date:20100106

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

2010-01-06 Thread Doug

At 00:22 1/7/2010, Tzafrir Cohen wrote:
 >On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote:
 >> At 16:49 1/5/2010, Tzafrir Cohen wrote:
 >>  >On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
 >>  >> Hi,
 >>  >>
 >>  >> Having problems with getting either RxFax or FaxReceive
 >>  >> to compile.  Running Asterisk 1.4 on CentOS 5.
 >>  >
 >>  >What version of SpanDSP do you use?
 >>
 >>spandsp-0.0.6pre12.tgz
 >>
 >> and:
 >>
 >>libtiff-3.8.2-7.el5_3.4
 >>libtiff-devel-3.8.2-7.el5_3.4
 >>
 >> Which do you recommend?
 >
 >What errors do you get? I'm using a backport of app_fax.c and it works
 >well.

Do you have the link for the "C" source?




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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread DHAVAL INDRODIYA

hi arun can you paste a dialplan here

and version of asterisk

regards
dhaval

On Thu, Jan 7, 2010 at 11:51 AM, Tzafrir Cohen wrote:

> On Thu, Jan 07, 2010 at 09:54:09AM +0530, Arun Sasidhar wrote:
> > hi,
> >
> >I made changes in zapata.conf but no result.
>
> You use zapata.conf . I suppose you use asterisk 1.4 . Give asterisk
> 1.6.0 or newer a shot.
>
> --
>Tzafrir Cohen
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> jabber:tzafrir.co...@xorcom.com
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> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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Re: [asterisk-users] compile one additional module without recompiling all asterisk

2010-01-06 Thread Tzafrir Cohen

On Thu, Jan 07, 2010 at 08:08:07AM +0200, Giedrius Augys wrote:
> Hello,
> 
>   Maybe there is the easiest way to compile additional my module without
> recompiling all asterisk?

'make' if you already have a fully-built source tree.
Optionally manual copying instead of 'make install'

If you want to build it out-of-tree, it will take more tweaking, but
doable .

-- 
   Tzafrir Cohen
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Re: [asterisk-users] Faxing: Anyone have a compiled executable?

2010-01-06 Thread Tzafrir Cohen

On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote:
> At 16:49 1/5/2010, Tzafrir Cohen wrote:
>  >On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
>  >> Hi,
>  >>
>  >> Having problems with getting either RxFax or FaxReceive
>  >> to compile.  Running Asterisk 1.4 on CentOS 5.
>  >
>  >What version of SpanDSP do you use?
> 
>spandsp-0.0.6pre12.tgz
> 
> and:
> 
>libtiff-3.8.2-7.el5_3.4
>libtiff-devel-3.8.2-7.el5_3.4
> 
> Which do you recommend?

What errors do you get? I'm using a backport of app_fax.c and it works
well.

-- 
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+972-50-7952406   mailto:tzafrir.co...@xorcom.com
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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Tzafrir Cohen

On Thu, Jan 07, 2010 at 09:54:09AM +0530, Arun Sasidhar wrote:
> hi,
> 
>I made changes in zapata.conf but no result.

You use zapata.conf . I suppose you use asterisk 1.4 . Give asterisk
1.6.0 or newer a shot.

-- 
   Tzafrir Cohen
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Re: [asterisk-users] Faxing: Anyone have a compiled executable?

2010-01-06 Thread Juan E. Rodríguez

I have it running on * 1.4.19. You can get it on the internet, .so and an 
intaller that checks for dependencies.

--Mensaje original--
De: Doug
Remitente: asterisk-users-boun...@lists.digium.com
Para: asterisk-users@lists.digium.com
Responder a: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Faxing: Anyone have a compiled executable?
Enviado: 7 Ene, 2010 01:41

At 16:49 1/5/2010, Tzafrir Cohen wrote:
 >On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
 >> Hi,
 >>
 >> Having problems with getting either RxFax or FaxReceive
 >> to compile.  Running Asterisk 1.4 on CentOS 5.
 >
 >What version of SpanDSP do you use?

   spandsp-0.0.6pre12.tgz

and:

   libtiff-3.8.2-7.el5_3.4
   libtiff-devel-3.8.2-7.el5_3.4

Which do you recommend?


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Saludos,
Juan E. Rodríguez
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[asterisk-users] compile one additional module without recompiling all asterisk

2010-01-06 Thread Giedrius Augys

Hello,

  Maybe there is the easiest way to compile additional my module without
recompiling all asterisk?
Thanks


-- 
Pagarbiai  / Best Regards,
Giedrius
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Re: [asterisk-users] Faxing: Anyone have a compiled executable?

