Re: [asterisk-users] how to know status of asterisk from php

[virendra bhati]

Hi,

As per you suggestion I write small php scripts but didn't get result. Below
is the php script and output of programs too.

*PHP Script:-*

".$priline;
echo "";
echo "pri=>".$pri;
echo "";
echo "asterisk=>".$asterisk;
echo "";
echo "asterisks=>".$asterisks;
echo "";
echo "mysql=>".$mysql;
echo "";
echo "mysqls=>".$mysqls;
echo "";
?>

*Output:-*

Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
priline=>Unable to connect to remote asterisk (does /var/run/asterisk.ctl
exist?)
pri=>1
asterisk=>
asterisks=>127
mysql=>
mysqls=>127


On Wed, Apr 27, 2011 at 8:43 PM, Juan David Diaz wrote:

> Hi:
>
> http://php.net/manual/en/function.system.php
>
> Then, the commands you shoul run:
>
> /usr/sbin/asterisk -rnx"pri show spans"
> /etc/init.d/asterisk status
> /etc/init.d/mysql status
> .
> .
> .
> .
> and so on!!
>
> good luck!
>
> Regards.
>
> Juan.
> Linux User #441131
>
>
> On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati wrote:
>
>> Hi
>>
>> How to know status of Asterisk,Mysql. PRI lines and other services from
>> PHP scripts ?
>>
>> 
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9172341457
>>
>>
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Thanks and regards

 Virendra Bhati
+91-9172341457
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Kevin Keane]

> We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. 
> According to the release plans, support for 1.4 was scheduled to close in 
> April 2011 - basically now. After that, only security patches would be 
> committed. This is already a delay from the original plan published by 
> Russell Bryant.

> Unfortunately, I think this is way too early.

I don't have first-hand experience or an opinion on this matter, but just 
wanted to comment on how refreshingly welcome it is to have this discussion at 
all - without Open Source, we'd simply be stuck with a "Vista" type software 
(if I believe those who expressed concerns about 1.8).

I do have one question: what about the ecosystem? Many people don't use 
Asterisk by itself, but as part of distributions (PBX in a Flash, Trixbox, ...) 
and with tools such as FreePBX to configure it. How ready is the ecosystem for 
moving to a new Asterisk version?


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Michael L. Young]

- Original Message -
> From: "Olle E. Johansson" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Wednesday, April 27, 2011 3:34:03 PM
> Subject: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
>
> Friends,
>
> We have a discussion on asterisk-dev about the maintenance of the 1.4
> branch. According to the release plans, support for 1.4 was
> scheduled to close in April 2011 - basically now. After that, only
> security patches would be committed. This is already a delay from
> the original plan published by Russell Bryant.
>
> Unfortunately, I think this is way too early. My feeling and
> experience is that 1.8 is not ready for production in the
> environments I work in - large scale installations. Customers are
> not planning migration and all new installs are still 1.4. Tests
> we've been doing with 1.8 has failed within just a short time and so
> badly that customers has not paid me to spend any further time with
> 1.8.
>

Whats the game plan to get 1.8 "ready for production"?  To me, for which I say 
this with all respect, some are focusing still on 1.4 instead of getting 1.8 to 
the level that some of the members of the community are wanting to see.  1.4 
has been very stable for a while.  To the point that I only pay attention to 
security releases to be honest.  It has been this way for quite a while now.  I 
personally have been focused more on using 1.8 when I can, mainly on 
non-critical servers, yet I will admit that I have enough confidence in it now 
to use on main servers.  Why?  Because I want to get my production servers off 
of 1.4 and 1.6.2 due to new features.  But, even if I didn't need or want the 
new features, the current state of 1.4 is excellent.  If I don't ever make the 
move beyond 1.4, how can I contribute to a better product?  By experimenting 
and not giving up at the first sign of trouble with the latest version, I feel 
that I can help to make 1.8 better which ultimately benefits me and the 
community.  I would like to hear a game plan before we just say, yes, lets keep 
focusing on 1.4 and then we will decide a deadline to stop support.  I am 
afraid that software is programmed by imperfect humans and there will always be 
a bug or two that crops up from time to time.  Do we want to keep waiting until 
we feel it is "perfect"?

One thing I have noticed, is that the bug fixes and patches being contributed 
for 1.8 and trunk are not being taken care of as quickly as it used to back in 
the early 1.4 days.  My feelings are that it is because there have been too 
many releases to work on.  Going back to focusing on just 1.8 and trunk, would 
go a long way to speeding up bug fixes to 1.8.  Again, just my opinion.

> Last time we went through this process with a LTS release (which we
> did not know then) it took over one year before we had a stable
> product to migrate away from 1.2 and jump on the 1.4 track.
> Hopefully, with the help of community, we can move up to 1.8 late
> this year or early next year. For me 1.8 is the focus, it's the LTS
> release.
>
> Not having a supported 1.4 version from the Digium-hosted
> repositories will mean that we will have to move to separate
> repositories or branch off from the main track. I already maintain a
> ton of subversion branches with various patches to 1.4 It takes a
> lot of time to manage this version that is a fork from the main 1.4
> branch. I will soon have to start working with subversion branches
> for 1.8 to create a compatible version for my customers to test,
> since most of the patches is not part of 1.8. After a few years of
> doing this, I know the work involved with managing code myself.
>
> The Digium team wants to go ahead and not support 1.4 any more, I
> want to keep 1.4 open for normal bug fixes. What do you think?

Was this really Digium's decision?  You keep mentioning Digium and implying 
them as the evil one in all of this (perhaps I am just misunderstanding your 
tone in your emails and if I am, I sincerely apologize for this) when I seem to 
recall plenty of discussion around these time lines and it was the community 
who set the deadlines, not Digium.  Digium is just trying to abide by the time 
lines outlined for them by the community.  They have already been nice enough 
to extend the deadline in order to finish working on outstanding bug reports 
and patches.  They have bills to pay too and have really tried to extend an 
olive branch to everyone in the community.  There has been a lot of activity on 
the 1.4 branch lately.  If I am wrong, I will gladly retract my comments.

>
> Kevin proposed that the community maintains the 1.4 branch without
> support from the Digium team. I don't think that's a good solution,
> but it may be the only solution.  I haven't got the resources to
> manage the 1.4 code myself, so I won't step forward as a maintainer
> if I can't get proper funding. Anyone else out there that has the
> time and resources to manage the code

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Matteo Piazza]

I agree 100%, it's too early.
There is a lot of businnes out of there based on 1.4 (even still 1.2), 
and my feelings is that a lot of people is not going to upgrade the 
asterisk version, they are going to stay with 1.4 for a long time yet.


