[asterisk-users] obd call drops after few seconds : only for mobile numbers

[Dharmesh Garg]

Hi,

my obd calls to all Idea mobile numbers drops after few sec.
where as, with  same configuration , and making obd on to pri it works 
properly.
Also i had words with Idea team, they says release is from my point code, 
whereas on debugging at my asterisk server i received release from their 
point code.


using :- dahdi-tools-2.3 , libss7-1.0 , asterisk-1.6.2.

Any suggestions ...?


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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Andrew Thomas]

Wow! How self-promoting was that post?

As for a simple 'that worked' post - as others have already pointed out
before you, it's not for self-gratification - it's to help anyone else
who has the same/similar problem.  I used the list archives quite a lot
in my early days - and having the last post in a thread say 'try this,
this or this' and no comeback is a pain.  A simple 'option 2 worked for
me' post at then end would make everything a lot simpler (and beat those
deadlines you talked about).

As for 'off-list' mailing - please do NOT do it without
asking/permission as most people get enough e-mails as it is (from
paying customers).

Thanks all and have a nice day!


 
 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: 11 May 2011 05:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] When someone helps you, at least let them
know if the problem is resolved or not




On Tue, May 10, 2011 at 8:30 PM, Sherwood McGowan
 wrote:



+1 from me too.  The other thing is that when you answer
to say the problem has been solved this goes into the archives meaning
that people can use Google to answer their own questions rather than
having to even ask the list.

There have been times when I've searched for a solution
to a problem, found like 10 answers, and nobody has said whether they
work or not so you have to try all of them.

-- 
Cheers,

Matt Riddell



Believe me mate, I feel you, on that note. Not only because of
my time when I was asking more questions than I was answering, but also
from the standpoint of wishing the answers were a little more prevalent
for the searching party to find so that I didn't see s many repeats
on the list ;-) 

Cheers guys! 

-- 
Sherwood McGowan
Telecommunications and VOIP Consultant




-1

Since I was the number 1 poster on this list a couple of years ago, I
think I can speak with some authority.

I just assume that if that person does not ask any more questions, that
they have either solved the problem on their own, or I helped them by
giving the answer or steering them to it.  

I don't need a public or private "Thank You"  When I was posting all the
time, I figured the ratio of "Thank you" emails to silence to be about
20 to 1, maybe as high as 50 to 1.

People are busy, under a deadline or whatever,  I offer help and do not
expect anything in return, not even a thank you.  Probably because I
have and will be one of those people, although my questions are usually
a little over the top for the list or can be pointed to something in
bugtracker, I have asked many questions when I was stuck and under an
all nighter deadline.

I would like to thank anyone out there that has helped me over the many,
many years dealing with Asterisk and VoIP.  It is a blanket thank you
for all times I simply moved onto then next hurdle to get my
deliverables out on time and working properly and neglected to post a
thank you.

Before there was any documentation, voip-info  amd this list was my
savior.  The volume of traffic has fallen to almost nothing over the
last year or two.

I wonder if Digium could post totals as it did when I was shocked to
find my name as the #1 poster.  It would be cool to see who is the #1
poster now, but I am more interested in what I perceive to be a huge
fall off of posting.

It could be my email server, since I was getting notices from the list
about excessive email bounces and removing me if I did not click a link.
That seems to have stopped, and I don't think it was on my side.

Back to getting credit or a thank you.

What I have received by answering questions or helping to troubleshoot
is worth way more than a thank you.  I get some name recognition, paid
work, large call centers, Sr Positions in high profile jobs.  Enough to
make a nice living, whether I am independent or in a salaried position.
Asterisk has literally taken me all over the world.  My last trip was to
Iraq, but I have been to Senegal, Sierra Leone, Guinea, Ghana, Liberia
to help rebuild the infrastructure for USAID. 

I don't really do job searches,  I am usually offered a job or project
and approached by the client.

For the Dept of State, I set up prepaid call centers to answer questions
and getting a reservation at the various Embassies about obtaining a
visa to come to the US.  It is called the US visa Information Service 

For DoD/Dos, I cannot really say much except I can say is that I am
probably one of the few Asterisk people that were issued a Glock and M4,
bullet proof vests, armored cars, and a PSD team..  How many VoIP guys
were taking ak47 rounds while I was on top of the Iraqi Government
building, setting up the Motorola Canop

Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[randulo]

On Wed, May 11, 2011 at 6:47 AM, Steve Totaro
 wrote:
> I don't need a public or private "Thank You"  When I was posting all the
> time, I figured the ratio of "Thank you" emails to silence to be about 20 to
> 1, maybe as high as 50 to 1.

I agree with the others who are saying that at least a results post
(It worked!) for the benefit of people trying to accomplish or fix
something or even an added RESOLVED in the subject wouldn't kill
anyone, busy or not. Ending it with a "thx" is optional, but like
chicken soup for a cold, "It wouldn't hoit".

btw Steve, for the last five years or so, every post you ever write
ends with "Thanks", so you got it covered.

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Re: [asterisk-users] how to play music when dial fail or time out

[John Wu]

Thanks Matt
the problem is solved.

On Wed, May 11, 2011 at 11:24 AM, Matt Riddell wrote:

> On 11/05/11 3:11 PM, John Wu wrote:
>
>> Hi Enrico
>> thanks I do what u said but meet this problem:
>> [May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File
>> fail.wav does not exist in any format
>> [May 11 11:08:49] WARNING[4545]: file.c:953 ast_streamfile: Unable to
>> open fail.wav (format 0x2 (gsm)): No such file or directory
>> [May 11 11:08:49] WARNING[4545]: app_playback.c:471 playback_exec:
>> ast_streamfile failed on SIP/IMSI460020656177633- for fail.wav
>>
>
> When you playback a file in Asterisk you don't provide the extension.
>
> So you'd do Playback(fail) rather than Playback(fail.wav)
>
> That way if you have 10 files all called fail (i.e. fail.wav, fail.gsm,
> fail.ulaw, fail.alaw etc etc) it will pick the one which matches the phone
> calls.
>
> For example in the above example you were making a call in the GSM format
> but it couldn't find a file called fail.wav.gsm so it looked for fail.wav.*
> and couldn't find anything.
>
> Basically just drop the extension.
>
> --
> Cheers,
>
> Matt Riddell
> ___
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/exchange.php (Full ITSP Solution)
> http://www.venturevoip.com/cc.php (Call Centre Solutions)
>
>
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Re: [asterisk-users] OUTBOUND CALLER ID

[A J Stiles]

On Wednesday 11 May 2011, mahesh katta wrote:
> Sir,
> I set the below configured in Zapata.conf file. and A .J given Dialplan .
> that's it is working now
>
> hidecallerid=no
> restrictcid=yes

Glad you got it all sorted -- I was going to suggest a few more things you 
could try this morning, but got beaten to it.

Just one thing, though:  You might want to take the NoOp() statements out of 
your dialplan, now you've got it working.  It makes little sense to clutter 
up your console and log files with unnecessary diagnostic messages.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Steve Totaro]

On Wed, May 11, 2011 at 3:55 AM, randulo  wrote:

> On Wed, May 11, 2011 at 6:47 AM, Steve Totaro
>  wrote:
> > I don't need a public or private "Thank You"  When I was posting all the
> > time, I figured the ratio of "Thank you" emails to silence to be about 20
> to
> > 1, maybe as high as 50 to 1.
>
> I agree with the others who are saying that at least a results post
> (It worked!) for the benefit of people trying to accomplish or fix
> something or even an added RESOLVED in the subject wouldn't kill
> anyone, busy or not. Ending it with a "thx" is optional, but like
> chicken soup for a cold, "It wouldn't hoit".
>
> btw Steve, for the last five years or so, every post you ever write
> ends with "Thanks", so you got it covered.
>
>
Thanks Randulo,

I am surprised you noticed that.

I truly give thanks to all productive members of the Asterisk community.

Would you say that I am a productive member of the list and go pretty far
out of my way to help people?  Most of the time give useful info, like the
Outbound Caller ID thread?

I may off topic sometimes but that is because I don't think linearly.  Tools
such as Freemind work very well for me for many applications.

Thanks,
Steve Totaro
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[randulo]

On Wed, May 11, 2011 at 12:48 PM, Steve Totaro
 wrote:
> Thanks Randulo,
>
> I am surprised you noticed that.
> I truly give thanks to all productive members of the Asterisk community.

Second that!

> Would you say that I am a productive member of the list and go pretty far
> out of my way to help people?  Most of the time give useful info, like the
> Outbound Caller ID thread?

I believe so, yes. Especially since you were careful to include the
"most of the time" disclaimer.

> I may off topic sometimes

I'm sorry, I think I can claim the award for OT posts over the past 8
years or so. That includes one that allowed me to find a perl genius
(Hi Dave VG) who did some great but totally unrelated work for me. I
knew this community would have people like that, so I threw caution to
the winds and did the OT post. Got a lot of great answers, too.

Thanks!