2010-01-06 Thread Doug

At 16:49 1/5/2010, Tzafrir Cohen wrote:
 >On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
 >> Hi,
 >>
 >> Having problems with getting either RxFax or FaxReceive
 >> to compile.  Running Asterisk 1.4 on CentOS 5.
 >
 >What version of SpanDSP do you use?

   spandsp-0.0.6pre12.tgz

and:

   libtiff-3.8.2-7.el5_3.4
   libtiff-devel-3.8.2-7.el5_3.4

Which do you recommend?


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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Arun Sasidhar

hi,

   I made changes in zapata.conf but no result.
I tried different settings. I am getting differnt logs But no result

when i use "cidstart=ring"
I am getting this in my asterisk log

[Jan 7 09:31:13] VERBOSE[7129] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 7 09:31:14] ERROR[7129] callerid.c: No start bit found in fsk data.
[Jan 7 09:31:14] WARNING[7129] chan_dahdi.c: CallerID feed failed: Success
[Jan 7 09:31:14] WARNING[7129] chan_dahdi.c: CallerID returned with error on
channel 'DAHDI/1-1'
[Jan 7 09:31:14] VERBOSE[7129] logger.c: -- Executing [...@from-pstn:1]
Wait("DAHDI/1-1", "5") in new stack
[Jan 7 09:31:19] VERBOSE[7129] logger.c: -- Executing [...@from-pstn:2]
Gosub("DAHDI/1-1", "app-blacklist-check|s|1") in new stack
[Jan 7 09:31:19] VERBOSE[7129] logger.c: -- Executing 
[...@app-blacklist-check:1]
LookupBlacklist("DAHDI/1-1", "") in new stack
[Jan 7 09:31:19] WARNING[7129] app_lookupblacklist.c: LookupBlacklist is
deprecated. Please use ${BLACKLIST()} instead.
[Jan 7 09:31:19] VERBOSE[7129] logger.c: -- Executing 
[...@app-blacklist-check:2]
GotoIf("DAHDI/1-1", "0?blacklisted") in new stack

And when i use "cidstart=polarity" I am getting this in my log

[Jan 7 09:35:16] VERBOSE[7300] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 7 09:35:16] VERBOSE[7300] logger.c: -- Executing [...@from-pstn:1]
Wait("DAHDI/1-1", "5") in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing [...@from-pstn:2]
Gosub("DAHDI/1-1", "app-blacklist-check|s|1") in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing 
[...@app-blacklist-check:1]
LookupBlacklist("DAHDI/1-1", "") in new stack
[Jan 7 09:35:22] WARNING[7300] app_lookupblacklist.c: LookupBlacklist is
deprecated. Please use ${BLACKLIST()} instead.
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing 
[...@app-blacklist-check:2]
GotoIf("DAHDI/1-1", "0?blacklisted") in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing 
[...@app-blacklist-check:3]
Set("DAHDI/1-1", "CALLED_BLACKLIST=1") in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing 
[...@app-blacklist-check:4]
Return("DAHDI/1-1", "") in new stack
[Jan 7 09:35:22] VERBOSE[7300] logger.c: -- Executing [...@from-pstn:3]
ExecIf("DAHDI/1-1", "1 |Set|CALLERID(name)=") in ne

And I tried another option "cidstart=polarity_IN"

Then my log shows this..

Jan 7 09:43:13] VERBOSE[7642] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 7 09:43:14] NOTICE[7642] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan 7 09:43:15] NOTICE[7642] chan_dahdi.c: Got event 2 (Ring/Answered)...
[Jan 7 09:43:17] NOTICE[7642] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan 7 09:43:17] VERBOSE[7642] logger.c: -- Executing [...@from-pstn:1]
Wait("DAHDI/1-1", "5") in new stack
[Jan 7 09:43:22] VERBOSE[7642] logger.c: -- Executing [...@from-pstn:2]
Gosub("DAHDI/1-1", "app-blacklist-check|s|1") in new stack
[Jan 7 09:43:22] VERBOSE[7642] logger.c: -- Executing 
[...@app-blacklist-check:1]
LookupBlacklist("DAHDI/1-1", "") in new stack
[Jan 7 09:43:22] WARNING[7642] app_lookupblacklist.c: LookupBlacklist is
deprecated. Please use ${BLACKLIST()} instead.
[Jan 7 09:43:22] VERBOSE[7642] logger.c: -- Executing 
[...@app-blacklist-check:2]
GotoIf("DAHDI/1-1", "0?blacklisted") in new stack

In all these settings I dont have any CID output.