Also i wanna add another little consideration. Voip is not only a 
software matter, is a Telecomunication matter. And into the 
Telecomunication world the first priority is the reliability and 
reliability and reliability without forget that usually the lifetime of 
a telecomunicaton product is much more than 4 years.


I'm not a code writer so I can't put my effort in maintaince stuff.

I think 1.4 should be open at least for some critical bug like for 
example segmentation fault or memory leack.

Matteo


Il 27/04/2011 21:34, Olle E. Johansson ha scritto:

Friends,

We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. 
According to the release plans, support for 1.4 was scheduled to close in April 
2011 - basically now. After that, only security patches would be committed. 
This is already a delay from the original plan published by Russell Bryant.

Unfortunately, I think this is way too early. My feeling and experience is that 
1.8 is not ready for production in the environments I work in - large scale 
installations. Customers are not planning migration and all new installs are 
still 1.4. Tests we've been doing with 1.8 has failed within just a short time 
and so badly that customers has not paid me to spend any further time with 1.8.

Last time we went through this process with a LTS release (which we did not 
know then) it took over one year before we had a stable product to migrate away 
from 1.2 and jump on the 1.4 track. Hopefully, with the help of community, we 
can move up to 1.8 late this year or early next year. For me 1.8 is the focus, 
it's the LTS release.

Not having a supported 1.4 version from the Digium-hosted repositories will 
mean that we will have to move to separate repositories or branch off from the 
main track. I already maintain a ton of subversion branches with various 
patches to 1.4 It takes a lot of time to manage this version that is a fork 
from the main 1.4 branch. I will soon have to start working with subversion 
branches for 1.8 to create a compatible version for my customers to test, since 
most of the patches is not part of 1.8. After a few years of doing this, I know 
the work involved with managing code myself.

The Digium team wants to go ahead and not support 1.4 any more, I want to keep 
1.4 open for normal bug fixes. What do you think?

Kevin proposed that the community maintains the 1.4 branch without support from 
the Digium team. I don't think that's a good solution, but it may be the only 
solution.  I haven't got the resources to manage the 1.4 code myself, so I 
won't step forward as a maintainer if I can't get proper funding. Anyone else 
out there that has the time and resources to manage the code?

Feel free to send me mail off list if you have ideas or suggestions on how to 
solve this - or continue the discussion here.

Regards,
/Olle

PS. Please don't start a discussion about 1.8 quality in this thread, that's a 
separate issue. I just want to know what you think about closing 1.4 support 
now. If you want to discuss 1.8 quality, start a new thread. Thanks.
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Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?

[Anthony Messina]

On 04/27/2011 02:06 PM, satish patel wrote:
> Which echo cancellation is good between OSLEC and MG2. Dahdi by default use 
> MG2 echo cancellation on channel.  If i want to use OSLEC then what should i 
> need to do ? Do i need to recompile dahdi with OSLEC ?

Yes, you would need to compile the OSLEC kernel module.  Or, if you are
using a RedHat/Fedora based distro, you're welcome to use the
dahdi-linux and dahdi-linux-kmod RPMS I build here.  I include OSLEC
with the dahdi-linux-kmod build.

http://messinet.com/rpms/

-- 
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E



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Re: [asterisk-users] PAP2T auto answer?

[Mike Diehl]

On Wednesday 27 April 2011 2:13:07 am C F wrote:
> The answer function on an analog line is accomplished by going off
> hook. Unless the line is controlled by an automated device (like
> answering machine) someone has to physically take the device off hook
> to answer it. The ATA has no way to do it as all it gives is the FXS
> signalling.

You are quite right.  Thank you.

> What exactly are you trying to accomplish?

I've got this working on my Polycoms and was simply hoping to use the PAP as 
an inexpensive intercom.  Guess not. ;^)

> Vikingeleoctronics makes a door box (E20 iirc) that is powered by an
> analog line and can do auto answer when it gets the first ring.

I'll look into it.  Thank you.

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Gordon Henderson]

On Wed, 27 Apr 2011, Olle E. Johansson wrote:

The Digium team wants to go ahead and not support 1.4 any more, I want 
to keep 1.4 open for normal bug fixes. What do you think?


I would like to see continued bug and security fixes for 1.4 for some time 
yet.


As well as a raft of hosted servers, I have several hundred systems out 
there - many small embedded type installations (boot from flash, run in 
ram - my own custom Linux install) - a lot still running 1.2 which will 
never be upgraded until they die.


I didn't find 1.4 stable enough for my needs until about the late 
1.4.20's.


And really, I don't have a need for 1.4 - 1.2 did (and still does) 
everything I need to build a PBX capable of a few 100 extensions, but I 
felt that if I didn't move then things would get tough - no bug fixes, no 
support, and being laughed at for being such a dinosaur...


Right now, I've no plans at all to move to 1.8, nor the time at present to 
even download, compile and test it. There are no features in it that I 
need and no bugs (that affect me & my simple needs) that it fixes in 1.4 
that I'm aware of. So I don't feel the point. (The one thing it might have 
- BRI support in DAHDI is now moot as I've decided to abandon all on-board 
PSTN hardware and use external devices as it's much less of a hassle 
all-round and I want to sell SIP minutes!)


Gordon

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Administrator TOOTAI]

Le 27/04/2011 21:34, Olle E. Johansson a écrit :

Friends,

We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. 
According to the release plans, support for 1.4 was scheduled to close in April 
2011 - basically now. After that, only security patches would be committed. 
This is already a delay from the original plan published by Russell Bryant.

Unfortunately, I think this is way too early. My feeling and experience is that 
1.8 is not ready for production in the environments I work in - large scale 
installations. Customers are not planning migration and all new installs are 
still 1.4. Tests we've been doing with 1.8 has failed within just a short time 
and so badly that customers has not paid me to spend any further time with 1.8.

[...]

Agree with you at 100%. 1.8 is not ready for production. I remember our 
switch from 1.2 to 1.4  very early and had huge problems (misdn and 
B410P just comes in my mind), had to work with trunk, aso. At 1.4.8 or 
so it started to be stable. We're now at 1.8.3 ...


Also, latest 1.4 had some regressions (eg voicemailbox sequences), which 
means that we're not, at this time, sure that basic stuffs are working 
smoothly with 1.4.41 What happends if new regressions appears?


