[asterisk-users] file command with alaw file

[Pham Quy]

hi all,

I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 #1 SMP Tue
Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) and do record
audio clip with mixmonitor() as alaw file (softphone is also configured
with alaw active only). Using file command i can get the following
information

983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio,
ITU G.711 A-law, mono 8000 Hz

But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5
#1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) i
could get the same information with file command. File command
recognized alaw file as JPEG image:

983006584-20100517-123825.alaw: JPEG image data

I guess i may miss something when i setup the new on on Centos 5.5, but
u dont know what it is. Anyone have idea about this?

please help.

Thanks in advance.
Quyps


-- 
_
-- 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

[asterisk-users] Microsoft Response Point Voip server discontinued

[Dean Collins]

Interesting announcement today;

 

 

http://www.crn.com.au/News/174872,microsoft-response-point-voip-gets-a-d
irt-nap.aspx

Microsoft this week revealed its intention to discontinue its Response
Point, its small business VoIP system for companies with up to 50
employees. 

While not unexpected, the move is significant because Response Point was
once a promising product in which Microsoft Chairman Bill Gates took a
particular interest. 

"After taking a good look at the Microsoft Response Point offering and
the needs of small businesses, we've decided to discontinue the sale,
support, and development of the Response Point phone system for small
businesses, effective August 31, 2010

 

 

 

Anyone got any thoughts about using OCS in smaller than 50 users?

 

Shows the naysayers who suggested 3com/MS voip was better because they
had "big company support" compared to the small asterisk vendors.

 

 

 

 

 

 

 

Regards, 
Dean Collins 
Cognation Inc 
d...@cognation.net 
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
 
Twitter - http://twitter.com/deancollins
Facebook - http://www.facebook.com/deancollins
Linked In - http://www.linkedin.com/in/deancollins 

 

<>-- 
_
-- 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] 1.6.2.7 SIP realtime problem

[Vardan]

If you become this message at start up of asterisk, then try it preload 
in module.conf

I think it will be help.
I have also became this problem on 1.6.1 version

Vardan


Bruce Ferrell wrote:
> On 05/14/2010 04:17 PM, Tilghman Lesher wrote:
>> On Friday 14 May 2010 16:09:58 Bruce Ferrell wrote:
>>
>>> I'm getting the following message in my full log at startup and my
>>> realtime sip peers aren't being found. My realtime extensions have no
>>> errors.  The table sippeers exists in the database.   Is this a known
>>> problem?
>>>
>>> res_config_mysql.c: Table sippeers not found in database.  This table
>>> should exist if you're using realtime.
>>>
>> Check your [context] in res_mysql.conf.  In previous versions, it was set as
>> [general], but extconfig.conf had "asterisk" as the name of the connection.
>> These two configuration files need to match.  It's correct in the sample
>> configs, but if you upgraded from a prior version, it's possible that you
>> still have the bad match.
>>
>>
> That did it.
>
> in res_mysql it's
>
> [general]
> dbhost = 127.0.0.1
> dbname = asterisk
> dbuser = astuser
> dbpass = astpass
> dbport = 3306
> dbsock = /var/run/mysql/mysql.sock
> requirements=warn
>
>
>
> in extconfig the line
>
>sippeers =>  mysql,asterisk,sip_buddies
>
> asterisk pointed to the dbname in res_mysql, not the context.
>
> It still works that way in 1.4.31
>
> That was fun... NOT! :)
>


-- 
_
-- 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] ISDN config: LBO values

[Tilghman Lesher]

On Sunday 16 May 2010 17:32:09 Jaap Winius wrote:
> Quoting Tilghman Lesher :
> >> The value selected should almost always be zero. However, if the cable
> >> is of a significant length, another value must be selected, but which
> >> one? There are two columns: CSU and DSX-1. When is it appropriate to
> >> use the one or the other to determine the correct LBO value?
> >
> > Each LBO value is a different amount of loss to be expected on the
> > line, and therefore the signal is amplified a commensurate amount.
> > What it really comes down to is what works for you.
>
> That's the usual approach, but if I was still happy with it I would
> not have asked the question. According to the manual, the values are
> found in a table, but what good is that if you can't make any sense of
> it?

http://www.digchip.com/datasheets/parts/datasheet/222/82V2088-pdf.php

See pages 17-18 of the associated PDF.  While this is not the T1 framer chip
used, the values are identical, which leads me to believe that these values
are actually industry standard.  The values used are merely inputs into the
firmware and the T1 framer does the rest.  BTW, you can find the datasheet
for the actual T1 framer chip, but it's less helpful than the one above.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

-- 
_
-- 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] OK, I'm stumped

[Jose Flores Galicia]

Yep, that's the way analog lines on asterisk works.