:r

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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Steve Totaro]

>
>
>
> On Tue, May 10, 2011 at 8:30 PM, Sherwood McGowan
>  wrote:
>
>
>
>+1 from me too.  The other thing is that when you answer
> to say the problem has been solved this goes into the archives meaning
> that people can use Google to answer their own questions rather than
> having to even ask the list.
>
>There have been times when I've searched for a solution
> to a problem, found like 10 answers, and nobody has said whether they
> work or not so you have to try all of them.
>
>--
>Cheers,
>
>Matt Riddell
>
>
>
>Believe me mate, I feel you, on that note. Not only because of
> my time when I was asking more questions than I was answering, but also
> from the standpoint of wishing the answers were a little more prevalent
> for the searching party to find so that I didn't see s many repeats
> on the list ;-)
>
>Cheers guys!
>
>--
>Sherwood McGowan
>Telecommunications and VOIP Consultant
>
>
>
>
> -1
>
> Since I was the number 1 poster on this list a couple of years ago, I
> think I can speak with some authority.
>
> I just assume that if that person does not ask any more questions, that
> they have either solved the problem on their own, or I helped them by
> giving the answer or steering them to it.
>
> I don't need a public or private "Thank You"  When I was posting all the
> time, I figured the ratio of "Thank you" emails to silence to be about
> 20 to 1, maybe as high as 50 to 1.
>
> People are busy, under a deadline or whatever,  I offer help and do not
> expect anything in return, not even a thank you.  Probably because I
> have and will be one of those people, although my questions are usually
> a little over the top for the list or can be pointed to something in
> bugtracker, I have asked many questions when I was stuck and under an
> all nighter deadline.
>
> I would like to thank anyone out there that has helped me over the many,
> many years dealing with Asterisk and VoIP.  It is a blanket thank you
> for all times I simply moved onto then next hurdle to get my
> deliverables out on time and working properly and neglected to post a
> thank you.
>
> Before there was any documentation, voip-info  amd this list was my
> savior.  The volume of traffic has fallen to almost nothing over the
> last year or two.
>
> I wonder if Digium could post totals as it did when I was shocked to
> find my name as the #1 poster.  It would be cool to see who is the #1
> poster now, but I am more interested in what I perceive to be a huge
> fall off of posting.
>
> It could be my email server, since I was getting notices from the list
> about excessive email bounces and removing me if I did not click a link.
> That seems to have stopped, and I don't think it was on my side.
>
> Back to getting credit or a thank you.
>
> What I have received by answering questions or helping to troubleshoot
> is worth way more than a thank you.  I get some name recognition, paid
> work, large call centers, Sr Positions in high profile jobs.  Enough to
> make a nice living, whether I am independent or in a salaried position.
> Asterisk has literally taken me all over the world.  My last trip was to
> Iraq, but I have been to Senegal, Sierra Leone, Guinea, Ghana, Liberia
> to help rebuild the infrastructure for USAID.
>
> I don't really do job searches,  I am usually offered a job or project
> and approached by the client.
>
> For the Dept of State, I set up prepaid call centers to answer questions
> and getting a reservation at the various Embassies about obtaining a
> visa to come to the US.  It is called the US visa Information Service
>
> For DoD/Dos, I cannot really say much except I can say is that I am
> probably one of the few Asterisk people that were issued a Glock and M4,
> bullet proof vests, armored cars, and a PSD team..  How many VoIP guys
> were taking ak47 rounds while I was on top of the Iraqi Government
> building, setting up the Motorola Canopy system.  Luckily the AK is no
> sniper rifle by an means.  I was in  IZ and the shooter was in the
> redzone.
>
> I don't need thank yous, although they are nice.  I truly have never
> expected anything when offering help or ideas.
>
> I do see why someone "should" be thanked, even if for nothing more than
> trying to help, and certainly a resolution to the problem for the
> archives, but I am not holding my breath.
>
> Additionally, having multiple possible answers is not a bad thing.  A
> similar symptom could be caused by many different things.  Having
> several different answers is a great help to me.
>
> At least there are many possible answers to try.  Judging by who was
> involved in any archived thread, I can usually pick the correct answer
> the first time.
>
> Thanks for listening to my perspective,
> Steve T
>


On Wed, May 11, 2011 at 3:49 AM, Andrew Thomas  wrote:
Wow! How self-promoting was that post?

As for a simple 'that worked' post - as

[asterisk-users] no audio with SIP:INFO in meetme

[Deka, Rajib IN MAA SL]

Hello List,

Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode 
enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the 
same.

Dialplan looks like
Exten => 100,1,MeetMe(100,dmF)

Sip.conf
dtmfmode=info

Regards,
Rajib


Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] no audio with SIP:INFO in meetme

[DHAVAL INDRODIYA]

Hi Rajib,

There is nothing like that Asterisk is blocking an audio if you use without
F it gives you and audio or not.

cheers
Dhaval

On Wed, May 11, 2011 at 5:34 PM, Deka, Rajib IN MAA SL <
rajib.d...@siemens.com> wrote:

>  Hello List,
>
>
>
> Asterisk is blocking audio if ‘F’ flag is enabled in meetme with DTMF mode
> enabled as INFO for SIP channel.
>
> If it is a bug in asterisk or something need to be enabled in sip.conf for
> the same.
>
>
>
> Dialplan looks like
>
> Exten => 100,1,MeetMe(100,dmF)
>
>
>
> Sip.conf
>
> dtmfmode=info
>
>
>
> Regards,
>
> Rajib
>
>
>
>
>
> *Rajib Deka*
>
> SIEMENS Ltd.
>
> Robert V Chandran Tower, First Floor, West Wing,
>
> #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
>
> www.siemens.com
>
>
>
> Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
>
>
>
> --
> Important notice: This e-mail and any attachment there to contains
> corporate proprietary information. If you have received it by mistake,
> please notify us immediately by reply e-mail and delete this e-mail and its
> attachments from your system.
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>
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Andrew Thomas]

Let's not get in to to pissing contest.  I am not new to this list (jfyi
- I am also a dCAp).  I do know who you are (and couldn't care less
anymore).  I, also, have paying customers (but don't feel the need to
gloat about it in here).  I am not pretending to know you - as I don't
know you on a personal level (and don't wish to).
 
Sorry that you feel the need to fish for compliments - but you just
don't get them like that from me (besides which - you have no bait!).
 
You carry on doing what you do - and I'll carry on doing what I do
(without braodcasting it to the community).
 
Oh, and pleae don't trip over your ego on the way out.
 
Have a nice day!
 


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Re: [asterisk-users] no audio with SIP:INFO in meetme

[Deka, Rajib IN MAA SL]

It's working now. I removed the 'm' flag from the meetme Dialplan. It was my 
mistake. Asterisk is working fine.
Exten => 100,1,MeetMe(100,dF)

Regards,
Rajib


From: Deka, Rajib IN MAA SL
Sent: Wednesday, May 11, 2011 5:35 PM
To: 'asterisk-users@lists.digium.com'
Subject: no audio with SIP:INFO in meetme

Hello List,

Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode 
enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the 
same.

Dialplan looks like
Exten => 100,1,MeetMe(100,dmF)

Sip.conf
dtmfmode=info

Regards,
Rajib


Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] asterisk HA for queue calls

[Deka, Rajib IN MAA SL]

Hi Dhaval,

Thanks for your much appreciable reply.
Sorry for late reply as I was out of office.
We considered the situation that pending queue call cannot be retrieved during 
failover, and hence it's ok with us if we loose the calls also.

Regards,
Rajib

Date: Wed, 4 May 2011 14:15:59 +0530
From: DHAVAL INDRODIYA 
Subject: Re: [asterisk-users] asterisk HA for queue calls
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: 
Content-Type: text/plain; charset="iso-8859-1"

Hi Rajib,

I think It is not possible with asterisk , as primary server goes down it
will stop asterisk services so once asterisk service down i think all
connected calls to queue will hangup automatically, and you cannot retrive
those calls as they all are disconnected .

I think you need to consider more on load balancing per asterisk server in
that case the problem of Availability is solved to some level, If You using
SIP protocol then you can think of OPENSER and from that you can use
loadbalancer which routed calls in a way an depend on machine strength.

I hope this idea will useful to solve your requirement.

Regards
Dhaval

On Wed, May 4, 2011 at 1:13 PM, Deka, Rajib IN MAA SL <
rajib.d...@siemens.com> wrote:

>  Hello List,
>
>
>
> We are running two asterisk machines in virtual IP as primary and secondary
> server.
>
> Initially virtual IP will be active in primary server; during the failure
> of primary secondary will get the virtual IP.
>
>
>
> Is there any way to retrieve pending queue calls from primary to secondary,
> in case primary fails?
>
> Does asterisk provide any interface to do it or we have to write some
> application on asterisk to do the same.
>
>
>
> Regards,
>
> Rajib
>
>
>

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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Steve Totaro]

Jealous much?

Your logic fails you.  If you didn't want to get into a pissing match then
why continue after your first statement.

dcap means nothing to me, it is like having your A+ cert but good for you.

You were the one that brought up paying customers.  I simply stated that by
helping people on the user's list for free has turned into m0oney making
enterprises.  I did not gloat.  I live in America and I am a capitalist,
making money is not a bad thing?  Why do you think making money is bad?
Please quote where I "Gloated", sorry you can't because I didn't.

You must have a reading compression problem.  I stated many times that I
didn't about a think you or anything else.  Where did I fish for
compliments?  Quote it, sorry you cannot because I didn't.

"Sorry that you feel the need to fish for compliments - but you just don't
get them like that from me (besides which - you have no bait!)."

Was that even English or gibberish.

Stating one's accomplishments is fine, it is all true.  Ego is a good thing
when kept in check.

Jealously and lack of reading and comprehension seem to be your order of the
day.