My vote goes to stay with 1.4 and continue to stabilize it (not asking 
to include new stuff) till community declare that 1.8 is at the level of 
1.4.


--
Daniel

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

> 
> I(me, my opinion, my feelings, my commercial view) am on the side of
> dropping support for 1.4 and 1.6.  1.8 had some major issues which are
> resolved/being worked on with more energy as older platforms are shut
> down. If a large enough security issue showed up, I hope we would all
> try to do the right thing and push it back to 1.6 and 1.4.
1.6.x is not an option for me at all. These' releases are not LTS. We can't 
upgrade as often as that release schedule required. I am very happy to see 
1.6.x disappear
in the darkness and from my hard disk drives.

> Support
> must end sometime. Merging changes across many versions is very
> difficult and time consuming.  
I fully agree here.

> Asterisk GUI is very limited do to its
> 1.4 support code.  There are users that still use 1.2 and are very
> happy.  They are not looking for new features. I hope the 1.4 / 1.6
> users can survive while they test the 1.8 branch and share why or why
> not it will fit their needs.

They will survive and they will merge their own bug fixes. I just wish we could 
share the work and maintain the branch in public instead of everyone managing 
it by their own. As long as 1.8 is not ready for the way we use it, we have no 
version to migrate to. 

I am sure that 1.8 will fit their needs and deliver a lot of extra. It's a cool 
new release. Everyone wants to go there. That's not the issue here. The issue 
is when it's ready for the larger installed base beyond the early adoptors.

I don't like the project I've been part of for many years not offering a 
supported option that fits the customers I work with. It's as simple as that. 

Saying that they should know better, that the project has posted the release 
plans for a long time warning about this - it  just doesn't cut it as long as 
we have no working code to replace the current version with. 

Compared with last time we had a painful migration (from 1.2 to 1.4) there are 
numerous other options out there.  I think the project have to be a bit more 
careful about our attitude towards the installed base. I want to keep them in 
the Asterisk project. That is where I belong and where they belong.

/O
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Andrew Latham]

On Wed, Apr 27, 2011 at 3:34 PM, Olle E. Johansson  wrote:
> Friends,
>
> We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. 
> According to the release plans, support for 1.4 was scheduled to close in 
> April 2011 - basically now. After that, only security patches would be 
> committed. This is already a delay from the original plan published by 
> Russell Bryant.
>
> Unfortunately, I think this is way too early. My feeling and experience is 
> that 1.8 is not ready for production in the environments I work in - large 
> scale installations. Customers are not planning migration and all new 
> installs are still 1.4. Tests we've been doing with 1.8 has failed within 
> just a short time and so badly that customers has not paid me to spend any 
> further time with 1.8.
>
> Last time we went through this process with a LTS release (which we did not 
> know then) it took over one year before we had a stable product to migrate 
> away from 1.2 and jump on the 1.4 track. Hopefully, with the help of 
> community, we can move up to 1.8 late this year or early next year. For me 
> 1.8 is the focus, it's the LTS release.
>
> Not having a supported 1.4 version from the Digium-hosted repositories will 
> mean that we will have to move to separate repositories or branch off from 
> the main track. I already maintain a ton of subversion branches with various 
> patches to 1.4 It takes a lot of time to manage this version that is a fork 
> from the main 1.4 branch. I will soon have to start working with subversion 
> branches for 1.8 to create a compatible version for my customers to test, 
> since most of the patches is not part of 1.8. After a few years of doing 
> this, I know the work involved with managing code myself.
>
> The Digium team wants to go ahead and not support 1.4 any more, I want to 
> keep 1.4 open for normal bug fixes. What do you think?
>
> Kevin proposed that the community maintains the 1.4 branch without support 
> from the Digium team. I don't think that's a good solution, but it may be the 
> only solution.  I haven't got the resources to manage the 1.4 code myself, so 
> I won't step forward as a maintainer if I can't get proper funding. Anyone 
> else out there that has the time and resources to manage the code?
>
> Feel free to send me mail off list if you have ideas or suggestions on how to 
> solve this - or continue the discussion here.
>
> Regards,
> /Olle
>
> PS. Please don't start a discussion about 1.8 quality in this thread, that's 
> a separate issue. I just want to know what you think about closing 1.4 
> support now. If you want to discuss 1.8 quality, start a new thread. Thanks.

Olle

I(me, my opinion, my feelings, my commercial view) am on the side of
dropping support for 1.4 and 1.6.  1.8 had some major issues which are
resolved/being worked on with more energy as older platforms are shut
down. If a large enough security issue showed up, I hope we would all
try to do the right thing and push it back to 1.6 and 1.4. Support
must end sometime. Merging changes across many versions is very
difficult and time consuming.  Asterisk GUI is very limited do to its
1.4 support code.  There are users that still use 1.2 and are very
happy.  They are not looking for new features. I hope the 1.4 / 1.6
users can survive while they test the 1.8 branch and share why or why
not it will fit their needs.

-- 
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Tim Nelson]

- Original Message -
> Friends,
> 
> SNIP
> 
> Unfortunately, I think this is way too early. My feeling and
> experience is that 1.8 is not ready for production in the environments
> I work in - large scale installations. Customers are not planning
> migration and all new installs are still 1.4. Tests we've been doing
> with 1.8 has failed within just a short time and so badly that
> customers has not paid me to spend any further time with 1.8.
> 
> SNIP
>

I've found the same issues. Asterisk 1.4.x is extremely stable and is running 
on all of our direct infrastructure as well as our customer owned 
infrastructure. Testing of the 1.8.x branch has shown quite a few problems and 
I'm definitely not putting it into production any time soon. It *IS* a good 
step forward, and I'm excited to see where it ends up, but I certainly don't 
find it to be a valid replacement for rock solid 1.4.x boxes right now.

I hope enough others echo simlar findings to allow a second look at a Digium 
maintained 1.4.x.

Unfortunately, we (personally, or company wise) cannot offer any development 
resources for a community maintained 1.4.x branch.