Callprogress had never work for production to me, the way I resolv this is
asking to Telco the polarity reverse on answer and it will fully work for an
production environment.

Switching to PRI is even better, but think will be an small scenario.

Jose Flores Galicia
<>
BriefCode && Code Based Training


2010/5/16 Pascal Bruno 

> That's how analog lines work. Asterisk do not know when the called is
> picked up so it goes straight to the context execution.  You may want to try
> setting callprogress=yes and answeronpolarity=yes on your chan dahdi conf
> file as a work around, or switch to PRI
>
>
>
> On Sun, May 16, 2010 at 3:38 PM, Adolphe Cher-aime wrote:
>
>> Mi too I've experienced the same problem with my script. Dahdi answers the
>> channel once Ami is running it's the same thing for call files . When using
>> sip Chanel or skype channel it work as I wanted. I thank that analog fxo is
>> the problem if automatic outgoing calls when you want the called party to
>> answer first befor moving to the context extension.
>>
>> Adolphe Cher-aime
>> From my Iphone
>>
>> On May 16, 2010, at 1:32 PM, Jose Flores Galicia 
>> wrote:
>>
>> Maybe because I am closer to several customers which often make questions
>> like yours.
>>
>> I can supposse you mean that the call is answered by the dahdi channel as
>> soon as you set the originate command on AMI, I supposse you are using an
>> FXO channel connected to your POTS line.
>>
>> Am I right?
>>
>> Jose Flores Galicia
>> << floj...@gmail.com>>
>> BriefCode && Code Based Training
>>
>>
>> 2010/5/16 Bruce Ferrell < bferr...@baywinds.org>
>>
>>> I'm trying to make an AMI call.  I want to call a number, play an
>>> announcement when the call is answered, then call a second number and
>>> connect the two when the second call is answered.
>>>
>>> I an able to make a simple call to two numbers and connect them using
>>> the manager API but playing the announcement has me beat.
>>>
>>> Suggestions anyone?
>>>
>>> Bruce Ferrell
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>> --
>> _
>> -- 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
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Pascal B.
> http://www.kameleonlabs.com/
> Twitter: @petchaw
>
> --
> _
> -- 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
>
-- 
_
-- 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] ISDN config: LBO values

[Jaap Winius]

Quoting Tilghman Lesher :

>> The value selected should almost always be zero. However, if the cable
>> is of a significant length, another value must be selected, but which
>> one? There are two columns: CSU and DSX-1. When is it appropriate to
>> use the one or the other to determine the correct LBO value?
>
> Each LBO value is a different amount of loss to be expected on the
> line, and therefore the signal is amplified a commensurate amount.
> What it really comes down to is what works for you.

That's the usual approach, but if I was still happy with it I would  
not have asked the question. According to the manual, the values are  
found in a table, but what good is that if you can't make any sense of  
it?

In the mean time, I've googled some more and found one text that  
suggests CSU and DSX-1 are both T1 trunk interface types, while  
another suggests that a DSX-1 is an interface that a CSU is attached to.

It seems to me that the table refers to two situations that used to  
(or maybe still do) occur in North America in which an ISDN card is be  
attached to a T1 trunk line via a CSU/DSU (the "DSX-1"), or only a  
CSU. In the latter case, the ISDN card must also act as a DSU.

Can anyone say is this is correct? Any further explanation would be welcome.

Cheers,

Jaap

-- 
_
-- 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] Digium TE121P + DAHDI

[hin lee]

I too, had really bad echoes on a TE121 w/ echo module.  When I removed the 
module, I haven't had as much echoes as before.





From: Sascha Ferley 
To: asterisk-users@lists.digium.com
Sent: Sun, May 16, 2010 12:00:31 PM
Subject: [asterisk-users] Digium TE121P + DAHDI

Hi, 

Just trying to figure out / solve some echo issues with one of our systems.
The server is a Dell R710 running Asterisk 1.6.x and Dahdi 2.2.x
The card in the server is a TE121P (with VPMADT032 echo canceller)

We are experiencing some weird echo issues with our Cisco 79xx phones, only
when we are dialing out via the PRI. The echo is like a side tone, where we
can hear ourselves speaking. The other party doesn't hear this however and
its perfectly clean.