Thanks,
Arun S




On Wed, Jan 6, 2010 at 11:16 PM, Tzafrir Cohen wrote:

> On Wed, Jan 06, 2010 at 08:30:48PM +0530, Arun Sasidhar wrote:
> > Hi,
> >
> >Its a free service here and My ordinary phone displaying the Caller ID
> > without any problem.
> > I have done some modifications in zapata.conf
> > Now it looks like this
> >
> > *[channels]
> > language=en
> > hanguponpolarityswitch=yes
> > answeronpolarityswitch=yes
> > busydetect=yes
> > busycount=6
> > callprogress=yes
>
> If hanguponpolarityswitch works for you, you don't need the hacks of
> busydetect and callprogress .
>
> > callerid=asreceived
> > hidecallerid=no
> > immediate=no
> > cidsignalling=dtmf
> > cidstart=polarity_IN
>
> This is not supported in Asterisk 1.4 . Only as of 1.6.0, IIRC.
>
> Generally the fixes from that bug report were not applied to Asterisk
> 1.4 .
>
> > cid_rxgain=6
> > useincomingcalleridonzaptransfer=yes
>
> Renamed to "useincomingcalleridondahditransfer" (s/zap/dahdi/).
>
> > rxgain=5.0
> > txgain=2.0
> > ;cidsignalling=bell
> > ;cidsignalling=v23
> > ;cidstart=polarity
> > ;cidstart=ring
> > ; include dahdi extensions defined in FreePBX
> > #include chan_dahdi_additional.conf
> >
> > ; XTDM20B Port #1,2 plugged into PSTN
> > ;AMPLABEL:Channel %c - Button %n
> > context=from-pstn
> > signalling=fxs_ks
> > faxdetect=incoming
> > usecallerid=yes
> > echocancel=yes
> > echocancelwhenbridged=no
> > echotraining=800
> > group=0
> > channel=1-2*
>
> What is this '*'?
>
> --
>Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.co

[asterisk-users] Question about PLC of Asterisk

2010-01-06 Thread nakaji

Hi,I want to know how to do to work PLC of Asterisk.
Anyone plz help me.

PLC (Packet Loss Concealment) is included in Asterisk,I read at voip-info.org 
or release note.
And I see in codecs.conf, "genelicplc" setting.

So I put codecs.conf in '/etc/asterisk' ,and wrote "genericplc => true".
And I worked Asterisk and tested.

I think PLC like this. 
When I send 3 packet (1,2,3), and caused loss (if No.2 is lossed),
then receive-packet after PLC should be same number 3 packet(1,2,3).
No.2 is new one. I don't know 2 is same sound to old 2,or is zero level sound.
But same number packets should be resend from Asterisk to receiver.

But when I send 3 packet, received-packet is 2 packet(1,3).
No.2 has lossed.
No interpolation has been done.

So I can't understand how to do to work PLC of Asterisk.

Please help me.

I tested Asterisk 1.4.* and 1.6.0.
Use uLaw to send and receive on SIP/RTP.
And caused 5-20% loss from send-packet and through Asterisk ,I checked 
receive-packet.
 
Is there another things to do without "to put codecs.conf in /etc/asterisk" ?
Or Asterisk has no PLC on uLaw ?








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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-06 Thread Allann Jones

>
> I think that's very wise advice.  To offer a commercial perspective, our
> customers willing to pay for sophisticated
> smart phone apps (currently gov/mil agencies and some mid-size telecoms)
> have very specific needs and care about reliable operation, development
> duration, long-term support -- all the things you would expect in a normal
> project.

Those aren't going to happen on a jailbreak system.  For example they're
> willing to use Droid and not iPhone if the
> job can get done.
>

Firstly... Linux rules! :) ... and Asterisk too (and the latter is the
"on-topic", we must remain this...).

The jailbreak system gives you the chance to transfer a native iPhone
application to the phone without the need to register to be a Apple
developer and put the application on Apple Store to, and in the following,
put on the phone to finally execute it without programming so long on a
emulator.

Droid (Linux) is great too. And it is better because it is open source, and
Asterisk too! :)

But these devices already exists in use on the world, like many other
devices. What can be done with it? Put it on the trash and build many others
without reusing nothing? Another technological trash?

And for many people that will get only a smartphone or some another hardware
in a life for many reasons, and $$$ is the major factor. Some people will
never get one (maybe they will find happiness).