--Tim

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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Danny Nicholas]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Olle E. Johansson
> Sent: Wednesday, April 27, 2011 2:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
> 
> Friends,
> 
> We have a discussion on asterisk-dev about the maintenance of the 1.4
> branch. According to the release plans, support for 1.4 was scheduled to
> close in April 2011 - basically now. After that, only security patches
> would be committed. This is already a delay from the original plan
> published by Russell Bryant.
> 
> Unfortunately, I think this is way too early. My feeling and experience is
> that 1.8 is not ready for production in the environments I work in - large
> scale installations. Customers are not planning migration and all new
> installs are still 1.4. Tests we've been doing with 1.8 has failed within
> just a short time and so badly that customers has not paid me to spend any
> further time with 1.8.
> 

> 
> Not having a supported 1.4 version from the Digium-hosted repositories
> will mean that we will have to move to separate repositories or branch off
> from the main track. I already maintain a ton of subversion branches with
> various patches to 1.4 It takes a lot of time to manage this version that
> is a fork from the main 1.4 branch. I will soon have to start working with
> subversion branches for 1.8 to create a compatible version for my
> customers to test, since most of the patches is not part of 1.8. After a
> few years of doing this, I know the work involved with managing code
> myself.
> 
> The Digium team wants to go ahead and not support 1.4 any more, I want to
> keep 1.4 open for normal bug fixes. What do you think?
> 
> Kevin proposed that the community maintains the 1.4 branch without support
> from the Digium team. I don't think that's a good solution, but it may be
> the only solution.  I haven't got the resources to manage the 1.4 code
> myself, so I won't step forward as a maintainer if I can't get proper
> funding. Anyone else out there that has the time and resources to manage
> the code?
> 
> Feel free to send me mail off list if you have ideas or suggestions on how
> to solve this - or continue the discussion here.
> 
> Regards,
> /Olle
[Danny Nicholas] 
IMO, 1.4 should be kept open for bug fixes since it is the "current working
standard" - until 1.8 works in parallel function-for-function with 1.4, it
is NOT a production-ready release.  What good is a Ferrari with 3 tires?


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Re: [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)

[John Alexis]

Unfortunatelly that doesn't change anything. I got exactly the same error
("Everyone is busy/congested at this time (1:0/0/1)" ... ).
I did a "dialplan reload" before testing of course.


2011/4/25 Camilo Echeverry 

> As I see in your iax.conf, IAX Peer belogs to "special" context, which
> means 444 is allowed to make calls to extensions only on the same context
> (Extension 111), can you call extension 111 ?
> may be the other extensions are in the " default context" and you can
> receive calls because extension 444 (dial IAX2/444) exists in that conext.
>
> Try adding this in the [special] conext
>
> ;this will dial any 3 digit Extension to IAX
> exten => _XXX,1,Dial(IAX2/${EXTEN})
> exten => _XXX,n,hangup()
>
> that may solve your problem
>
> On Sat, Apr 23, 2011 at 3:38 AM, John Alexis wrote:
>
>> Hi,
>>
>> Sorry to insist, but I still not have any solution. Does anybody have an
>> idea ?
>> Thanks!
>>
>> 2011/4/20 John Alexis 
>>
>>> Hi,
>>>
>>> I have a problem with IAX accounts...
>>> I set up a few months ago an Asterisk server, with mysql support to load
>>> iax accounts.
>>> Settings seems fine because apparently the system works as expected.
>>> Yesterday I tried to add another iax account in the iax.conf directly.
>>> And I have a problem with this new account (named 444).
>>> I can authenticate from 444 to the server, and I can receive calls from
>>> other softphones (which parameters are loaded from the mysql database
>>> iaxfriends).
>>> BUT, i cannot call other softphones. I always got a message in the log
>>> saying "Everyone is busy/congested at this time (1:0/0/1)".
>>> So, i don't know where is the probleme : is it from iax accounts loaded
>>> from the database, or the new account 444 ???
>>>
>>> Below are the conf files and verbose output.
>>>
>>> Thank you very much for your help :)
>>>
>>>
>>> -
>>> - iax.conf
>>> -
>>>
>>> [general]
>>> bindport=4569
>>> delayreject=yes
>>> language=fr
>>> autokill = yes
>>> calltokenoptional = 0.0.0.0/0.0.0.0
>>> minregexpire = 60
>>> maxregexpire = 500
>>> mohsuggest=default
>>> careinvite=no
>>> rtcachefriends=yes
>>>
>>>
>>> [444]
>>> type=friend
>>> host=dynamic
>>> context=special
>>> secret=iop
>>>
>>> -
>>> - extconfig.conf:
>>> -
>>>
>>> [general]
>>>
>>> [settings]
>>> iaxusers => mysql,asterisk,iaxfriends
>>> iaxpeers => mysql,asterisk,iaxfriends
>>> voicemail => mysql,asterisk,voicemail
>>>
>>>
>>> -
>>> - Mysqldump from iaxfriends
>>> -
>>> INSERT INTO iaxfriends
>>> (name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar)
>>> VALUES 
>>> ('admin.my.domain','friend','100','admin@my.domain','123','','default','','dynamic','10.0.100.56','26564','','','','','0','','','','','','en','admin.my.domain','','','','','md5','','','','','','','all','gsm,ulaw,alaw','','','','','','','','','')
>>> ;
>>> INSERT INTO iaxfriends
>>> (name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar)
>>> VALUES ('alice.my.domain','friend','111','admin@my.domain
>>> ','','alice@my.domain','456','','default','','dynamic','10.0.100.221','42478','','','','','1303301760','','','','','','en','alice.my.domain','','','','','md5','','','','','','','all','gsm,ulaw,alaw','','','','','','','','','')
>>> ;
>>>
>>>
>>> -
>>> - extensions.conf:
>>> -
>>>
>>> [general]
>>>
>>> [externe]
>>> exten => 555,1,Dial(IAX2/111)
>>> exten => 555,n,Hangup()
>>>
>>>
>>> [special]
>>> exten => 111,1,Dial(IAX2/111)
>>> exten => 111,n,Hangup()
>>>
>>> [default]
>>>
>>> exten => 444,1,Dial(IAX2/444)
>>> exten => 444,n,Hangup()
>>>
>>>
>>>
>>>
>>> - Sip.conf (SIP server):
>>>
>>> [general]
>>> context=default
>>> allowoverlap=no
>>> udpbindaddr=0.0.0.0
>>> tcpenable=no
>>> tcpbindaddr=0.0.0.0
>>> srvlookup=yes
>>>
>>>
>>> -
>>> - Logs server:
>>> -
>>>
>>> -- Accepting AUTHENTICATED call from 10.0.100.238:
>>>> requested format = gsm,
>>>> requested prefs = (),
>>>> actual format = ulaw,
>>>> host prefs = (),
>>>> priority = mine
>>> -- Executing [111@special:1] Dial("IAX2/444-436", "IAX2/111") in new
>>> stack
>>>   == Everyone is busy/congested at this time (1:0/0/1)
>>> -- Executing [111@special:2] Hangup(

[asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

Friends,

We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. 
According to the release plans, support for 1.4 was scheduled to close in April 
2011 - basically now. After that, only security patches would be committed. 
This is already a delay from the original plan published by Russell Bryant.

