do is devise a set of queries which will provide the 
information you want to present in your reports; then write a script in your 
favourite language to make a CSV file  (which will load into any modern 
spreadsheet program)  and e-mail it to whoever needs it.  Lastly, put an entry 
in your crontab to run the report whenever required.

Note that OpenOffice.org calc will treat an entry in a CSV starting with an = 
sign as a formula, so you can insert things such as

"${name}",${ext},${answd},${busy},${unobtain},=C${row}+D${row}+E${row}

and then column F in each row will contain the total of columns C, D and E in 
that row  (assuming you are properly updating $row as you go along .)  My 
own preference is to use the spreadsheet program to evaluate formulae rather 
than hard-code the answer into the spreadsheet.  That way, if you edit a cell, 
you will not throw everything else out.


I know from past experience that Microsoft Excel will happily accept CSV files 
with an XLS extension  (N.B. it might need \r\n for line endings);  however, I 
am unable to test whether or not it will accept formulae as above, as I do not 
have access to a machine with Windows and Excel.


-- 
AJS

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously

[virendra bhati]

Hi
1st check that how many manager is connected into the server. 1 or more then
you can say that 2 DTMF is capture by asterisk for same events.

 manager show connected
  Username IP Address
  root 127.0.0.1

it should be one only.

I face the same case then I found that more then 1 manager was working into
the server.


On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban
wrote:

> Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO
> simultaneously. That has the effect, that asterisk read every dtmf
> twice. and yes, it's mainly the carriers mistake. but is there a
> configure option, that asterisk accept only one DMTF method for
> inbound dtmf?
>
> Kristijan
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



-- 



-
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] broadcast

[virendra bhati]

Hi Sam,

I am doing the same things.
into your suggested script you join into context Konference and then .call
file start IVRs .

the same logic I have pasted in which I make .call file and then join into
the Konference and then .call file start it's work.

But As i know they are on  different -2 channels and not joined into same
conference. That's why no audio is able to broadcast into conference.

[broadcast-message]
exten => s,1,Answer()
exten => s,n,Set(p="/var/spool/
asterisk/monitor/")
exten => s,n,playback(${p}/LQA/12/Biology/Que3)
exten => s,n,playback(${p}/LQA/12/Biology/Que4)
exten => s,n,playback(${p}/LQA/12/Biology/Que5)
exten => s,n,playback(${p}/LQA/12/Biology/Que6)
exten => s,n,playback(${p}/LQA/12/Biology/Que7)
exten => s,n,Wait(10)
exten => s,n,Hangup()

 Where you have mention in which conf. it will be start ?

miss comunication in between .call and rest users.


On Tue, Sep 13, 2011 at 12:34 PM, Sam Govind  wrote:

> Virendra,
> you need to change your logic just a bit. in call file a Channel is one
> which needs to be dialled fires (See 
> link).
> this will be an extension where your Konference is Hosted for all the other
> callers to join. i.e  *Channel: local/s@Konference*
>
> [Konference]
> exten => s,1,ANSWER()
> exten => s,n,if(conference is already started//do nothing else: trigger the
> system command to make a call file...don't forget to move it to outgoing
> directory)
> exten => s,n,SET(some thing else you need to set for each incoming call i.e
> save CallerID etc)
> exten => s,n(message),Konference(43689956,ADMRSTV)
> exten => s,n,Hangup()
>
> Note that the call file should be triggered only for the first caller and
> not every time a participant joins in. That'll case overlap message
> broadcasts.
>
> Next thing in call file is the destination which will be playing broadcast
> message once Konference is called.
>
> *Context:*broadcast-message
> *Extension: *s
> *Priority: *1
> *
> *
> [broadcast-message]
> exten => s,1,Answer()
> exten => s,n,Set(p="/var/spool/asterisk/monitor/")
> exten => s,n,playback(${p}/LQA/12/Biology/Que3)
> exten => s,n,playback(${p}/LQA/12/Biology/Que4)
> exten => s,n,playback(${p}/LQA/12/Biology/Que5)
> exten => s,n,playback(${p}/LQA/12/Biology/Que6)
> exten => s,n,playback(${p}/LQA/12/Biology/Que7)
> exten => s,n,Wait(10)
> exten => s,n,Hangup()
>
> This should work and konference should listen to the playbacks.
>
> Regards,
> Sammy.
>
> On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati wrote:
>
>> Hi List,
>>
>> I make a script for .call file and then I started playback on local
>> channel but nothing was hearing at another channles.
>>
>> exten => 1234,1,Answer()
>> exten => 1234,n,System(echo -e "Channel: Channel: 
>> local/23@contest-call\\nContext:
>> contest-call\\nExtension: 23\\nPriority: 1" > /tmp/${UNIQUEID}.call)
>> exten => 1234,n,Konference(43689956,ADMRSTVL)
>>
>> [contest-call]
>>
>> exten => _X!,1,Answer()
>> exten => _X!,n,Set(p="/var/spool/asterisk/monitor/")
>> exten => _X!,n,playback(${p}/LQA/12/Biology/Que3)
>> exten => _X!,n,playback(${p}/LQA/12/Biology/Que4)
>> exten => _X!,n,playback(${p}/LQA/12/Biology/Que5)
>> exten => _X!,n,playback(${p}/LQA/12/Biology/Que6)
>> exten => _X!,n,playback(${p}/LQA/12/Biology/Que7)
>> exten => _X!,n,Konference(43689956,ADMRSTV)
>> exten => _X!,n,Wait(10)
>> exten => _X!,n,Hangup()
>>
>> in it I am dialing 1234 from softphone then join to conf in mute mode,
>> after it .call file start playback at it's own channels but I am not able to
>> hear anything into conf.
>>
>> As i know localdial is not joining into the conf. but how I will do it so
>> that I will be able to hear any played file into conference ?
>>
>>
>>
>> On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind  wrote:
>>
>>> Good to know,
>>>
>>> I think it'll be a feedback score or a poll from members of the
>>> conference. So if you use the R option and collect DTMF from members, and an
>>> AMI script listening to that particular DTMF event collects all. This way
>>> your AMI listener script should be able to tell you at the end of poll what
>>> user inserted with DTMF.
>>>
>>> So overall insertion of a broadcast message using Ahmed's method of .call
>>> file and later on collecting DTMF events from AMI script
>>> should theoretically work for you.
>>>
>>> On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati wrote:
>>>
 Hi Sam,