No worries, I have many jealous people, even in my immediate family.

I will continue to proper and do cool things.  I am looking at moving to
South Korea if they can pay me my target, tax free.

Thanks,
Steve Totaro

On Wed, May 11, 2011 at 8:45 AM, Andrew Thomas  wrote:

>  Let's not get in to to pissing contest.  I am not new to this list (jfyi
> - I am also a dCAp).  I do know who you are (and couldn't care less
> anymore).  I, also, have paying customers (but don't feel the need to gloat
> about it in here).  I am not pretending to know you - as I don't know you on
> a personal level (and don't wish to).
>
> Sorry that you feel the need to fish for compliments - but you just don't
> get them like that from me (besides which - you have no bait!).
>
> You carry on doing what you do - and I'll carry on doing what I do (without
> braodcasting it to the community).
>
> Oh, and pleae don't trip over your ego on the way out.
>
> Have a nice day!
>
>
>
>  If you have received this communication in error we would appreciate
> you advising us either by telephone or return of e-mail. The contents
> of this message, and any attachments, are the property of DataVox,
> and are intended for the confidential use of the named recipient only.
> If you are not the intended recipient, employee or agent responsible
> for delivery of this message to the intended recipient, take note that
> any dissemination, distribution or copying of this communication and
> its attachments is strictly prohibited, and may be subject to civil or
> criminal action for which you may be liable.
> Every effort has been made to ensure that this e-mail or any attachments
> are free from viruses. While the company has taken every reasonable
> precaution to minimise this risk, neither company, nor the sender can
> accept liability for any damage which you sustain as a result of viruses.
> It is recommended that you should carry out your own virus checks
> before opening any attachments.
>
> Registered in England. No. 27459085.
>
>
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Alex Balashov]

On 05/11/2011 09:29 AM, Steve Totaro wrote:


You must have a reading compression problem.


I would love to bzip2 or gzip the reading process.

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

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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Steve Totaro]

Alex, thanks for the laugh.

I have a wireless keyboard and the batteries are dying.  I have been lazy
and not picked up some AAAs.

I have been using spell check to help.  At least the wrong word was spelled
correctly, lol.

Or he is not really reading what I wrote which was along the same lines as
everyone else, but nobody has to post their solutions, it would be nice, but
it is a moot point.

As far as deadlines and taking a little extra time to send solved, yeah, in
a war zone, it could get you killed.

Thanks,
Steve Totaro

On Wed, May 11, 2011 at 9:32 AM, Alex Balashov wrote:

> On 05/11/2011 09:29 AM, Steve Totaro wrote:
>
>  You must have a reading compression problem.
>>
>
> I would love to bzip2 or gzip the reading process.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
>
> --
> _
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>  http://www.asterisk.org/hello
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Andrew Thomas]

Snore...
 
Now move along please...



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: 11 May 2011 14:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] When someone helps you, at least let them
know if the problem is resolved or not


Alex, thanks for the laugh.

I have a wireless keyboard and the batteries are dying.  I have been
lazy and not picked up some AAAs.

I have been using spell check to help.  At least the wrong word was
spelled correctly, lol.

Or he is not really reading what I wrote which was along the same lines
as everyone else, but nobody has to post their solutions, it would be
nice, but it is a moot point.

As far as deadlines and taking a little extra time to send solved, yeah,
in a war zone, it could get you killed.

Thanks,
Steve Totaro


On Wed, May 11, 2011 at 9:32 AM, Alex Balashov
 wrote:


On 05/11/2011 09:29 AM, Steve Totaro wrote:



You must have a reading compression problem.



I would love to bzip2 or gzip the reading process.

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


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you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
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any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
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Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Steve Totaro]

I think he does have a very lossy reading compression problem though.  It
actually works both ways.

On Wed, May 11, 2011 at 9:32 AM, Alex Balashov wrote:

> On 05/11/2011 09:29 AM, Steve Totaro wrote:
>
>  You must have a reading compression problem.
>>
>
> I would love to bzip2 or gzip the reading process.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
>
> --
> _
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Steve Totaro]

Exactly what I would expect from someone that doesn't have a leg to stand
on.

You move along, I am the one with the guns.

Thanks,
Steve Totaro

On Wed, May 11, 2011 at 9:41 AM, Andrew Thomas  wrote:

>  Snore...
>
> Now move along please...
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro
> *Sent:* 11 May 2011 14:39
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] When someone helps you, at least let them
> know if the problem is resolved or not
>
> Alex, thanks for the laugh.
>
> I have a wireless keyboard and the batteries are dying.  I have been lazy
> and not picked up some AAAs.
>
> I have been using spell check to help.  At least the wrong word was spelled
> correctly, lol.
>
> Or he is not really reading what I wrote which was along the same lines as
> everyone else, but nobody has to post their solutions, it would be nice, but
> it is a moot point.
>
> As far as deadlines and taking a little extra time to send solved, yeah, in
> a war zone, it could get you killed.
>
> Thanks,
> Steve Totaro
>
> On Wed, May 11, 2011 at 9:32 AM, Alex Balashov 
> wrote:
>
>> On 05/11/2011 09:29 AM, Steve Totaro wrote:
>>
>> You must have a reading compression problem.
>>>
>>
>> I would love to bzip2 or gzip the reading process.
>>
>> --
>> Alex Balashov - Principal
>> Evariste Systems LLC
>> 260 Peachtree Street NW
>> Suite 2200
>> Atlanta, GA 30303
>> Tel: +1-678-954-0670
>> Fax: +1-404-961-1892
>> Web: http://www.evaristesys.com/
>>
>>
>> --
>> _
>> -- 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>  If you have received this communication in error we would appreciate
> you advising us either by telephone or return of e-mail. The contents
> of this message, and any attachments, are the property of DataVox,
> and are intended for the confidential use of the named recipient only.
> If you are not the intended recipient, employee or agent responsible
> for delivery of this message to the intended recipient, take note that
> any dissemination, distribution or copying of this communication and
> its attachments is strictly prohibited, and may be subject to civil or
> criminal action for which you may be liable.
> Every effort has been made to ensure that this e-mail or any attachments
> are free from viruses. While the company has taken every reasonable
> precaution to minimise this risk, neither company, nor the sender can
> accept liability for any damage which you sustain as a result of viruses.
> It is recommended that you should carry out your own virus checks
> before opening any attachments.
>
> Registered in England. No. 27459085.
>
>
> --
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Re: [asterisk-users] Tricky: Progress, Delay, DTMF / background calling

[Markus]

Hi again,

no one got an idea? :-(   Or did my request not make any sense? Or is the
answer to obvious that no one bothers to reply? :-)

Thanks again!


> On a second thought, I don't need the predetermined delay. I can probably
> just set that with additional w's in the DialBackground command (which I
> made up).
>
> So rather something like:
>
> _X.,1,Progress
> _X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}ww))
> _X.,3,ConnectLegs
>
> Thanks again.
>
>
>> Hi,
>>
>> has the following been done before respectively is it possible with
>> Asterisk? I searched the archives but couldn't locate anything.
>>
>> 1. Call to  comes in via SIP.
>> 2. Call is not answered yet but progress continues.
>> 3. At the moment the call comes in something like this gets spawned in
>> the
>> background:
>>
>> Dial(SIP/123456@provider,,D(ww${EXTEN})
>> which should translate to:
>> Dial(SIP/123456@provider,,D(ww)
>> But even better would be take the ${EXTEN} and put some w's between
>> them:
>> Dial(SIP/123456@provider,,D(ww5ww5ww5ww5)
>>
>> 4. After a pretermined amount of time since the call came in
>> respectively
>> the Dial command was spawned "in the background", e.g. 15 seconds,
>> Asterisk answers the call and the call legs are connected together.
>>
>> So, with some fantasy commands, something like this:
>>
>> _X.,1,Progress
>> _X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}),ANSWER-AND-CONNECT-LEGS(15)
>>
>> I hope my request is not too cryptic. In short: I'd like to receive
>> calls
>> to arbitrary extensions, but not answer them directly, only after a Dial
>> command has been spawned and a predetermined amount of time has passed
>> since the Dial command has been spawned / since the Dial command has
>> connected to 123456.
>>
>> Possible?
>>
>> I'm new to the list, hi! :)
>>
>> Thank you!
>>
>>
>>
>>
>>
>> --
>> _
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>
>
>
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Re: [asterisk-users] Asterisk 1.8.4 Now Available

[Jeremy Kister]

On 5/10/2011 10:38 AM, Asterisk Development Team wrote:

Below is a sample of the issues resolved in this release:

[...]

For a full list of changes in this release candidate, please see the ChangeLog:


I'm a bit confused about this release (and previous releases on the 1.8 
track) so please bare with me.


I viewed the ChangeLog, but I don't see any of the 'sample issues' 
listed.  why is that ?  I would expect to see the 'sample issues' listed 
after 1.8.4-rc3.


Also, is there a reason/procedural error that patches such as:
https://issues.asterisk.org/view.php?id=18382
https://issues.asterisk.org/view.php?id=18742

didnt make it into this 1.8.4 release ?


--

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http://jeremy.kister.net./

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Re: [asterisk-users] Asterisk 1.8.4 Now Available

[Paul Belanger]

On 11-05-11 10:29 AM, Jeremy Kister wrote:

I'm a bit confused about this release (and previous releases on the 1.8
track) so please bare with me.