Unfortunately, I think this is way too early. My feeling and experience is that 
1.8 is not ready for production in the environments I work in - large scale 
installations. Customers are not planning migration and all new installs are 
still 1.4. Tests we've been doing with 1.8 has failed within just a short time 
and so badly that customers has not paid me to spend any further time with 1.8.

Last time we went through this process with a LTS release (which we did not 
know then) it took over one year before we had a stable product to migrate away 
from 1.2 and jump on the 1.4 track. Hopefully, with the help of community, we 
can move up to 1.8 late this year or early next year. For me 1.8 is the focus, 
it's the LTS release.

Not having a supported 1.4 version from the Digium-hosted repositories will 
mean that we will have to move to separate repositories or branch off from the 
main track. I already maintain a ton of subversion branches with various 
patches to 1.4 It takes a lot of time to manage this version that is a fork 
from the main 1.4 branch. I will soon have to start working with subversion 
branches for 1.8 to create a compatible version for my customers to test, since 
most of the patches is not part of 1.8. After a few years of doing this, I know 
the work involved with managing code myself.

The Digium team wants to go ahead and not support 1.4 any more, I want to keep 
1.4 open for normal bug fixes. What do you think?

Kevin proposed that the community maintains the 1.4 branch without support from 
the Digium team. I don't think that's a good solution, but it may be the only 
solution.  I haven't got the resources to manage the 1.4 code myself, so I 
won't step forward as a maintainer if I can't get proper funding. Anyone else 
out there that has the time and resources to manage the code?

Feel free to send me mail off list if you have ideas or suggestions on how to 
solve this - or continue the discussion here.

Regards,
/Olle

PS. Please don't start a discussion about 1.8 quality in this thread, that's a 
separate issue. I just want to know what you think about closing 1.4 support 
now. If you want to discuss 1.8 quality, start a new thread. Thanks.
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[asterisk-users] Echocancellation OSLEC vs MG2 ?

[satish patel]

Hi All,

Which echo cancellation is good between OSLEC and MG2. Dahdi by default use MG2 
echo cancellation on channel.  If i want to use OSLEC then what should i need 
to do ? Do i need to recompile dahdi with OSLEC ?

-S
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Re: [asterisk-users] h323 with NAT

[Danny Nicholas]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Jose P. Espinal
> Sent: Wednesday, April 27, 2011 1:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] h323 with NAT
> 
> 
> > [Danny Nicholas]
> > Thanks for the information - but this doesn't seem to play well with
> SUSE.
> > Any ideas?
> 
> If you are open to the possibility of building from source I think I
> might have a little white paper based on the scripts (about installing
> latest version of H323plus on 1.4.X) by today, after I get home (like
> 7:30 pm, GMT -4); so you can test with native chan_h323.
> 
> Meanwhile, do you see anything weird (after enabling 'ooh323 debug') on
> the CLI?
> 
> Is there a possibility to test with SIP, to see if the audio problem is
> explicitly H323 related, and not a networking issue?
> 
> 
> --
> Jose P. Espinal
> http://www.eSlackware.com
> IRC: Khratos @ #asterisk / -doc / -bugs
[Danny Nicholas] 
Works like a champ with SIP - nothing I can see that is weird on CLI output
in H323


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Re: [asterisk-users] h323 with NAT

[Jose P. Espinal]

[Danny Nicholas] 
Thanks for the information - but this doesn't seem to play well with SUSE.

Any ideas?


If you are open to the possibility of building from source I think I 
might have a little white paper based on the scripts (about installing 
latest version of H323plus on 1.4.X) by today, after I get home (like 
7:30 pm, GMT -4); so you can test with native chan_h323.


Meanwhile, do you see anything weird (after enabling 'ooh323 debug') on 
the CLI?


Is there a possibility to test with SIP, to see if the audio problem is 
explicitly H323 related, and not a networking issue?



--
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http://www.eSlackware.com
IRC: Khratos @ #asterisk / -doc / -bugs


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Re: [asterisk-users] DHCP / DNS

[Eric Wieling]

No.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Sent: Wednesday, April 27, 2011 2:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DHCP / DNS

Are there any internal DHCP or DNS services built-in to the Asterisk code?


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Re: [asterisk-users] DHCP / DNS

[Andrew Latham]

No [1]

1. AsteriskNOW does have some of these services as do many
distributions like Zentyal.

On Wed, Apr 27, 2011 at 2:04 PM, Thomas Perron  wrote:
> Are there any internal DHCP or DNS services built-in to the Asterisk code?
>
> --
>
>
>
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[asterisk-users] DHCP / DNS

[Thomas Perron]

Are there any internal DHCP or DNS services built-in to the Asterisk code?


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Re: [asterisk-users] Asterisk, SIP & Firewalls

[Ryan Wagoner]

On Wed, Apr 27, 2011 at 1:16 PM, Myles Wakeham  wrote:
> It kinda scares me though.  I know that SIP is an attractive attack-vector,
> and that there are scripts out there that target SIP devices.  I know I
> could run Fail2Ban on the server, which is fine (we're doing that anyway
> now), but before I go down this path, I wanted to get general feedback if we
> are using our Asterisk system using 'best practices' or whether it should
> never be sitting behind a Firewall, despite the fact that it is working
> pretty close to perfect as it is right now.  I just want to find a way to
> reduce the latency.

I have placed Asterisk outside the firewall / nat router to avoid the
translation. I usually will setup the server with dual NICs. One has
the public IP and another has the internal private IP. Set the default
gateway to the public IP gateway. Then just configure iptables to
firewall the server interfaces accordingly. This configuration allows
Asterisk to sit directly on the Internet while keeping your internal
phones from going out your nat router and back to Asterisk. Basically
the best of both worlds.