Droid generally runs on ARM, like iPhone, like many others.

A perfect hardware (that doesn't exist) runs buggy applications. A perfect
operating system (that doesn't exist too) runs buggy applications too. A
perfect developer (that doesn't exist too) can develop buggy applications
too (a paradox here!) because there are many perspectives in a infinite set
of perspectives that is not considered during a lifetime.

There are many purposes in the world, not only critical, commercial or
military purposes.

Will you put on trash a old router that is capable of run Linux or FreeBSD,
and buy a newer, ... or will you try to use it and learn something
(distributed processing)? It is the same for WinModems that, maybe, exists
in a greater number than smartphones in the world :)

But I know... Linux rules! :) and we see it growing (fast and faster and
faster!).. and Asterisk too.

On telefony you can get a bug too, like a timeout when programming a
operation to be executed during some ms, but for some reason the operating
system doesn't respond because it's executing another thread, or a
electrical failure, or the hardware simply doesn't respond in correct time
(a insect running on it?). All developer spend some hours on headache
engineering.

iPhone runs Objective-C, like GNUStep, NextStep, and the beatiful and
incredible WindowMaker :) that is supported by GNU GCC. so it has a good
environment to reuse to learn about a programming language, about software
engineering, about telefony, or about etc... (like many other devices) ...
to finally... in some day... develop a critical application (if really
capable) or create a better (or not) platform (reusing something (a idea, a
soft or hard)?).

There are many people that had and have fun on something like:

- http://palm-linux.sourceforge.net
- http://penguinppc.org
- http://www.arm.linux.org.uk
- http://linux-sh.org
- http://linuxoniphone.blogspot.com
- http://cydia.saurik.com
- http://www.kernel.org ;)
- http://www.asterisk.org :)

... between many others... and the list will always grows in anyway making
the technological world more interesting and the ozone decreases, and the
world clearer, etc, etc and etc ;)


Regards, apologies and thank you.


-- 
___
Allann J.

"I received the fundamentals of my education in school, but that was not
enough. My real education, the superstructure, the details, the true
architecture, I got out of the public library. For an impoverished child
whose family could not afford to buy books, the library was the open door to
wonder and achievement, and I can never be sufficiently grateful that I had
the wit to charge through that door and make the most of it." (from I.
Asimov, 1994)
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Re: [asterisk-users] How to see STDERR message?

2010-01-06 Thread Steve Edwards

On Thu, 7 Jan 2010, Zhang Shukun wrote:

> i use agi to send message back to Asterisk by STDERR, but why i could't 
> see the message in asterisk CLI?

Output to STDERR does nothing for me either.

I prefer to use syslog() to log the messages via syslogd.

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

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[asterisk-users] How to see STDERR message?

2010-01-06 Thread Zhang Shukun

hi,

i use agi to send message back to Asterisk by STDERR, but why i
could't see the message in asterisk CLI?

i start asterisk use " asterisk -vc" in order to see all message.

Thanks

-- 
Best regards,
Sucan

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Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P

2010-01-06 Thread David Backeberg

On Wed, Jan 6, 2010 at 6:23 PM, Olivier  wrote:
> The second time I'm dialing an internal extension attached to the same
> ReceiveFAX application :
>
> 2.   sendfax/hylafax/iaxmodem > asterisk > spandsp
>
> In the 2nd case, I've got 3 craches out of 3 attempts (with a rough estimee,
> the crash occurs 2 or 4s after ReceiveFAX's start).
> Before wasting anybody's time and effort within Asterisk support team, I
> would like to double check here if the case that crashes Asterisk is within
> specifications of involved apps.
> In other words, can you normally use Hylafax to send faxes to inner
> extensions or do you hace to stick to PSTN numbers ?

I've never successfully done what you're trying to do, so I came to
the conclusion that it was not supported. When I wanted to test
faxing, I ended up using two systems, sometimes with just LAN
in-between and sometimes with PSTN in-between.

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[asterisk-users] video with x-lite

2010-01-06 Thread Jerry Geis

I download the x-lite software for windows.
Put it on two laptops.
Using asterisk 1.4.28
set videosupport=yes in sip.conf general.

disallow=all
allow=h264
allow=ulaw
allow=alaw
videosupport=yes

have the above for both softphone defintions. I heard audio but no video 
when I call.
I click the "start" video on x-lite and it says video cannot be started.

Is there something else I am missing?