I viewed the ChangeLog, but I don't see any of the 'sample issues'
listed. why is that ? I would expect to see the 'sample issues' listed
after 1.8.4-rc3.

Also, is there a reason/procedural error that patches such as:
https://issues.asterisk.org/view.php?id=18382
https://issues.asterisk.org/view.php?id=18742

didnt make it into this 1.8.4 release ?


Correct, they will appear in 1.8.5-rc1 forward. When -rc1 is created, it 
will be tagged from the HEAD of branches/1.8..  If 1.8.5-rc2 is create, 
it is because of an issue / bug was found in 1.8.5-rc1, and will include 
that fix only.


If a new issue is reported after 1.8.5-rc1 and fixed in branches/1.8, it 
will not be added into 1.8.5 release, but will wait until 1.8.6-rc1.


--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Asterisk 1.8.4 Now Available

[Leif Madsen]

On 11-05-11 10:46 AM, Paul Belanger wrote:
> On 11-05-11 10:29 AM, Jeremy Kister wrote:
>> I'm a bit confused about this release (and previous releases on the 1.8
>> track) so please bare with me.
>>
>> I viewed the ChangeLog, but I don't see any of the 'sample issues'
>> listed. why is that ? I would expect to see the 'sample issues' listed
>> after 1.8.4-rc3.
>>
>> Also, is there a reason/procedural error that patches such as:
>> https://issues.asterisk.org/view.php?id=18382
>> https://issues.asterisk.org/view.php?id=18742
>>
>> didnt make it into this 1.8.4 release ?
>>
>>
> Correct, they will appear in 1.8.5-rc1 forward. When -rc1 is created, it will 
> be
> tagged from the HEAD of branches/1.8..  If 1.8.5-rc2 is create, it is because 
> of
> an issue / bug was found in 1.8.5-rc1, and will include that fix only.
> 
> If a new issue is reported after 1.8.5-rc1 and fixed in branches/1.8, it will
> not be added into 1.8.5 release, but will wait until 1.8.6-rc1.
> 

More information about this is documented at
http://blogs.asterisk.org/2010/09/02/the-monthly-asterisk-release-cycle/

Basically, it doesn't make sense to do release candidate RC1+x from the head of
branches/1.8, because then there'd never be a solid base to test from. When we
do an RC1, it is pulled directly from the branch, which gets all changes since
the previous RC1.

Once and RC1 is created, if something is deemed to trigger a new RC (a
regression is a good example), then the RC1 is copied to RC2, and the specific
changes are merged into that RC. No further changes other than what was merged
in are added (i.e. not all changes in the branch are part of RC2). If additional
changes need to be made, then RC2 is copied to RC3, and specific fixes are
merged to that RC.

This continues until the full release is made, which is an exact copy of the
latest RC. So if we had an RC3, then RC3 is copied, without changes, to the
release version. So in this case, tags/1.8.4-rc3 as copied to tags/1.8.4, and
the only changes were made to the .version file and ChangeLog. Then the standard
release process is followed to turn that tag into a .tar.gz and get it onto the
downloads site.

Any changes made after 1.8.4-rc1 (for example) would then become available in
1.8.5-rc1, because only RC1s contain all changes from the branch directly.

HTH,
Leif Madsen.

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Re: [asterisk-users] Asterisk 1.8.4 Now Available

[Danny Nicholas]

 
[Danny Nicholas]  
Paul, this is probably a "dumb question", but why are some (or is it all and
I just don't notice it) modules "fundamentally changed" from release to
release (or version to version)?  As a C-dabbler, it seems to me that if I
do gcc app_voicemail.c (using voicemail as an example) on 1.6 or 1.8, it
should be fundamentally similar to doing the same thing in 1.4.  It seems to
me that the 9500 line module that compiles and runs in 1.4 should be pretty
much the same as what is in 1.6 or 1.8, but on closer inspection it is often
not.  I know that some things have to change to add the new enhanced
functionality, but from a "dinosaur programmer" perspective, the fewer
background changes you have from release to release and version to version,
the less chance you have of something breaking and having unhappy users.

Please elaborate on what is wrong with my thought.

Thanks in advance.


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[asterisk-users] With what options is asterisk compiled in rpm's

[isrlgb]

Hi,

I'm trying to add modules compiled from source into a rpm install of asterisk 
(from digium) on centos and asterisk complains that its not compiled with same 
options so it won't load it

I know I could install the entire thing from source but for other reasons I 
would like to keep the main things installed from rpm and install whatever else 
I need from source (or roll my own rpm for those) 

Thanks,
Israel 

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Re: [asterisk-users] With what options is asterisk compiled in rpm's

[Danny Nicholas]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of isr...@gmail.com
> Sent: Wednesday, May 11, 2011 10:07 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] With what options is asterisk compiled in rpm's
> 
> Hi,
> 
> I'm trying to add modules compiled from source into a rpm install of
> asterisk (from digium) on centos and asterisk complains that its not
> compiled with same options so it won't load it
> 
> I know I could install the entire thing from source but for other reasons
> I would like to keep the main things installed from rpm and install
> whatever else I need from source (or roll my own rpm for those)
> 
> Thanks,
> Israel
[Danny Nicholas] 
This might work for you - download the source and do ./configure
--libdir=/fromsrclib --bindir=/fromsrcbin

Once you've done the make menuselect, make and make install, you should be
able to copy the module(s) from /fromsrclib to /usr/lib/asterisk/modules 



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Re: [asterisk-users] Asterisk 1.8.4 Now Available

[|| dave cantera Mobile]

danny,
not that it matters, but I agree. if the design is a good design, it 
would not have to be redesigned on every release.  in fact, the modules 
template should also follow this philosophy that way you can concentrate 
on adding functions and not the design...


sometimes, it is smarter to scrap what you have and use the knowledge 
gained to come up with a design that provides for a good, tightly 
integrated, but flexible facility to interact with multiple components.  
for example, the admin interface on * is just such of a system. you can 
add commands as required (I haven't looked at the code to see if it is 
easy to do though). 

the more time you put into design, the less time you have to spend 
programming.

IMHO,
daveC

Danny Nicholas wrote:
 
[Danny Nicholas]  
Paul, this is probably a "dumb question", but why are some (or is it all and

I just don't notice it) modules "fundamentally changed" from release to
release (or version to version)?  As a C-dabbler, it seems to me that if I
do gcc app_voicemail.c (using voicemail as an example) on 1.6 or 1.8, it
should be fundamentally similar to doing the same thing in 1.4.  It seems to
me that the 9500 line module that compiles and runs in 1.4 should be pretty
much the same as what is in 1.6 or 1.8, but on closer inspection it is often
not.  I know that some things have to change to add the new enhanced
functionality, but from a "dinosaur programmer" perspective, the fewer
background changes you have from release to release and version to version,
the less chance you have of something breaking and having unhappy users.

Please elaborate on what is wrong with my thought.

Thanks in advance.


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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Sherwood McGowan]

Wow...somehow this turned into a something so much darker than the original
intent*sits back and watches the show*

Thanks guys, that little mini bonfire made an otherwise boring day into an
entertaining Asterisk-Users version of WWE Raw.

Cheers!
Sherwood McGowan
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Re: [asterisk-users] Tricky: Progress, Delay, DTMF / background calling

[Sherwood McGowan]

Try reading up on Local channels, it will accomplish everything you wish.



On Wed, May 11, 2011 at 8:59 AM, Markus  wrote:

> Hi again,
>
> no one got an idea? :-(   Or did my request not make any sense? Or is the
> answer to obvious that no one bothers to reply? :-)
>
> Thanks again!
>
>
> > On a second thought, I don't need the predetermined delay. I can probably
> > just set that with additional w's in the DialBackground command (which I
> > made up).
> >
> > So rather something like:
> >
> > _X.,1,Progress
> > _X.,2,DialBackground(SIP/123456@provider
> ,,D(ww${EwwXwwTwwEwwN}ww))
> > _X.,3,ConnectLegs
> >
> > Thanks again.
> >
> >
> >> Hi,
> >>
> >> has the following been done before respectively is it possible with
> >> Asterisk? I searched the archives but couldn't locate anything.
> >>
> >> 1. Call to  comes in via SIP.
> >> 2. Call is not answered yet but progress continues.
> >> 3. At the moment the call comes in something like this gets spawned in
> >> the
> >> background:
> >>
> >> Dial(SIP/123456@provider,,D(ww${EXTEN})
> >> which should translate to:
> >> Dial(SIP/123456@provider,,D(ww)
> >> But even better would be take the ${EXTEN} and put some w's between
> >> them:
> >> Dial(SIP/123456@provider,,D(ww5ww5ww5ww5)
> >>
> >> 4. After a pretermined amount of time since the call came in
> >> respectively
> >> the Dial command was spawned "in the background", e.g. 15 seconds,
> >> Asterisk answers the call and the call legs are connected together.
> >>
> >> So, with some fantasy commands, something like this:
> >>
> >> _X.,1,Progress
> >> _X.,2,DialBackground(SIP/123456@provider
> ,,D(ww${EwwXwwTwwEwwN}),ANSWER-AND-CONNECT-LEGS(15)
> >>
> >> I hope my request is not too cryptic. In short: I'd like to receive
> >> calls
> >> to arbitrary extensions, but not answer them directly, only after a Dial
> >> command has been spawned and a predetermined amount of time has passed
> >> since the Dial command has been spawned / since the Dial command has
> >> connected to 123456.
> >>
> >> Possible?
> >>
> >> I'm new to the list, hi! :)
> >>
> >> Thank you!
> >>
> >>
> >>
> >>
> >>
> >> --
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> >>
> >
> >
> >
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> >
>
>
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Telecommunications and VOIP Consultant
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Steve Totaro]

Yeah, I am not sure why dude went on the offensive.  Got emotional but could
not produce a single example of the name calling and insults he was hurling
at me.