Ryan

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Re: [asterisk-users] Asterisk, SIP & Firewalls

[Stelios Koroneos]

On Wed, 2011-04-27 at 10:16 -0700, Myles Wakeham wrote:
> Well there is one 'optimization' that I need to sort out.  There seems 
> to be some latency between the Asterisk server (and the SIP Phones) and 
> callers.  Depending on the caller's network (ie. POTS, Cell phone, other 
> Voip, etc.) we find about 30% of the time that there is a small delay 
> (about 1/2 a second) between us talking and the caller hearing it, which 
> makes it sound like the caller is talking to an offshore company located 
> in South Asia.  I have read numerous posts, discussions, etc. about this 
> sort of thing and it seems that it has something to do with our 
> Firewall, QoS, etc. and I'm entertaining moving the entire Asterisk 
> server outside of our Firewall, and connecting the SIP phones to it on 
> an entirely separate sub-net with a dedicated NAT router.
> 
1/2  second latency i dough it could be attributed to a firewall/qos,
unless your Internet connection is saturated with p2p or some other high
volume traffic (movie/radio streaming) or your firewall is running on
some slow machine with too many rules for packet inspection etc.
If that's the case moving asterisk to public ip wan't fix it.

As a first indication you could add a "qualify=yes" in all your sip
peers to see how long it takes them to "talk" to asterisk.



> It kinda scares me though.  I know that SIP is an attractive 
> attack-vector, and that there are scripts out there that target SIP 
> devices.  I know I could run Fail2Ban on the server, which is fine 
> (we're doing that anyway now), but before I go down this path, I wanted 
> to get general feedback if we are using our Asterisk system using 'best 
> practices' or whether it should never be sitting behind a Firewall, 
> despite the fact that it is working pretty close to perfect as it is 
> right now.  I just want to find a way to reduce the latency.
> 
> Does anyone have any thoughts about this?
> 

90% of the problems i see with asterisk security has to do with bad
configuration, bad dialplans and bad security policies (weak
passwords,no monitoring) etc.
The other 10% can be protocol or asterisk security issues but usually
these get fixed before script-kiddies get a chance to use them.

In your case since all your sip traffic would be coming from a single IP
address (of your provider) things are a bit easier to setup.

IMHO try to avoid as much as you can exposing asterisk to a public
ip/network and use it as a last resort method if everything else fails.


Stelios


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Re: [asterisk-users] asterisk practices

[David]

Hey,

You could test it? Setup a second server that calls the voicemail on the 
first server. Test different call volumes that way. Write a simple 
script that calls asterisk manager and originate your 50 calls 
asynchronously, see what happens.


If it's a production server, you may want to wait for closing hours.

David

On 2011-04-27 13:34, vip killa wrote:
I just completed building a feature rich asterisk voicemail system 
using perl, php, and mysql.
My only concern is that the system i built will not be able to handle 
the call volume needed. Let me start by explaining my setup.


Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> 
voicemailbox (using mysql odbc) or terminate with wrong number message


if a message is left in a voicemailbox the following happens:
externnotify -> notify.pl  (perl -> mysql lookup) -> 
up to 2 calls originated (using AMI), up to 4 emails sent out (with up 
to 2 attachemnts of voicemail)


this system may need to handle up to 50 concurrent calls. the 
"notify.pl " script may be called several times a 
second.
My question is, will asterisk be able to handle calling the "notify.pl 
" script that many times? or is there a better way 
to handle large volumes of voicemail notification, thank you in 
advance for your input.





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Re: [asterisk-users] h323 with NAT

[Danny Nicholas]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Jose P. Espinal
> Sent: Wednesday, April 27, 2011 11:04 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] h323 with NAT
> 
> 
> Danny Nicholas wrote:
> > Hi list,
> > I've been beating my head for about 3 days on this one.  I have
> > Asterisk 1.4.41 installed using openh323.  As long as I'm inside my
> > firewall, everything is hunky-dory.  When I move to server on another
> > subnet, I'm still able to connect, but no longer have sound.  Any good
> > pointers or suggestions?
> >
> > Thanks
> > Danny Nicholas
> 
> 
> I had a similar problem once while using ooh323 with Asterisk 1.4.XX.
> 
> What I did was to use the most recent version of H323plus with Asterisk
> and got better results with chan_h323.
> 
> As (AFAIK) OpenH323 was renamed to H323plus, and several improvements
> has been made to it, you might want to take a look at it.
> 
> Note: if you are building Asterisk from source, then the source expects
> a very old version of OpenH323 and PTLib.
> 
> You can take a look to the tasks performed by these scripts:
> http://lists.digium.com/pipermail/asterisk-users/2011-January/258119.html
> to see how to compile Asterisk with the latest version of H323Plus and
> PTlib.
> 
> If you need any additional information about the scripts, just let me
> know.
> 
> Regards,
> 
> --
> Jose P. Espinal
> http://www.eSlackware.com
> IRC: Khratos @ #asterisk / -doc / -bugs
[Danny Nicholas] 
Thanks for the information - but this doesn't seem to play well with SUSE.
Any ideas?


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[asterisk-users] asterisk practices

[vip killa]

I just completed building a feature rich asterisk voicemail system using
perl, php, and mysql.
My only concern is that the system i built will not be able to handle the
call volume needed. Let me start by explaining my setup.

Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> voicemailbox
(using mysql odbc) or terminate with wrong number message

if a message is left in a voicemailbox the following happens:
externnotify -> notify.pl (perl -> mysql lookup) -> up to 2 calls originated
(using AMI), up to 4 emails sent out (with up to 2 attachemnts of voicemail)

this system may need to handle up to 50 concurrent calls. the "notify.pl"
script may be called several times a second.
My question is, will asterisk be able to handle calling the "notify.pl"
script that many times? or is there a better way to handle large volumes of
voicemail notification, thank you in advance for your input.
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[asterisk-users] Asterisk, SIP & Firewalls

[Myles Wakeham]

Hi all,

I'm trying to get my head around our Asterisk network configuration. 
We've been using it for about 2 years now (home office) and it works 
great.  Its Asterisk 1.4.2 with SIP through external provider(s).


We have the Asterisk server behind our IPCop firewall, and have a 
dedicated IP address that comes to the firewall from our ISP (Cox) and 
that is routed to our Asterisk box using SIP ports, etc.  It works fine, 
connects without issue and we then have all of our SIP Phones throughout 
the house for the calls.  My wife & I run businesses from our home, so 
we have multiple numbers coming into Asterisk and with some fancy 
Asterisk scripting, etc. we have the one system acting as a phone system 
for 4 companies.  Works great.


Well there is one 'optimization' that I need to sort out.  There seems 
to be some latency between the Asterisk server (and the SIP Phones) and 
callers.  Depending on the caller's network (ie. POTS, Cell phone, other 
Voip, etc.) we find about 30% of the time that there is a small delay 
(about 1/2 a second) between us talking and the caller hearing it, which 
makes it sound like the caller is talking to an offshore company located 
in South Asia.  I have read numerous posts, discussions, etc. about this 
sort of thing and it seems that it has something to do with our 
Firewall, QoS, etc. and I'm entertaining moving the entire Asterisk 
server outside of our Firewall, and connecting the SIP phones to it on 
an entirely separate sub-net with a dedicated NAT router.