After solving some of the IRQ/processing issues in which we seemed to get
HDLC errors pop up after a week and take down the connection completely
(only solvable with a reboot); we changed the smp_affinity settings for all
devices to basically run on their own core.
This seemed to have solved us getting extended IRQ misses and increase in
delays. However it hadn't fixed the echo issue.

After some digging on this it seems that there could be a problem with DAHDI
and I want to get to the bottom of it.
Reference: https://issues.asterisk.org/view.php?id=15724

In my dmesg logs I see that dahdi loads the MG2 echo canceller, is this
correct? I thought that the dahdi firmware for VPMADT032  would take care of
that. Thus I want to insure that we are actually using the hardware module
for echo cancellation and not some software canceller, which is acting
sub-par. 

If this is correct, is there any solution to this problem sofar?

Thanks

S.



-- 
_
-- 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



  -- 
_
-- 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] OK, I'm stumped

[Pascal Bruno]

That's how analog lines work. Asterisk do not know when the called is picked
up so it goes straight to the context execution.  You may want to try
setting callprogress=yes and answeronpolarity=yes on your chan dahdi conf
file as a work around, or switch to PRI



On Sun, May 16, 2010 at 3:38 PM, Adolphe Cher-aime wrote:

> Mi too I've experienced the same problem with my script. Dahdi answers the
> channel once Ami is running it's the same thing for call files . When using
> sip Chanel or skype channel it work as I wanted. I thank that analog fxo is
> the problem if automatic outgoing calls when you want the called party to
> answer first befor moving to the context extension.
>
> Adolphe Cher-aime
> From my Iphone
>
> On May 16, 2010, at 1:32 PM, Jose Flores Galicia 
> wrote:
>
> Maybe because I am closer to several customers which often make questions
> like yours.
>
> I can supposse you mean that the call is answered by the dahdi channel as
> soon as you set the originate command on AMI, I supposse you are using an
> FXO channel connected to your POTS line.
>
> Am I right?
>
> Jose Flores Galicia
> << floj...@gmail.com>>
> BriefCode && Code Based Training
>
>
> 2010/5/16 Bruce Ferrell < bferr...@baywinds.org>
>
>> I'm trying to make an AMI call.  I want to call a number, play an
>> announcement when the call is answered, then call a second number and
>> connect the two when the second call is answered.
>>
>> I an able to make a simple call to two numbers and connect them using
>> the manager API but playing the announcement has me beat.
>>
>> Suggestions anyone?
>>
>> Bruce Ferrell
>>
>> --
>> _
>> -- 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
>>
>
> --
> _
> -- 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
>
>
> --
> _
> -- 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
>



-- 
Pascal B.
http://www.kameleonlabs.com/
Twitter: @petchaw
-- 
_
-- 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] OK, I'm stumped

[Adolphe Cher-aime]

Mi too I've experienced the same problem with my script. Dahdi answers  
the channel once Ami is running it's the same thing for call files .  
When using sip Chanel or skype channel it work as I wanted. I thank  
that analog fxo is the problem if automatic outgoing calls when you  
want the called party to answer first befor moving to the context  
extension.


Adolphe Cher-aime
From my Iphone

On May 16, 2010, at 1:32 PM, Jose Flores Galicia   
wrote:


Maybe because I am closer to several customers which often make  
questions like yours.


I can supposse you mean that the call is answered by the dahdi  
channel as soon as you set the originate command on AMI, I supposse  
you are using an FXO channel connected to your POTS line.


Am I right?

Jose Flores Galicia
<>
BriefCode && Code Based Training


2010/5/16 Bruce Ferrell 
I'm trying to make an AMI call.  I want to call a number, play an
announcement when the call is answered, then call a second number and
connect the two when the second call is answered.

I an able to make a simple call to two numbers and connect them using
the manager API but playing the announcement has me beat.

Suggestions anyone?

Bruce Ferrell

--
_
-- 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

--
_
-- 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
-- 
_
-- 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

[asterisk-users] Digium TE121P + DAHDI

[Sascha Ferley]

Hi, 

Just trying to figure out / solve some echo issues with one of our systems.
The server is a Dell R710 running Asterisk 1.6.x and Dahdi 2.2.x
The card in the server is a TE121P (with VPMADT032 echo canceller)

We are experiencing some weird echo issues with our Cisco 79xx phones, only
when we are dialing out via the PRI. The echo is like a side tone, where we
can hear ourselves speaking. The other party doesn't hear this however and
its perfectly clean.