Jerry

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[asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P

2010-01-06 Thread Olivier

Hi,

I'm sending twice the same file using Hylafax's sendfax app.
The first time I'm "dialing" a DID attached to the ReceiveFAX application.
The second time I'm dialing an internal extension attached to the same
ReceiveFAX application :

1.   sendfax/hylafax/iaxmodem > asterisk > dadhi > dahdi >
asterisk > spandsp
2.   sendfax/hylafax/iaxmodem > asterisk > spandsp

In the 1st case, I've got 3 successes out of 3 attempts.
In the 2nd case, I've got 3 craches out of 3 attempts (with a rough estimee,
the crash occurs 2 or 4s after ReceiveFAX's start).

To complete description, I should say :
- I'm using a single asterisk instance
- dahdi to dahdi communication is done by a simple patch cord going from one
port to another port of a unique Digium B410P,
- everything (Hylafax, iaxmodem, asterisk) is installed on the same Lenny
server.

Before wasting anybody's time and effort within Asterisk support team, I
would like to double check here if the case that crashes Asterisk is within
specifications of involved apps.
In other words, can you normally use Hylafax to send faxes to inner
extensions or do you hace to stick to PSTN numbers ?

Regards
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[asterisk-users] Urgent: Which spandsp version is recommended for 1.6.1 ?

2010-01-06 Thread Olivier

Hi,

I need to install (within the next couple of hours) a 1.6.1.11 server with a
Digium B410P board.
One of this system's DID is dedicated to Fax reception (I don't need to send
faxes).
As I'm a bit familiar with it, I would like to use spandsp and ReceiveFAX
application to enable this feature.

For various reasons, I can't really test this configuration.

Could you suggest a Spandsp version you're already using ? 0.0.5 ?
0.0.6pre12 ? other ? another 1.6.1.X/spandsp combination ?

Regards
.
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[asterisk-users] DEVICE STATE "In use"

2010-01-06 Thread Tiago Geada

Hi

We have an operator that his device state on all queues is "In use" where it
should be "Not in use".

how can we manually change the state of a device?

I looked into the devstate function and tryed the following:

perfpbxr*CLI> devstate list
perfpbxr*CLI>
-
--- Custom Device States 
-
---
--- Name: 'Custom:notinuse'  State: 'NOT_INUSE'
---
-
-

then tried:

[default]

exten => ,hint,Custom:notinuse

So when any body dials  would change the devstate back to NOT_INUSE

doesn't seem to work.

How can we set the devstate?
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Re: [asterisk-users] question on makefile

2010-01-06 Thread Kevin P. Fleming

Jerry Geis wrote:

> is there no method by the configure command to --disable-FEATURE???

There is not. The Asterisk configure script is used for platform
specific settings, locating libraries and header files and the like. It
is not used (directly) for controlling which portions of Asterisk are
built or are not built.

> the help says its there but doesnt seem to do anything for me.
> 
> example: ./configure --disable-codec_lpc10
> 
> doesnt seem to do anything. I was trying to find a way without running 
> "make  menuselect" to not compile in certain items.

What help says it is there? The help text that documents how
'--disable-FEATURE' works assumes there is a list of FEATUREs
somewhere... but Asterisk does not have such a list since we don't
control features from the configure script.

You will have to use menuselect to disable modules from being built,
although it can be automated so that you don't have to use the
interactive mode to do it. Many Linux distributions (and also
AsteriskNOW) do this already in their package building recipes, so you
might want to take a look at one of them to see how it is done.

In addition, if you are scripting this build, you could just as easily
'rm codecs/codec_lpc10.c' before running the 'make' and 'make install'
steps.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
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Re: [asterisk-users] question on makefile

2010-01-06 Thread Jerry Geis

>
> It's a Makefile command.  See:
> http://www.gnu.org/software/automake/manual/make/Text-Functions.html#index-filter-554
>
>   
great - thanks

is there no method by the configure command to --disable-FEATURE???
the help says its there but doesnt seem to do anything for me.

example: ./configure --disable-codec_lpc10

doesnt seem to do anything. I was trying to find a way without running 
"make  menuselect" to not compile in certain items.

Thanks,

jerry

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Re: [asterisk-users] question on makefile

2010-01-06 Thread Tilghman Lesher

On Wednesday 06 January 2010 13:45:55 Jerry Geis wrote:
> There is a line like in codes/Makefile
>
> $(if $(filter
> codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10)
>
> What is filter? Where is filter?
>
> "whereis filter" doesnt return anything
> "find . | grep filter" in asterisk root directory returns nothing.