It kinda scares me though.  I know that SIP is an attractive 
attack-vector, and that there are scripts out there that target SIP 
devices.  I know I could run Fail2Ban on the server, which is fine 
(we're doing that anyway now), but before I go down this path, I wanted 
to get general feedback if we are using our Asterisk system using 'best 
practices' or whether it should never be sitting behind a Firewall, 
despite the fact that it is working pretty close to perfect as it is 
right now.  I just want to find a way to reduce the latency.


Does anyone have any thoughts about this?

Thanks in advance for any comments or suggestions.

Myles
--
-
Myles Wakeham
Director of Engineering
Tech Solutions USA LLC
www.techsolusa.com
Phone +1-480-451-7440


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Re: [asterisk-users] h323 with NAT

[Jose P. Espinal]

Danny Nicholas wrote:

Hi list,
I've been beating my head for about 3 days on this one.  I have
Asterisk 1.4.41 installed using openh323.  As long as I'm inside my
firewall, everything is hunky-dory.  When I move to server on another
subnet, I'm still able to connect, but no longer have sound.  Any good
pointers or suggestions?

Thanks
Danny Nicholas



I had a similar problem once while using ooh323 with Asterisk 1.4.XX.

What I did was to use the most recent version of H323plus with Asterisk 
and got better results with chan_h323.


As (AFAIK) OpenH323 was renamed to H323plus, and several improvements 
has been made to it, you might want to take a look at it.


Note: if you are building Asterisk from source, then the source expects 
a very old version of OpenH323 and PTLib.


You can take a look to the tasks performed by these scripts:
http://lists.digium.com/pipermail/asterisk-users/2011-January/258119.html
to see how to compile Asterisk with the latest version of H323Plus and 
PTlib.


If you need any additional information about the scripts, just let me know.

Regards,

--
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http://www.eSlackware.com
IRC: Khratos @ #asterisk / -doc / -bugs

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[asterisk-users] h323 with NAT

[Danny Nicholas]

Hi list,
I've been beating my head for about 3 days on this one.  I have
Asterisk 1.4.41 installed using openh323.  As long as I'm inside my
firewall, everything is hunky-dory.  When I move to server on another
subnet, I'm still able to connect, but no longer have sound.  Any good
pointers or suggestions?

Thanks
Danny Nicholas


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Re: [asterisk-users] how to know status of asterisk from php

[nik600]

Hi, you can use the PHPAgi project

http://phpagi.sourceforge.net/

Otherwise, if you want a more high-level approach you can use the MXML
interface, you will communicate with HTTP GET request and obtaing XML
response directly from Asterisk.

Enabling the http manager interface you will get enabled some manager
commands on the port 8088

Ie, you can Login with:

http://your-asterisk-ip:8088/mxml?action=login&username=$this->user&secret=$this->pass

Some example commands:

http://your-asterisk-ip:8088/mxml?action=queuestatus
http://your-asterisk-ip:8088/mxml?action=SipPeers
http://your-asterisk-ip:8088/mxml?action=status
http://your-asterisk-ip:8088/mxml?action=DBput&family=$family&key=$key&Val=$val
http://your-asterisk-ip:8088/mxml?action=QueueAdd&queue=$queue&interface=$interface
http://your-asterisk-ip:8088/mxml?action=QueueRemove&queue=$queue&interface=$interface
http://your-asterisk-ip:8088/mxml?action=QueuePause&queue=$queue&interface=$interface&Paused=1
http://your-asterisk-ip:8088/mxml?action=QueuePause&queue=$queue&interface=$interface&Paused=0

And so on


On Wed, Apr 27, 2011 at 1:22 PM, virendra bhati  wrote:
> Hi
>
> How to know status of Asterisk,Mysql. PRI lines and other services from PHP
> scripts ?
>
> 
> Thanks and regards
>
>  Virendra Bhati
> +91-9172341457
>
>
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>



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http://www.kumbe.it

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Re: [asterisk-users] how to know status of asterisk from php

[Juan David Diaz]

Hi:

http://php.net/manual/en/function.system.php

Then, the commands you shoul run:

/usr/sbin/asterisk -rnx"pri show spans"
/etc/init.d/asterisk status
/etc/init.d/mysql status
.
.
.
.
and so on!!

good luck!

Regards.

Juan.
Linux User #441131


On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati  wrote:

> Hi
>
> How to know status of Asterisk,Mysql. PRI lines and other services from PHP
> scripts ?
>
> 
> Thanks and regards
>
>  Virendra Bhati
> +91-9172341457
>
>
> --
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>
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Re: [asterisk-users] Digium WCTDM24XXP DTMF CallerID

[Shaun Ruffell]

On Wed, Apr 27, 2011 at 09:26:24AM -0300, Antonio Modesto wrote:
> Good morning,
> 
> I have a digium wctdm24xxp in my asterisk box, i am not able to see
> the callerid when the call is incoming from the fxo line, i live in
> Brazil, how can i change the signaling from fsk to dtmf?

Hello Antonio,

In chan_dahdi.conf you would need to set cidsignalling=dtmf and cidstart=dtmf.

However there may currently be an issue at least with Asterisk 1.6.2 where the
start of the caller ID is not detected.  I would be interested to hear what
sort of results you have when you try to set this.

Thanks,
Shaun

-- 
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] AGI WAIT FOR DIGIT - key press BEFORE command

[David]

Hi,

Consider the following situation :

AGI Rx << WAIT FOR DIGIT 3000
AGI Tx >> 200 result=48
AGI Rx << WAIT FOR DIGIT 3000
AGI Tx >> 200 result=48
AGI Rx << WAIT FOR DIGIT 3000
AGI Tx >> 200 result=48
AGI Rx << WAIT FOR DIGIT 3000

What happens if the user enters a digit between the "200 result=" and 
the next "WAIT FOR DIGIT"? Will the next WAIT FOR DIGIT catch the digit? 
Is the digit lost? How can I insure I don't lose the digit ? I am 
calling one digit at a time because I want to validate the user's entry 
at each key press.


Thanks,

David


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[asterisk-users] Digium WCTDM24XXP DTMF CallerID

[Antonio Modesto]

Good morning,

I have a digium wctdm24xxp in my asterisk box, i am not able to see
the callerid when the call is incoming from the fxo line, i live in
Brazil, how can i change the signaling from fsk to dtmf?