After solving some of the IRQ/processing issues in which we seemed to get
HDLC errors pop up after a week and take down the connection completely
(only solvable with a reboot); we changed the smp_affinity settings for all
devices to basically run on their own core.
This seemed to have solved us getting extended IRQ misses and increase in
delays. However it hadn't fixed the echo issue.

After some digging on this it seems that there could be a problem with DAHDI
and I want to get to the bottom of it.
Reference: https://issues.asterisk.org/view.php?id=15724

In my dmesg logs I see that dahdi loads the MG2 echo canceller, is this
correct? I thought that the dahdi firmware for VPMADT032  would take care of
that. Thus I want to insure that we are actually using the hardware module
for echo cancellation and not some software canceller, which is acting
sub-par. 

If this is correct, is there any solution to this problem sofar?

Thanks

S.



-- 
_
-- 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] OK, I'm stumped

[Jose Flores Galicia]

Maybe because I am closer to several customers which often make questions
like yours.

I can supposse you mean that the call is answered by the dahdi channel as
soon as you set the originate command on AMI, I supposse you are using an
FXO channel connected to your POTS line.

Am I right?

Jose Flores Galicia
<>
BriefCode && Code Based Training


2010/5/16 Bruce Ferrell 

> I'm trying to make an AMI call.  I want to call a number, play an
> announcement when the call is answered, then call a second number and
> connect the two when the second call is answered.
>
> I an able to make a simple call to two numbers and connect them using
> the manager API but playing the announcement has me beat.
>
> Suggestions anyone?
>
> Bruce Ferrell
>
> --
> _
> -- 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
>
-- 
_
-- 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] Aastra SIP phone regisration problems

[Jose Flores Galicia]

Hi Mike.

I can tell AASTRA phones include an registration timeout that is usually set
to 0, or not set, but ehen not set or 0, it means that it will no
re-register every amount of time, so, you should look at that value; also
review that value is set accordly to minexpiry (default 60s) and maxexpiry
(default 3600 sec) values on sip.conf, I noticed on re-registrations AASTRA
firmware is a little bit buggy.

Best Regards.
Jose Flores Galicia
<>
BriefCode && Code Based Training


2010/5/16 mike mosier 

> I have 8 aastra phones that are loosing registration. On the phone gui it
> says 408 as the registation error after a minute or say they register. In
> the cli it eill say the phone is now unreachable then it will show it
> registering then available. At first they did it every hour all the phones.
> After messing with the experation it does it every 15 nin.
>
> Any ideas on how to troubleshoot this? I tried a debug on the phone in the
> cli but I ended up with over 100 pages of text. Via sip show debug peer 2000
> or something like that.
>
> Respectfully
> Michael D Mosier
> Ftoc Certified
>
> --
> _
> -- 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
>
-- 
_
-- 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] OK, I'm stumped

[Jeff Brower]

Bruce-

> I'm trying to make an AMI call.  I want to call a number, play an
> announcement when the call is answered, then call a second number and
> connect the two when the second call is answered.
>
> I an able to make a simple call to two numbers and connect them using
> the manager API but playing the announcement has me beat.
>
> Suggestions anyone?

Suggest:

  -use an accurate subject line

  -be very specific about the problem

  -show an example of your script or command sequence
   that is not working

When people search Asterisk archives for a solution they're typically going to 
search for subject lines containing
technical terms related to their problem.

-Jeff



-- 
_
-- 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] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality

[Vieri]

--- On Fri, 5/14/10, Steve Edwards  wrote:

> > I'm supposing my system is using the DAHDI-driven
> Digium cards on my 
> > motherboard. I don't know how hardware timers work and
> if Digium 
> > hardware rely on the motherboard (my system clock is
> going too fast and 
> > my ntpd is constantly adjusting the clock by -2.6
> seconds every 20 
> > minutes). In any case, since I'm on a dedicated LAN I
> guess I can safely 
> > set trunk=no.
> 
> Maybe it's just me, but I'd be thinking if the mobo
> manufacturer did such 
> a crappy job on the clock, what else is wrong. I'd be
> looking for a better 
> mobo.

The manufacturer is ASUS.
The mobo is M4A77TD PRO, latest BIOS update.