Here is an email I received a very short time ago.  Sender and company's
name have been removed.

*| to Steve *
*show details 9:15 AM (2 hours ago)*
*
* * steve,
I haven't been active in the * community for a while but ran across an
interesting project that I would like to pursue. [COMPANY] in springfield
needs a * admin part time and I could use the steady income plus I would
like to get my hands back into *...  I wanted to check with you first
because this is your neck of the woods...  do you have any experience with
them?

recently, I have been just lurking on the [asterisk-] lists. thanks for
supporting the [asterisk-] groups in a big way. *



On Wed, May 11, 2011 at 11:28 AM, Sherwood McGowan <
sherwood.mcgo...@gmail.com> wrote:

> Wow...somehow this turned into a something so much darker than the original
> intent*sits back and watches the show*
>
> Thanks guys, that little mini bonfire made an otherwise boring day into an
> entertaining Asterisk-Users version of WWE Raw.
>
> Cheers!
> Sherwood McGowan
>
>
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Myles Wakeham]

On 05/11/2011 08:39 AM, asterisk-users-requ...@lists.digium.com wrote:

Snore...

Now move along please...



OK, but how about you respecting some basic mailing list etiquette and 
not quoting the entire thread in your posts so that those of us who wade 
through these messages as digests don't get carpal tunnel with the 
scroll wheel on the mouse sifting through your pissing match with 
another on this list.


Take it outside.  Your welcome.

Myles


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[asterisk-users] CLI - displaying all channel variables

[Paddy Grice]

Hi List
 
This may be a silly question by web searches etc don't seem to answer it. 
 
Is there a CLI command to display ALL channel variables - standard and user
created - for a specific channel?
 
something like show channel SIP/Test123 all
 
I'm using Version 1.4.33.1
 
PG
 
 
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Re: [asterisk-users] CLI - displaying all channel variables

[Eric Wieling]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Paddy Grice
> Sent: Wednesday, May 11, 2011 11:49 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] CLI - displaying all channel variables
>
> Hi List
>
> This may be a silly question by web searches etc don't seem
> to answer it.
>
> Is there a CLI command to display ALL channel variables -
> standard and user created - for a specific channel?
>
> something like show channel SIP/Test123 all

The dialplan application DumpChan dumps information about the channel, however, 
it does not display all the variables you are looking for.  Generally you 
should insert a Noop in the dialplan to examine variables.Noop(EXTEN is 
${EXTEN}) for example.

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Re: [asterisk-users] CLI - displaying all channel variables

[Danny Nicholas]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Eric Wieling
> Sent: Wednesday, May 11, 2011 10:55 AM
> To: pa...@wizaner.com; Asterisk Users Mailing List -Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] CLI - displaying all channel variables
> 
> 
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > Paddy Grice
> > Sent: Wednesday, May 11, 2011 11:49 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] CLI - displaying all channel variables
> >
> > Hi List
> >
> > This may be a silly question by web searches etc don't seem
> > to answer it.
> >
> > Is there a CLI command to display ALL channel variables -
> > standard and user created - for a specific channel?
> >
> > something like show channel SIP/Test123 all
> 
> The dialplan application DumpChan dumps information about the channel,
> however, it does not display all the variables you are looking for.
> Generally you should insert a Noop in the dialplan to examine variables.
> Noop(EXTEN is ${EXTEN}) for example.
> 
[Danny Nicholas] 
Try core show channel sip/test123-001 - you will probably have to do a
core show channels first to get the proper -001 value.




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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Andrew Thomas]

Seems I have upset the God that is Steve Totaro!
 
You want an example?  OK - your last post.  Has nothing to do with the
thread (or our 'discussion') but yet you chose to post it as yet another
self pat-on-the-back!  I could produce a lot more - but you now bore me.
 
You know it must be so hard being so perfect Steve.  I so wish I was
you!
 
Have a really nice day :)



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: 11 May 2011 16:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] When someone helps you, at least let them
know if the problem is resolved or not


Yeah, I am not sure why dude went on the offensive.  Got emotional but
could not produce a single example of the name calling and insults he
was hurling at me.

Here is an email I received a very short time ago.  Sender and company's
name have been removed.  


| to Steve 
show details 9:15 AM (2 hours ago) 


steve,
I haven't been active in the * community for a while but ran across an
interesting project that I would like to pursue. [COMPANY] in
springfield needs a * admin part time and I could use the steady income
plus I would like to get my hands back into *...  I wanted to check with
you first because this is your neck of the woods...  do you have any
experience with them?

recently, I have been just lurking on the [asterisk-] lists. thanks for
supporting the [asterisk-] groups in a big way. 




On Wed, May 11, 2011 at 11:28 AM, Sherwood McGowan
 wrote:


Wow...somehow this turned into a something so much darker than
the original intent*sits back and watches the show*

Thanks guys, that little mini bonfire made an otherwise boring
day into an entertaining Asterisk-Users version of WWE Raw.

Cheers!  
Sherwood McGowan


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Re: [asterisk-users] CLI - displaying all channel variables

[Paddy Grice]

Thanks Danny - That displays "user" created variables - by "user" this could
be application like dial but not the predefined channel variables.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 11 May 2011 17:03
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CLI - displaying all channel variables

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- 
> boun...@lists.digium.com] On Behalf Of Eric Wieling
> Sent: Wednesday, May 11, 2011 10:55 AM
> To: pa...@wizaner.com; Asterisk Users Mailing List -Non-Commercial 
> Discussion
> Subject: Re: [asterisk-users] CLI - displaying all channel variables
> 
> 
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy 
> > Grice
> > Sent: Wednesday, May 11, 2011 11:49 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] CLI - displaying all channel variables
> >
> > Hi List
> >
> > This may be a silly question by web searches etc don't seem to 
> > answer it.
> >
> > Is there a CLI command to display ALL channel variables - standard 
> > and user created - for a specific channel?
> >
> > something like show channel SIP/Test123 all
> 
> The dialplan application DumpChan dumps information about the channel, 
> however, it does not display all the variables you are looking for.
> Generally you should insert a Noop in the dialplan to examine variables.
> Noop(EXTEN is ${EXTEN}) for example.
> 
[Danny Nicholas]
Try core show channel sip/test123-001 - you will probably have to do a
core show channels first to get the proper -001 value.




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Re: [asterisk-users] CLI - displaying all channel variables

[Danny Nicholas]

You should probably use an AGI to get this rather than depending on CLI
commands.  It's possible that you could do a bash AGI and call that from CLI
but that's not something I've dabbled with.

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Paddy Grice
> Sent: Wednesday, May 11, 2011 11:21 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] CLI - displaying all channel variables
> 
> Thanks Danny - That displays "user" created variables - by "user" this
> could
> be application like dial but not the predefined channel variables.
> 
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
> Nicholas
> Sent: 11 May 2011 17:03
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] CLI - displaying all channel variables
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Eric Wieling
> > Sent: Wednesday, May 11, 2011 10:55 AM
> > To: pa...@wizaner.com; Asterisk Users Mailing List -Non-Commercial
> > Discussion
> > Subject: Re: [asterisk-users] CLI - displaying all channel variables
> >
> >
> >
> > > -Original Message-
> > > From: asterisk-users-boun...@lists.digium.com
> > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy
> > > Grice
> > > Sent: Wednesday, May 11, 2011 11:49 AM
> > > To: asterisk-users@lists.digium.com
> > > Subject: [asterisk-users] CLI - displaying all channel variables
> > >
> > > Hi List
> > >
> > > This may be a silly question by web searches etc don't seem to
> > > answer it.
> > >
> > > Is there a CLI command to display ALL channel variables - standard
> > > and user created - for a specific channel?
> > >
> > > something like show channel SIP/Test123 all
> >
> > The dialplan application DumpChan dumps information about the channel,
> > however, it does not display all the variables you are looking for.
> > Generally you should insert a Noop in the dialplan to examine variables.
> > Noop(EXTEN is ${EXTEN}) for example.
> >
> [Danny Nicholas]
> Try core show channel sip/test123-001 - you will probably have to do a
> core show channels first to get the proper -001 value.
> 
> 
> 
> 
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Re: [asterisk-users] CLI - displaying all channel variables

[Steve Edwards]

On Wed, 11 May 2011, Eric Wieling wrote:

Generally you should insert a Noop in the dialplan to examine variables. 
Noop(EXTEN is ${EXTEN}) for example.


The 'verbose()' application would be an example of 'better practices.'

It's function is obvious rather than just a convenient side-effect.

It has additional functionality in that you can specify the 'verbosity' 
level needed.


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-
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

[Steve Totaro]

I am not upset in least, well I am but that's because I own thousands of
ounces of silver bullion and I am watching in get pummeled again.  Good
thing I bought the bulk of it when it was only $12 an ounce.

http://www.kitco.com/charts/livesilver.html

You are an angry person and it is sad.