Thanks.
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Re: [asterisk-users] Orginate not working well with PSTN lines

[Eric Wieling]

When dialing is finished on an analog FXO Asterisk considers it answered.  The 
solution is to use something that is not an analog FXO like PRI or SIP to a 
carrier.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashik Ali
Sent: Wednesday, April 27, 2011 7:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Orginate not working well with PSTN lines

Thanks for your solution.

Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco  ? .

Does anybody successed; to make asterisk to detect ring back tone from
PSTN telco ?

Thanks,
Ashik

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[asterisk-users] Konference module issue

[virendra bhati]

HI,

I have installed asterisk 1.6.2.18 with konference 1.7, All things are
working fine but when we start taking DTMF then key 3 not get my asterisk.
When we use landline number(dedicated number) than all DTMF is capture and
asterisk work fine. In case of mobile only key 3 don't work. Strange when I
use my touch screen number then most of the DTMF digits don't get's my
asterisk


But the same module work with asterisk 1.4.41 without any issue.

So is it asterisk issue or konference module issue please guide me.
Or all problem is created my mobile divices ?

Thanks in advance .
-- 



-
Thanks and regards

 Virendra Bhati
+91-9172341457
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[asterisk-users] how to know status of asterisk from php

[virendra bhati]

Hi

How to know status of Asterisk,Mysql. PRI lines and other services from PHP
scripts ?


Thanks and regards

 Virendra Bhati
+91-9172341457
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Re: [asterisk-users] Orginate not working well with PSTN lines

[Ashik Ali]

Thanks for your solution.

Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco  ? .

Does anybody successed; to make asterisk to detect ring back tone from
PSTN telco ?

Thanks,
Ashik

On Wed, Apr 27, 2011 at 12:44 PM, Gilles  wrote:
> On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali
>  wrote:
>>The problem here is that as soon as asterisk dialing on fxo lines it
>>sets channel status as "answered"  although the chennel is getting
>>ring back tone from
>>other party.
>>
>>Anyone can suggest me to solve this issue ?
>
> The only solution I know is to have Asterisk play a message in a loop
> for eg. 1mn, prompting the callee to hit a key to let the server know
> that the call was 1) answered 2) by a human being.
>
>
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Re: [asterisk-users] Orginate not working well with PSTN lines

[Gilles]

On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali
 wrote:
>The problem here is that as soon as asterisk dialing on fxo lines it
>sets channel status as "answered"  although the chennel is getting
>ring back tone from
>other party.
>
>Anyone can suggest me to solve this issue ?

The only solution I know is to have Asterisk play a message in a loop
for eg. 1mn, prompting the callee to hit a key to let the server know
that the call was 1) answered 2) by a human being.


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Re: [asterisk-users] Orginate not working well with PSTN lines

[Ashik Ali]

Dear all,

The problem here is that as soon as asterisk dialing on fxo lines it
sets channel status as "answered"  although the chennel is getting
ring back tone from
other party.

Anyone can suggest me to solve this issue ?

Thanks ,
Ashik

On Tue, Apr 26, 2011 at 4:28 PM, Jim Dickenson  wrote:
> "Originate successfully queued" only means that the originate action was 
> handed off to asterisk not that is was executed yet. There are other events, 
> depending on which events you are "reading", that tell you the call was 
> answered and such.
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com
>
> CfMC
> http://www.cfmc.com/
>
>
>
> On Apr 26, 2011, at 2:43 AM, Ashik Ali wrote:
>
>> Dear all,
>>
>> I am from Saudi  Arabiya and I am using digium hardware with asterisk 1.6.
>>
>> When I am executing following AMI originate API. Orginate start to
>> execute extenstion without knowing of PSTN(FXO) channel is ringing.
>>
>> Any one can help me to  resolve this issue ?
>>
>> Action: Originate
>> Channel: Dahdi/g0/2923878
>> Context: outbound-ivr
>> Exten: 1234
>> Priority: 1
>> ActionID: ABC45678901234567890
>>
>>
>> Response: Success
>> ActionID: ABC45678901234567890
>> Message: Originate successfully queued
>>
>>
>>  -- Remote UNIX connection disconnected
>>> Channel DAHDI/1-1 was answered.
>>    -- Executing [1234@outbound-ivr:1] SayDigits("DAHDI/1-1", "1234")
>> in new stack
>>    --  Playing 'digits/1.gsm' (language 'en')
>>    --  Playing 'digits/2.gsm' (language 'en')
>>    --  Playing 'digits/3.gsm' (language 'en')
>>    --  Playing 'digits/4.gsm' (language 'en')
>>    -- Executing [1234@outbound-ivr:2] Playback("DAHDI/1-1",
>> "demo-congrats") in new stack
>>    --  Playing 'demo-congrats.gsm' (language 'en')
>>    -- Executing [1234@outbound-ivr:3] Hangup("DAHDI/1-1", "") in new stack
>>  == Spawn extension (outbound-ivr, 1234, 3) exited non-zero on 'DAHDI/1-1'
>>    -- Hungup 'DAHDI/1-1'
>>
>>
>> Thanks & Regards,
>> Ashik
>>
>> --
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Re: [asterisk-users] RTP keepalive doesn't work

[Alok]

Kevin P. Fleming  digium.com> writes:

> Yes, it was lost during a merge of code into Asterisk trunk after 1.6.2 
> was branched (so only 1.8.0 and trunk are missing the code). Leif Madsen 
> entered an issue on Mantis as a blocker for any more 1.8.x releases 
> until this is resolved, as it is clearly a regression in the 1.8.x series.
> 


So is this working in which release 1.6.2.18 ?



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Re: [asterisk-users] PAP2T auto answer?

[C F]

The answer function on an analog line is accomplished by going off
hook. Unless the line is controlled by an automated device (like
answering machine) someone has to physically take the device off hook
to answer it. The ATA has no way to do it as all it gives is the FXS
signalling.
What exactly are you trying to accomplish?
Vikingeleoctronics makes a door box (E20 iirc) that is powered by an
analog line and can do auto answer when it gets the first ring.

On 4/25/11, Mike Diehl  wrote:
> Hi all,
>
> Is it possible to send a SIP header to a PAP2T or SPA and cause the
> device
> to automatically answer?  I can do this with my Polycom phones and would
> like
> to do it with my ATA's.
>
> Any ideas?
>
> --
>
> Take care and have fun,
> Mike Diehl.
>
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