Supposedly manufacturers such as HP and Dell should be better but people 
"usually" have a good opinion on Asus.

Beats me.




  

-- 
_
-- 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] OK, I'm stumped

[Steve Edwards]

On Sun, 16 May 2010, Bruce Ferrell wrote:

> Suggestions anyone?

We all get "stumped" from time to time.

A better subject yields better responses.

-- 
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] OK, I'm stumped

[Tzafrir Cohen]

On Sun, May 16, 2010 at 07:15:50AM -0700, Bruce Ferrell wrote:
> I'm trying to make an AMI call.  I want to call a number, play an
> announcement when the call is answered, then call a second number and
> connect the two when the second call is answered.
> 
> I an able to make a simple call to two numbers and connect them using
> the manager API but playing the announcement has me beat.

Isn't it simple? What am I missing?

Action: Originate
Channel: $channel_for_first_number
Context: whatever
Extension: $second_number

...

[whatever]
exten=> _X.,1,Playback(an-announcement)
exten=> _X.,n,Dial(${some_trunk}/${EXTEN})

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- 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] OK, I'm stumped

[Jim Dickenson]

Use a dialplan to do what you want and dial that.

Originate a call to the first person and point it at context, exten, priority 
that plays the sound file and then does a dial command to the number you want 
them to talk to.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On May 16, 2010, at 7:15 AM, Bruce Ferrell wrote:

> I'm trying to make an AMI call.  I want to call a number, play an
> announcement when the call is answered, then call a second number and
> connect the two when the second call is answered.
> 
> I an able to make a simple call to two numbers and connect them using
> the manager API but playing the announcement has me beat.
> 
> Suggestions anyone?
> 
> Bruce Ferrell
> 
> -- 
> _
> -- 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


-- 
_
-- 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] Problems with Asterisk and two Linksys SPA941

[Dan Journo]

Hi,

I have the same setup as you.

I didn't bother mapping any ports.
I just enabled nat and keepalive.

Here is a screenshot of the config on the phones. For some reason, on some 
phones I had to turn the NAT Mapping Enabled to Off otherwise call transfer 
didn't work.

http://www.postimage.org/image.php?v=Tsz5WIJ

The sip.conf setup for this phone is:-

[winsor_202]
type=friend
context=winsor_phones
host=dynamic
secret=passwordhere
nat=yes
disallow=all
allow=gsm
allow=ulaw
canreinvite=yes
vmexte...@winsor
mailbo...@winsor
pickupgroup=1
callgroup=1

Hope that helps.
Dan

-- 
_
-- 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] OK, I'm stumped

[bruce bruce]

Maybe drop the call in a Meetme room and have an announcement?

On Sun, May 16, 2010 at 10:15 AM, Bruce Ferrell wrote:

> I'm trying to make an AMI call.  I want to call a number, play an
> announcement when the call is answered, then call a second number and
> connect the two when the second call is answered.
>
> I an able to make a simple call to two numbers and connect them using
> the manager API but playing the announcement has me beat.
>
> Suggestions anyone?
>
> Bruce Ferrell
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] OK, I'm stumped

[Bruce Ferrell]

I'm trying to make an AMI call.  I want to call a number, play an
announcement when the call is answered, then call a second number and
connect the two when the second call is answered.

I an able to make a simple call to two numbers and connect them using
the manager API but playing the announcement has me beat.

Suggestions anyone?

Bruce Ferrell

-- 
_
-- 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] Re-compiling q931.c

[Tzafrir Cohen]

On Sun, May 16, 2010 at 09:18:13AM -0400, bruce bruce wrote:
> Thanks for the input.
> 
> I need a bit more clarification.
> 
> I just changed the condition for an "if" in the q931.c from Equal to Not
> Equal. In order to get that reflected do I have to do "gcc q931.c" to
> recompile it so Asterisk can read it? or does asterisk read the .c file?

Asterisk reads the compiled code generated from that.