It is also sad that the example I requested earlier is something posted
later.  The only reason for that is because you had nothing to back up any
of your rage.

Seek help, please.  If you feel like you want to hurt yourself or others,
have yourself committed right away.  I am serious.  If you are voluntary,
you can leave when you want.

Thanks,
Steve Totaro

On Wed, May 11, 2011 at 12:13 PM, Andrew Thomas  wrote:

>  Seems I have upset the God that is Steve Totaro!
>
> You want an example?  OK - your last post.  Has nothing to do with the
> thread (or our 'discussion') but yet you chose to post it as yet another
> self pat-on-the-back!  I could produce a lot more - but you now bore me.
>
> You know it must be so hard being so perfect Steve.  I so wish I was you!
>
> Have a really nice day :)
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro
> *Sent:* 11 May 2011 16:38
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] When someone helps you, at least let them
> know if the problem is resolved or not
>
> Yeah, I am not sure why dude went on the offensive.  Got emotional but
> could not produce a single example of the name calling and insults he was
> hurling at me.
>
> Here is an email I received a very short time ago.  Sender and company's
> name have been removed.
>
> *| to Steve *
>  *show details 9:15 AM (2 hours ago)*
> *
> **steve,
> I haven't been active in the * community for a while but ran across an
> interesting project that I would like to pursue. [COMPANY] in springfield
> needs a * admin part time and I could use the steady income plus I would
> like to get my hands back into *...  I wanted to check with you first
> because this is your neck of the woods...  do you have any experience with
> them?
>
> recently, I have been just lurking on the [asterisk-] lists. thanks for
> supporting the [asterisk-] groups in a big way. *
>
>
>
> On Wed, May 11, 2011 at 11:28 AM, Sherwood McGowan <
> sherwood.mcgo...@gmail.com> wrote:
>
>> Wow...somehow this turned into a something so much darker than the
>> original intent*sits back and watches the show*
>>
>> Thanks guys, that little mini bonfire made an otherwise boring day into an
>> entertaining Asterisk-Users version of WWE Raw.
>>
>> Cheers!
>> Sherwood McGowan
>>
>>
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[asterisk-users] concurrent call tracking

[Skyler]

Hi all,

 

 I would like to track/store concurrent call usage per user by
day/week/month and get server totals by day/week/month. Google comes up with
mostly info regarding concurrent call limits, though my goal is to calculate
actual concurrent channel usage and add it into reporting. I'm using * 1.6.2
+ mysql - realtime (no gui). Any suggestions / open-source / AGI on where to
start looking into implementing something like this? 

 

TIA,

 

Skyler

 

 

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Re: [asterisk-users] concurrent call tracking

[Leif Madsen]

On 11-05-11 12:57 PM, Skyler wrote:
>  I would like to track/store concurrent call usage per user by
> day/week/month and get server totals by day/week/month. Google comes up with
> mostly info regarding concurrent call limits, though my goal is to calculate
> actual concurrent channel usage and add it into reporting. I'm using * 1.6.2
> + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to
> start looking into implementing something like this? 

Just use SNMP to get the channel usage. If you don't want to use SNMP, then just
use something like GROUP(), GROUP_COUNT() and func_odbc to write channel usage
to the database. Something like


[Outgoing]
exten => _NXXNXX,1,NoOp()
same => n,GoSub(subTotalCallCounter,start,1(outgoing))

[subTotalCallCounter]
exten => start,1,NoOp()
same => n,Set(GROUP(totalcalls)=${ARG1})
same => n,Set(ODBC_TOTAL_CALLS(${ARG1})=${GROUP_COUNT(${ARG1}@totalcalls)})
same => n,Return()

[Incoming]
exten => 4165551212,1,NoOp()
same => n,GoSub(subTotalCallCounter,start,1(incoming))

[LocalSets]
exten => _1XX,1,NoOp()
same => n,GoSub(subTotalCallCounter,start,1(internal))




func_odbc
-

[TOTAL_CALLS]
dsn=myDatabase
writesql=INSERT INTO totalCalls ('type','callcount') VALUES 
('${VAL1}','${ARG1}')




Something like that. Totally untested and only written in this email :)

Leif.

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[asterisk-users] kernel: dahdi: Master changed to B4/0/x

[Laurent Caron]

Hi,

I did replace an old asterisk box by a new shiny one (2 BRI ports used
on a quad port card - BeroNet PCI Express)

I noticed a message in the logs that puzzles me:

May 11 19:10:02  kernel: dahdi: Master changed to B4/0/1
May 11 19:10:08  kernel: wcb4xxp :05:04.0: PCI INT A disabled
May 11 19:10:08  kernel: wcb4xxp :05:04.0: Driver unloaded.
May 11 19:10:08  kernel: dahdi: Telephony Interface Unloaded
May 11 19:10:09  kernel: dahdi: Telephony Interface Registered on major 196
May 11 19:10:09  kernel: dahdi: Version: 2.3.0.1
May 11 19:10:09  kernel: wcb4xxp :05:04.0: probe called for b4xx...
May 11 19:10:09  kernel: wcb4xxp :05:04.0: PCI INT A -> GSI 32 (level, low) 
-> IRQ 32
May 11 19:10:09  kernel: wcb4xxp :05:04.0: Identified BeroNet BN4S0 
(controller rev 1) at 0001bc00, IRQ 32
May 11 19:10:09  kernel: wcb4xxp :05:04.0: NOTE: hardware echo cancellation 
has been disabled
May 11 19:10:09  kernel: wcb4xxp :05:04.0: Port 1: TE mode
May 11 19:10:09  kernel: wcb4xxp :05:04.0: Port 2: TE mode
May 11 19:10:09  kernel: wcb4xxp :05:04.0: Port 3: TE mode
May 11 19:10:09  kernel: wcb4xxp :05:04.0: Port 4: TE mode
May 11 19:10:09  kernel: wcb4xxp :05:04.0: Did not do the highestorder stuff
May 11 19:10:09  kernel: dahdi_echocan_oslec: Registered echo canceler 'OSLEC'
May 11 19:10:09  kernel: dahdi: Registered tone zone 2 (France)
May 11 19:10:09  kernel: wcb4xxp :05:04.0: new card sync source: port 1

May 11 19:10:30  kernel: dahdi: Master changed to B4/0/2
May 11 19:10:36  kernel: dahdi: Master changed to B4/0/1
May 11 19:10:56  kernel: dahdi: Master changed to B4/0/2
May 11 19:11:02  kernel: dahdi: Master changed to B4/0/1
May 11 19:11:12  kernel: dahdi: Master changed to B4/0/2
May 11 19:12:07  kernel: dahdi: Master changed to B4/0/1
May 11 19:12:17  kernel: dahdi: Master changed to B4/0/2
May 11 19:12:19  kernel: dahdi: Master changed to B4/0/1
May 11 19:12:40  kernel: dahdi: Master changed to B4/0/2
May 11 19:12:42  kernel: dahdi: Master changed to B4/0/1

When those messages appear a RED alarm is triggered on the corresponding
span.



Hardware:
05:04.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)

Modules:
dahdi_echocan_oslec 1386  8 
echo3354  1 dahdi_echocan_oslec
wcb4xxp74747  6 
dahdi 193086  14 dahdi_echocan_oslec,wcb4xxp

version:2.3.0.1
alias:  dahdi_dummy


asterisk version:
Asterisk 1.6.2.9-2, Copyright (C) 1999 - 2010 Digium, Inc. and others.

Channel dumps:
May 11 19:18:19 kernel: dahdi: Dump of DAHDI Channel 1 (B4/0/1/1,1,1):
May 11 19:18:19 kernel:
May 11 19:18:19 kernel: dahdi: flags: 501 hex, writechunk: 88013b74008c, 
readchunk: 88013b7400a4
May 11 19:18:19 kernel: dahdi: rxgain: a0276170, txgain: 
a0276170, gainalloc: 0
May 11 19:18:19 kernel: dahdi: span: 88013b740128, sig: 80 hex, sigcap: 
10080 hex
May 11 19:18:19 kernel: dahdi: inreadbuf: -1, outreadbuf: 0, inwritebuf: 0, 
outwritebuf: -1
May 11 19:18:19 kernel: dahdi: blocksize: 160, numbufs: 4, txbufpolicy: 0, 
txbufpolicy: 0
May 11 19:18:19 kernel: dahdi: txdisable: 0, rxdisable: 0, iomask: 0
May 11 19:18:19 kernel: dahdi: curzone: 88013cce, tonezone: 2, curtone: 
(null), tonep: 0
May 11 19:18:19 kernel: dahdi: digitmode: 0, txdialbuf: , dialing: 0, 
aftdialtimer: 0, cadpos. 0
May 11 19:18:19 kernel: dahdi: confna: 0, confn: 0, confmode: 0, confmute: 0
May 11 19:18:19 kernel: dahdi: ec: (null), deflaw: 0, xlaw: a0257d50
May 11 19:18:19 kernel: dahdi: itimer: 0, otimer: 0, ringdebtimer: 0
May 11 19:18:19 kernel:
May 11 19:18:21 kernel: dahdi: Dump of DAHDI Channel 2 (B4/0/1/2,2,2):
May 11 19:18:21 kernel:
May 11 19:18:21 kernel: dahdi: flags: 501 hex, writechunk: 88013b740094, 
readchunk: 88013b7400ac
May 11 19:18:21 kernel: dahdi: rxgain: a0276170, txgain: 
a0276170, gainalloc: 0
May 11 19:18:21 kernel: dahdi: span: 88013b740128, sig: 80 hex, sigcap: 
10080 hex
May 11 19:18:21 kernel: dahdi: inreadbuf: -1, outreadbuf: 0, inwritebuf: 0, 
outwritebuf: -1
May 11 19:18:21 kernel: dahdi: blocksize: 160, numbufs: 4, txbufpolicy: 0, 
txbufpolicy: 0
May 11 19:18:21 kernel: dahdi: txdisable: 0, rxdisable: 0, iomask: 0
May 11 19:18:21 kernel: dahdi: curzone: 88013cce, tonezone: 2, curtone: 
(null), tonep: 0
May 11 19:18:21 kernel: dahdi: digitmode: 0, txdialbuf: , dialing: 0, 
aftdialtimer: 0, cadpos. 0
May 11 19:18:21 kernel: dahdi: confna: 0, confn: 0, confmode: 0, confmute: 0
May 11 19:18:21 kernel: dahdi: ec: (null), deflaw: 0, xlaw: a0257d50
May 11 19:18:21 kernel: dahdi: itimer: 0, otimer: 0, ringdebtimer: 0


Do you guys have a clue about it ?