 You are right. I am looking for the same

 On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind  wrote:

> IMHO, I think Bhaati is trying to get feedback from
> multiple conference users. See DTMF options in Konference module.
>  'R' : enable DTMF relay: DTMF tones generate a manager event
>  If neither 'X' nor 'R' are present, DTMF tones will be forwarded to
> all members in the conference
>
> While some file is played and users press any DTMF collect the AMI
> events from each user and use them as you require.

Re: [asterisk-users] broadcast

[Gohar Ahmed]

Hey there

You are not moving the call file to spool/outgoing directory. Maybe that's
why you aren't getting anything. I don't feel good about the call file also.
Its not doing what you want it to do.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Tuesday, September 13, 2011 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] broadcast

 

Hi List,

I make a script for .call file and then I started playback on local channel
but nothing was hearing at another channles.

exten => 1234,1,Answer()
exten => 1234,n,System(echo -e "Channel: Channel:
local/23@contest-call\\nContext: contest-call\\nExtension: 23\\nPriority: 1"
> /tmp/${UNIQUEID}.call)
exten => 1234,n,Konference(43689956,ADMRSTVL)

[contest-call]

exten => _X!,1,Answer()
exten => _X!,n,Set(p="/var/spool/asterisk/monitor/")
exten => _X!,n,playback(${p}/LQA/12/Biology/Que3)
exten => _X!,n,playback(${p}/LQA/12/Biology/Que4)
exten => _X!,n,playback(${p}/LQA/12/Biology/Que5)
exten => _X!,n,playback(${p}/LQA/12/Biology/Que6)
exten => _X!,n,playback(${p}/LQA/12/Biology/Que7)
exten => _X!,n,Konference(43689956,ADMRSTV)
exten => _X!,n,Wait(10)
exten => _X!,n,Hangup()

in it I am dialing 1234 from softphone then join to conf in mute mode, after
it .call file start playback at it's own channels but I am not able to hear
anything into conf.

As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?

 

On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind  wrote:

Good to know,

 

I think it'll be a feedback score or a poll from members of the conference.
So if you use the R option and collect DTMF from members, and an AMI script
listening to that particular DTMF event collects all. This way your AMI
listener script should be able to tell you at the end of poll what user
inserted with DTMF.

 

So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script should
theoretically work for you. 

 

On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati  wrote:

Hi Sam,

You are right. I am looking for the same 

 

On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind  wrote:

IMHO, I think Bhaati is trying to get feedback from multiple conference
users. See DTMF options in Konference module. 

 'R' : enable DTMF relay: DTMF tones generate a manager event 
 If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
members in the conference

 

While some file is played and users press any DTMF collect the AMI events
from each user and use them as you require.

 

Ref: http://main.voiptoday.org/index.php?option=com_content

&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-av
ailable&catid=35:general&Itemid=173

 

 

On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati  wrote:

Hi Ahmed,

Konference is also an conferencing application.

On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed  wrote:

Hhhmmm..I dunt have any experience with module Konference. Maybe anyone else
can help you on that. 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, September 12, 2011 1:28 PM


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] broadcast

 

Hi Ahmed,

I did the same thing earlier to test the load of Digium card. But this time
I want to play file and want to get some DTMF from all the members of
conference.

So in this case I need more control into Konference module. But when I use
.call files then control will not go longer with all events.