Normally projects use build instructions in the form of a 'Makefile' or
something similar. Which means you only need to run 'make' in the
directory of the code

(Actually most projects have some sort of configuration script, and you
need to run './configure' once. But this is not the case with libpri)

So, 

1. Edit q931.c
2. Run 'make'

Then use the trick with LD_LIBRARY_PATH (again, it may not work for
every library out there, but libpri has a very simple structure and thus
it works)

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- 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] play a sound file directly to a caller channel

[Jim Dickenson]

We do the following:

Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280_18-0002
ActionID: PlayBack
Async: true


exten => do_playback,1,Answer()
exten => do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID} & 
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_playback,n,Wait(0.3)
exten => do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten => do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID} & 
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear} & 
${PLAYBACKSTATUS})
exten => do_playback,n,Hangup()


exten => do_chanspy,1,Answer()
exten => do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID} & 
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten => do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID} & 
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_chanspy,n,Hangup()


-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On May 16, 2010, at 4:16 AM, Daniel Knoll wrote:

> Hello User-List,
> is it possible to play a sound file directly to a caller channel? 
> 
> Like this AMI command
> 
> Action: Originate
> Channel: SIP/20-1d41
> Application: Playback
> Data: /path/to/audio/file
> 
> I get an Error Message. My intension is to play a sound file to a caller and 
> the other callers don't hear this.
> Can someone help me ?
> 
> Thanks a lot 
> Bye Daniel
> 
> 
> 
> 
> -- 
> _
> -- 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


-- 
_
-- 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] Re-compiling q931.c

[bruce bruce]

Thanks for the input.

I need a bit more clarification.

I just changed the condition for an "if" in the q931.c from Equal to Not
Equal. In order to get that reflected do I have to do "gcc q931.c" to
recompile it so Asterisk can read it? or does asterisk read the .c file?

Thanks again,
Bruce

On Sat, May 15, 2010 at 4:56 PM, Tzafrir Cohen wrote:

> On Sat, May 15, 2010 at 04:32:19PM -0400, bruce bruce wrote:
> > Hi Guys,
> >
> > Can q931.c be re-compiled using gcc or something else without the need to
> > re-do the whole libpri? Some changes were made to q931.c and I want those
> to
> > be reflected in .a .o .so .lo files as I think those are the files read
> by
> > Asterisk vs the .c file.
>
> Given that libpri is such a small library, I wouldn't bother. If the
> change did not break the binary ABI (e.g.: changed the size of a struct
> that is exposed through some interface, added/removed variables to some
> function) there should be no issue here.
>
> If you want to test things, try just building libpri (and not installing
> it) and start asterisk with:
>
>  LD_LIBRARY_PATH=/path/to/libpri/source asterisk
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] play a sound file directly to a caller channel

[Daniel Knoll]

Hello User-List,
is it possible to play a sound file directly to a caller channel? 

Like this AMI command

Action: Originate
Channel: SIP/20-1d41
Application: Playback
Data: /path/to/audio/file

I get an Error Message. My intension is to play a sound file to a caller and 
the other callers don't hear this.
Can someone help me ?

Thanks a lot 
Bye Daniel




-- 
_
-- 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] Problems with Asterisk and two Linksys SPA941

[mike mosier]

Do you have UDP 1 to 2 port forward to your router?

What kind of router is it?

Respectfully
Michael D Mosier
Ftoc Certified

On May 16, 2010 12:27 AM, "Olivier CALVANO"  wrote:

Hi

I have a big problems on my Asterisk systems :

I have one Asterisk Server with static IP (no nat) and
6 Linksys SPA941.

All SPA are after a router with NAT:

* SPA-1 and SPA-2 are on the same network,
   we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router

* SPA-3,
   we have a pat 5062 => SPA-3

* SPA-4,
   we have a pat 5063 => SPA-4

* SPA-5,
   we have a pat 5064 => SPA-5

* SPA-6,
   we have a pat 5065 => SPA-6

On the Asterisk Sip conf, we have nat=yes and dynamic host.


The problems are SPA-1 and SPA-2 can call to all other SPA except SPA-3
with SPA-3, i speak, it's good, spa-3 have the sound, but spa-3 speak
i don't have the sound.

SPA-3 can speak with SPA-4,5 and 6 without problems

a idea of the problems ?

Thanks
Bye

--
_
-- 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
-- 
_
-- 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

[asterisk-users] Aastra SIP phone regisration problems

[mike mosier]

I have 8 aastra phones that are loosing registration. On the phone gui it
says 408 as the registation error after a minute or say they register. In
the cli it eill say the phone is now unreachable then it will show it
registering then available. At first they did it every hour all the phones.
After messing with the experation it does it every 15 nin.

Any ideas on how to troubleshoot this? I tried a debug on the phone in the
cli but I ended up with over 100 pages of text. Via sip show debug peer 2000
or something like that.

Respectfully
Michael D Mosier
Ftoc Certified
-- 
_
-- 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