Thanks



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Re: [asterisk-users] concurrent call tracking

[Stelios Koroneos]

You can use the manager api (interface) and "poll" that info and then
store it in a MYSQL table etc.
You can do this outside asterisk,even from a different machine using
your preferred  dev language as there are manager libraries/bindings for
most major dev languages

'Actual' is the key word though.
To get the actual concurrent channels you should poll the system, at
least every second, and that means 3600 records per hour or 86.400 per
day. That would end up taking a alot of time to average using mysql
queries.

Alternatively you could do N minutes averages and store them in the db
i.e read every second but save the average of 60 reads which is 1 minute
etc


Stelios


On Wed, 2011-05-11 at 09:57 -0700, Skyler wrote:
> Hi all,
>  
>  
> 
>  I would like to track/store concurrent call usage per user by
> day/week/month and get server totals by day/week/month. Google comes
> up with mostly info regarding concurrent call limits, though my goal
> is to calculate actual concurrent channel usage and add it into
> reporting. I’m using * 1.6.2 + mysql – realtime (no gui). Any
> suggestions / open-source / AGI on where to start looking into
> implementing something like this? 
> 
>  
> 
> TIA,
> 
>  
> 
> Skyler
> 
>  
> 
>  
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> 
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>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] Asterisk SIP Trunking with Cisco UC 560

[Darrin Henshaw]

Hello,

I'm interested in knowing if anyone out there has successfully connected
Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that
we put in an Asterisk install, one of their sister companies who we don't
control is putting in a Cisco UC 560. From my looking I think it can be
done, but the vendor is telling them it can't. Thought I'd ask around here
and see if anyone has done it? Thanks.

Cheers,

Darrin Henshaw
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Re: [asterisk-users] Asterisk SIP Trunking with Cisco UC 560

[Alex Balashov]

On 05/11/2011 01:30 PM, Darrin Henshaw wrote:


I'm interested in knowing if anyone out there has successfully
connected Asterisk to a Cisco UC 560 via SIP trunking? We have a
client of ours that we put in an Asterisk install, one of their
sister companies who we don't control is putting in a Cisco UC 560.
From my looking I think it can be done, but the vendor is telling
them it can't. Thought I'd ask around here and see if anyone has done
it? Thanks.


I see no reason why it can't be done in principle;  the vendor or VAR is 
just being dismissive of open-source, lest the customer suspect they 
might be able to do everything the UC560 can do with open-source 
components without paying for a UC560.


The only sticking point is whether this endpoint, like Cisco Call 
Manager, by default does SDP offer in the 200 OK and answer in the ACK. 
 This is an SDP offer-answer flow prescribed by RFC 3261 as valid but 
not supported by Asterisk, as best as I can tell, even in 1.8.


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

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Re: [asterisk-users] Asterisk SIP Trunking with Cisco UC 560

[Warren Selby]

On Wed, May 11, 2011 at 12:30 PM, Darrin Henshaw
wrote:

> Hello,
>
> I'm interested in knowing if anyone out there has successfully connected
> Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that
> we put in an Asterisk install, one of their sister companies who we don't
> control is putting in a Cisco UC 560. From my looking I think it can be
> done, but the vendor is telling them it can't. Thought I'd ask around here
> and see if anyone has done it? Thanks.
>
> Cheers,
>
> Darrin Henshaw
>

>From a brief glance through the UC560 specs, it looks as though you should
be able to setup a SIP trunk.  I've been able to successfully integrate a
Cisco CallManager 7.x system with Asterisk using SIP trunking, so I imagine
you should be able to do the same here.

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http://www.selbytech.com
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Re: [asterisk-users] CLI - displaying all channel variables

[Leif Madsen]

On 11-05-11 12:29 PM, Steve Edwards wrote:
> On Wed, 11 May 2011, Eric Wieling wrote:
> 
>> Generally you should insert a Noop in the dialplan to examine variables.
>> Noop(EXTEN is ${EXTEN}) for example.
> 
> The 'verbose()' application would be an example of 'better practices.'
> 
> It's function is obvious rather than just a convenient side-effect.
> 
> It has additional functionality in that you can specify the 'verbosity' level
> needed.

Agreed. I tend to use NoOp() for an actual No Operation, such as using it on the
first line of an extension:

exten => something_awesome,1,NoOp()
same => n,Verbose(2,Incoming call from ${CALLERID(all)})
same => n,Dial(SIP/someone_awesome)
same => n,Hangup

That way if you want to place things ahead of any line, you can do that without
impunity. Even using Verbose() on the first line can cause problems if you want
to move the Verbose() around and place something before it -- now you have to do
some copy/pasting, and extra work that could be avoided :)

Leif.

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Re: [asterisk-users] concurrent call tracking

[Dovid Bender]

What I do is when ever a call comes in I update a table in MySQL to active = 
(active +1). On hang up I do active = (active -1).

I have a cron that checks once a minute to see how many active and stores it 
along with epoch in db.

I then have a graph that shows channel usage. If you want the code let me know.

  - Original Message - 
  From: Skyler 
  To: asterisk-users@lists.digium.com 
  Sent: Wednesday, May 11, 2011 19:57
  Subject: [asterisk-users] concurrent call tracking


  Hi all,

   

   I would like to track/store concurrent call usage per user by day/week/month 
and get server totals by day/week/month. Google comes up with mostly info 
regarding concurrent call limits, though my goal is to calculate actual 
concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - 
realtime (no gui). Any suggestions / open-source / AGI on where to start 
looking into implementing something like this? 

   

  TIA,

   

  Skyler

   

   



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Re: [asterisk-users] concurrent call tracking

[Steve Edwards]

On Thu, 12 May 2011, Dovid Bender wrote:

What I do is when ever a call comes in I update a table in MySQL to 
active = (active +1). On hang up I do active = (active -1).


I have a cron that checks once a minute to see how many active and 
stores it along with epoch in db.


I then have a graph that shows channel usage. If you want the code let 
me know.


How do you handle table locking in case more than 1 call arrives 'at the 
same time?'


How do you handle a crash?

Using group() and group_count() with a bit of AMI to retrieve the count 
and stuff it in the database sounds like a 'better practice.'


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Re: [asterisk-users] concurrent call tracking

[Skyler]

Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd
like to take a look at it for sure. The dial plan example Leif replied with
is pretty much what I was thinking, just didn't have a clue how to go about
it. ;)

 

 Haven't figured out how I'm going to display the usage info either so if
you don't mind sharing the graph/code as well that would be sweet.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: Wednesday, May 11, 2011 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] concurrent call tracking

 

What I do is when ever a call comes in I update a table in MySQL to active =
(active +1). On hang up I do active = (active -1).

 

I have a cron that checks once a minute to see how many active and stores it
along with epoch in db.

 

I then have a graph that shows channel usage. If you want the code let me
know.

 

- Original Message - 

From: Skyler   

To: asterisk-users@lists.digium.com 

Sent: Wednesday, May 11, 2011 19:57

Subject: [asterisk-users] concurrent call tracking

 

Hi all,

 

 I would like to track/store concurrent call usage per user by
day/week/month and get server totals by day/week/month. Google comes up with
mostly info regarding concurrent call limits, though my goal is to calculate
actual concurrent channel usage and add it into reporting. I'm using * 1.6.2
+ mysql - realtime (no gui). Any suggestions / open-source / AGI on where to
start looking into implementing something like this? 

 

TIA,

 

Skyler

 

 


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No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1325 / Virus Database: 1500/3630 - Release Date: 05/11/11

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Re: [asterisk-users] About X100P and TDM400P analog card in China

[Gilles]

On Wed, 11 May 2011 01:09:16 +0800, Scott Zhang
 wrote:
>So does this mean no solution when used ZAP/DAHDI with PSTN line?
>
>If I installed an E1, will that work?

Before getting an E1, maybe ISDN provides call supervision?


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[asterisk-users] Higher CPU usage on 1.6.1 than 1.4?

[David Cunningham]

Hello,

We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now
experiencing higher CPU utilization on their server. I can't see anything
wrong, so is this just expected with 1.6? Can anyone help explain it?

Thanks for any advice.