Is there any alternate way to do it? 

I appreciate your suggestion and will doing in parallel at higher priority

On Mon, Sep 12, 2011 at 12:33 PM, Gohar Ahmed 
wrote:

Make a .call file..join one leg to local extension which plays the file and
the other leg to conference. The local extension will be like a conference
member.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, September 12, 2011 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] broadcast

 

Hi List,

Is there any way by which I can broadcast any audio file to all members into
the conference ?
I don't want to play file individual channels.

-- 




-
Thanks and regards

 Virendra Bhati
+91-9172341457  
Software Engineer

 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www

Re: [asterisk-users] broadcast

[Sam Govind]

Virendra,
you need to change your logic just a bit. in call file a Channel is one
which needs to be dialled fires (See
link).
this will be an extension where your Konference is Hosted for all the other
callers to join. i.e *Channel: local/s@Konference*

[Konference]
exten => s,1,ANSWER()
exten => s,n,if(conference is already started//do nothing else: trigger the
system command to make a call file...don't forget to move it to outgoing
directory)
exten => s,n,SET(some thing else you need to set for each incoming call i.e
save CallerID etc)
exten => s,n(message),Konference(43689956,ADMRSTV)
exten => s,n,Hangup()

Note that the call file should be triggered only for the first caller and
not every time a participant joins in. That'll case overlap message
broadcasts.

Next thing in call file is the destination which will be playing broadcast
message once Konference is called.

*Context:*broadcast-message
*Extension: *s
*Priority: *1
*
*
[broadcast-message]
exten => s,1,Answer()
exten => s,n,Set(p="/var/spool/asterisk/monitor/")
exten => s,n,playback(${p}/LQA/12/Biology/Que3)
exten => s,n,playback(${p}/LQA/12/Biology/Que4)
exten => s,n,playback(${p}/LQA/12/Biology/Que5)
exten => s,n,playback(${p}/LQA/12/Biology/Que6)
exten => s,n,playback(${p}/LQA/12/Biology/Que7)
exten => s,n,Wait(10)
exten => s,n,Hangup()

This should work and konference should listen to the playbacks.

Regards,
Sammy.

On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati  wrote:

> Hi List,
>
> I make a script for .call file and then I started playback on local channel
> but nothing was hearing at another channles.
>
> exten => 1234,1,Answer()
> exten => 1234,n,System(echo -e "Channel: Channel: 
> local/23@contest-call\\nContext:
> contest-call\\nExtension: 23\\nPriority: 1" > /tmp/${UNIQUEID}.call)
> exten => 1234,n,Konference(43689956,ADMRSTVL)
>
> [contest-call]
>
> exten => _X!,1,Answer()
> exten => _X!,n,Set(p="/var/spool/asterisk/monitor/")
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que3)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que4)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que5)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que6)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que7)
> exten => _X!,n,Konference(43689956,ADMRSTV)
> exten => _X!,n,Wait(10)
> exten => _X!,n,Hangup()
>
> in it I am dialing 1234 from softphone then join to conf in mute mode,
> after it .call file start playback at it's own channels but I am not able to
> hear anything into conf.
>
> As i know localdial is not joining into the conf. but how I will do it so
> that I will be able to hear any played file into conference ?
>
>
>
> On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind  wrote:
>
>> Good to know,
>>
>> I think it'll be a feedback score or a poll from members of the
>> conference. So if you use the R option and collect DTMF from members, and an
>> AMI script listening to that particular DTMF event collects all. This way
>> your AMI listener script should be able to tell you at the end of poll what
>> user inserted with DTMF.
>>
>> So overall insertion of a broadcast message using Ahmed's method of .call
>> file and later on collecting DTMF events from AMI script
>> should theoretically work for you.
>>
>> On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati wrote:
>>
>>> Hi Sam,
>>>
>>> You are right. I am looking for the same
>>>
>>> On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind  wrote:
>>>
 IMHO, I think Bhaati is trying to get feedback from
 multiple conference users. See DTMF options in Konference module.
  'R' : enable DTMF relay: DTMF tones generate a manager event
  If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
 members in the conference

 While some file is played and users press any DTMF collect the AMI
 events from each user and use them as you require.

 Ref:
 http://main.voiptoday.org/index.php?option=com_content&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-available&catid=35:general&Itemid=173


 On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati wrote:

> Hi Ahmed,
>
> Konference is also an conferencing application.
>
> On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed 
> wrote:
>
>> Hhhmmm..I dunt have any experience with module Konference. Maybe
>> anyone else can help you on that. 
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra
>> bhati
>> *Sent:* Monday, September 12, 2011 1:28 PM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] broadcast
>>
>>  ** **
>>
>> Hi Ahmed,
>>
>> I did the same thing earlier to test the load of Digium card. But this
>> time I want to play file and want to get some DTMF from all the members 
>>