-- 
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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Re: [asterisk-users] Asterisk 1.8.4 Now Available

[Jose P. Espinal]

Hello Folks,

Download links on the website have not been updated (asterisk.org)


Regards,


--
Jose P. Espinal
http://www.eslackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs

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[asterisk-users] Disabling echo cancellation by software

[Alejandro Cabrera Obed]

Dear, I have Asterisk 1.6 with an E1 Digium card with echo
cancellation module. So I need to use just the echo cancellation by
hardware and disable the echo cancellation by software. I use DAHDI
for my telephony hardware.

If the lines involved with echo cancel are:

In /etc/dahdi/system.conf:

echocanceller=mg2,1-15,17-31

In /etc/asterisk/chan_dahdi.conf:

echocancel=yes
echocancelwhenbridged=no
echotraining=800

What lines do I have to comment or to change the value if I want to
disable echo cancellation by software ???

Thanks a lot,

Alejandro

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Re: [asterisk-users] Trying out a new version with sangoma card

[Nicolas Ross]

Le 2011-05-09 09:31, Jim Dickenson a écrit :

Make sure the firmware on the card is latest. I had a problem, not like your, 
and flashing the card to the latest firmware resolved it.

It appears it did not change anything...

So, to re-cap, I have a sangoma A101 card, with the firmware uptodate, 
on the asterisk 1.8.3.3, dahdi-linux 2.4.1.2, and dahdi-tools 2.4.1.


When asterisk is running, cat /proc/dahdi/1 yields :

Span 1: WPT1/0 "wanpipe1 card 0" (MASTER) B8ZS/ESF RED

   1 WPT1/0/1 Clear (In use)
   2 WPT1/0/2 Clear (In use)
(...)
  24 WPT1/0/24 Hardware-assisted HDLC (In use)

And when it's not, the (In use) go away.

When, dialing I get "Unable to create channel of type 'DAHDI' (cause 34 
- Circuit/channel congestion)"


So, does anybody got any idea ?

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Re: [asterisk-users] Trying out a new version with sangoma card

[Jim Dickenson]

In asterisk CLI do "pri show spans". The fact the card is in RED alert means 
the hardware does not "see" the pri line connected to the card.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On May 11, 2011, at 6:55 PM, Nicolas Ross wrote:

> Le 2011-05-09 09:31, Jim Dickenson a écrit :
>> Make sure the firmware on the card is latest. I had a problem, not like 
>> your, and flashing the card to the latest firmware resolved it.
> It appears it did not change anything...
> 
> So, to re-cap, I have a sangoma A101 card, with the firmware uptodate, on the 
> asterisk 1.8.3.3, dahdi-linux 2.4.1.2, and dahdi-tools 2.4.1.
> 
> When asterisk is running, cat /proc/dahdi/1 yields :
> 
> Span 1: WPT1/0 "wanpipe1 card 0" (MASTER) B8ZS/ESF RED
> 
>   1 WPT1/0/1 Clear (In use)
>   2 WPT1/0/2 Clear (In use)
> (...)
>  24 WPT1/0/24 Hardware-assisted HDLC (In use)
> 
> And when it's not, the (In use) go away.
> 
> When, dialing I get "Unable to create channel of type 'DAHDI' (cause 34 - 
> Circuit/channel congestion)"
> 
> So, does anybody got any idea ?
> 
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Re: [asterisk-users] Disabling echo cancellation by software

[Shaun Ruffell]

On Wed, May 11, 2011 at 10:54:06PM -0300, Alejandro Cabrera Obed wrote:
> Dear, I have Asterisk 1.6 with an E1 Digium card with echo
> cancellation module. So I need to use just the echo cancellation by
> hardware and disable the echo cancellation by software. I use DAHDI
> for my telephony hardware.
> 
> If the lines involved with echo cancel are:
> 
> In /etc/dahdi/system.conf:
> 
> echocanceller=mg2,1-15,17-31
> 
> In /etc/asterisk/chan_dahdi.conf:
> 
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800
> 
> What lines do I have to comment or to change the value if I want to
> disable echo cancellation by software ???

With all current releases of DAHDI if a span has a hardware echocanceler
available, it will be used regardless of the settings in
/etc/dahdi/system.conf.

You can disable the hardware echocan in the wcte12xp and wct4xxp drivers by
setting the vpmsupport module parameter to 0 when you load the driver. You 
do this from the command line (for the wct4xxp driver in these examples) with
'modprobe wct4xxp vpmsupport=0' or set 'options wct4xxp vpmsupport=0' in
/etc/modprobe.d/dahdi.conf in order for it to take effect each time the driver
is loaded.

The v2.5.0 version of DAHDI, which is not yet released, will require that
hardware echocan be set in /etc/dahdi/system.conf for each channel. This will
allow a span to use a combination of software and hardware echo cancelers in
addition to providing a way to disable the hardware echo canceler without
reloading the drivers. Support for this feature is already committed to the
trunk of DAHDI Linux.

If you do check out the trunk, and want to use the hardware
echocan, be sure to set the echocanceller line in /etc/dahdi/system.conf to
'hwec' for each channel. dahdi_genconf does not yet setup the hwec by default
on the spans that have it available but it will before the 2.5.0 release.

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

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Re: [asterisk-users] Disabling echo cancellation by software

[Shaun Ruffell]

I just reread your original email, and I think I answered the wrong
question.

On Wed, May 11, 2011 at 11:36:04PM -0500, Shaun Ruffell wrote:
> On Wed, May 11, 2011 at 10:54:06PM -0300, Alejandro Cabrera Obed wrote:
> > Dear, I have Asterisk 1.6 with an E1 Digium card with echo
> > cancellation module. So I need to use just the echo cancellation by
> > hardware and disable the echo cancellation by software. I use DAHDI
> > for my telephony hardware.
...
> With all current releases of DAHDI if a span has a hardware echocanceler
> available, it will be used regardless of the settings in
> /etc/dahdi/system.conf.

The above paragraph is the answer to your original question. If you have
a hardware echo canceler module installed it will always be used in
preference to the software echo cancelers.

Although still again this will not be true in DAHDI Linux 2.5.0 when
it's released.  Hardware echocan will then need to be configured just
like the software echo cancelers.

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

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[asterisk-users] Different IP addresss for SIP and RTP

[mayamatakeshi]

Hello,
is it possible to set an IP address for RTP different than the one used for
SIP?
I want to use asterisk behind a sip proxy (opensips), but I was thinking if
I could avoid having to run rtpproxy on the sip proxy server and let
asterisk itself take care of it. So that:
  Asterisk SIP address : local ip address
  Asterisk RTP address : global ip address

regards,
takeshi
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Re: [asterisk-users] Asterisk 1.8.4 Now Available

[Vladimir Mikhelson]

Paul,

I have kind of a related question.

asterisk-1.8.4-summary.txt  does not always properly link specific
patches to issues. For example, revision 307509 is associated with issue
18542, and it is not reflected in the summary.  There may be more like this.

I tried to report this inconsistency timely, issue #18933, but it is
still listed as "new"

What is the right way of reporting documentation issues?

Thank you,
Vladimir



On 5/11/2011 9:46 AM, Paul Belanger wrote:
> On 11-05-11 10:29 AM, Jeremy Kister wrote:
>> I'm a bit confused about this release (and previous releases on the 1.8
>> track) so please bare with me.
>>
>> I viewed the ChangeLog, but I don't see any of the 'sample issues'
>> listed. why is that ? I would expect to see the 'sample issues' listed
>> after 1.8.4-rc3.
>>
>> Also, is there a reason/procedural error that patches such as:
>> https://issues.asterisk.org/view.php?id=18382
>> https://issues.asterisk.org/view.php?id=18742
>>
>> didnt make it into this 1.8.4 release ?
>>
>>
> Correct, they will appear in 1.8.5-rc1 forward. When -rc1 is created,
> it will be tagged from the HEAD of branches/1.8..  If 1.8.5-rc2 is
> create, it is because of an issue / bug was found in 1.8.5-rc1, and
> will include that fix only.
>
> If a new issue is reported after 1.8.5-rc1 and fixed in branches/1.8,
> it will not be added into 1.8.5 release, but will wait until 1.8.6-rc1.
>

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Re: [asterisk-users] With what options is asterisk compiled in rpm's

[Vladimir Mikhelson]

On 5/11/2011 10:15 AM, Danny Nicholas wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of isr...@gmail.com
>> Sent: Wednesday, May 11, 2011 10:07 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] With what options is asterisk compiled in rpm's
>>
>> Hi,
>>
>> I'm trying to add modules compiled from source into a rpm install of
>> asterisk (from digium) on centos and asterisk complains that its not
>> compiled with same options so it won't load it
>>
>> I know I could install the entire thing from source but for other reasons
>> I would like to keep the main things installed from rpm and install
>> whatever else I need from source (or roll my own rpm for those)
>>
>> Thanks,
>> Israel
> [Danny Nicholas] 
> This might work for you - download the source and do ./configure
> --libdir=/fromsrclib --bindir=/fromsrcbin
>
> Once you've done the make menuselect, make and make install, you should be
> able to copy the module(s) from /fromsrclib to /usr/lib/asterisk/modules 
>
>
>
Danny,

Can you please clarify.  Are the "/fromsrclib" and the "/fromsrcbin"
actual directories or keywords?  If these are directories should they be
created empty or populated with something?

Thank you,
Vladimir

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