It's a Makefile command.  See:
http://www.gnu.org/software/automake/manual/make/Text-Functions.html#index-filter-554

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
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[asterisk-users] question on makefile

2010-01-06 Thread Jerry Geis

There is a line like in codes/Makefile

$(if $(filter 
codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10)

What is filter? Where is filter?

"whereis filter" doesnt return anything
"find . | grep filter" in asterisk root directory returns nothing.

Thanks,

Jerry

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Re: [asterisk-users] Merlin Legend integration not routing calls back to PSTN.

2010-01-06 Thread Shane Brath

I fixed it, the problem was that the 2nd T1 didn't have a Switch Identifyer
set.
I set the Switch Identifyer and now I can route calls to the PSTN. Merlin
has a default that a trunk with a Null Switch Identifyer  is considered a CO
trunk. So the Merlin was getting confused, and routing it to the unknown
extension because it didn't want to create a routing loop

I sourced 2 documents that lead me to this conslusion, and after much
research. So I post this for the good of Humanity!

*
http://marketingtools.avaya.com/knowledgebase/ipoffice/mergedProjects/bulletins/techtips/134_techtip.htm

* http://www.tek-tips.com/viewthread.cfm?qid=1486737&page=9


Now my only issue is that the Merlin is stripping the CID from the Tandem
connection, and only sending the area code to the PSTN.. Any sugestions
:)


On Tue, Jan 5, 2010 at 10:00 PM, C F  wrote:

> On Tue, Jan 5, 2010 at 9:40 PM, Shane Brath  wrote:
> >
> > Folks,
> >
> > I have a Merlin Legend R7 V10.0 with a 2 100D cards.
>
> Sorry, I feel your pain.
>
> >
> > I have 1 card in slot 4 going to CenturyTel, and the card in slot 10
> going
> > to a flip cable to a TE110P card in a Asterisk 1.6.x box.
> >
> > I have routing setup on the Merlin to send a block of numbers to the
> > Asterisk.
> >
> > Currently the PSTN can dial the Asterisk Extensions.
> > The Asterisk can dial Merlin Extensions.
> > The Merlin can Dial Asterisk extensions.
> >
> > But the Asterisk can't dial out to the PSTN?
> >
> > I have tried everything, and I'm hoping someone else can shed some light
> on
> > this. I'm open to ideas.
> > I've already removed the barrier codes, and disable access code
> requirements
> > on Tie and Non-Tie lines, with no effect.
> > I made sure that the Asterisk is dialing 9XXX when sending the call
> over
> > the DAHDI trunk to the Merlin.
> >
> > Whenever you call from the Asterisk to the Merlin you are redirected to
> the
> > "Unassigned Extension" extension, and dropped to the Operator. I have a
> > suspicion that this might have something to do with the NetwkService on
> the
> > Slot 4 100D card ( out to PSTN ).
> >
> > Here are some relavant files for comment:
> >
> > Merlin PRIINFO:
> > A PRI INFORMATION
> >
> >
> >
> > A Slot 4 Switch: 5ESS
> >
> > A Slot 10 Switch: Legend-Ntwk
> >
> > A System: By line
> >
> > A BchnlGrp #: Slot: TestTelNum: NtwkServ: Incoming Routing:
> > A 1 4 CallbyCall By Dial Plan
> >
> > A Channel ID: 23 22 21 20 19 18 17 16 15 14
> > A 13 12 11 10 9 8 7 6 5 4
> > A 3 2 1
> >
> > A Line PhoneNumber NumberToSend
> > A 801 NPANXX
> > A 802 NPANXX
> > A 803 NPANXX
> > A 804 NPANXX
> > A 805 NPANXX
> > A 806 NPANXX
> > A 807 NPANXX
> > A 808 NPANXX
> > A 809 NPANXX
> > A 810 NPANXX
> > A 811 NPANXX
> > A 812 NPANXX
> > A 813 NPANXX
> > A 814 NPANXX
> > A 815 NPANXX
> > A 816 NPANXX
> > A 817 NPANXX
> > A 818 NPANXX
> > A 819 NPANXX
> > A 820 NPANXX
> > A 821 NPANXX
> > A 822 NPANXX
> > A 823 NPANXX
> >
> > A BchnlGrp #: Slot: TestTelNum: NtwkServ: Incoming Routing:
> > A 80 10 ElecTandNtwk Route Directly to UDP
> >
> > A Channel ID: 23 22 21 20 19 18 17 16 15 14
> > A 13 12 11 10 9 8 7 6 5 4
> > A 3 2 1
> >
> > A Line PhoneNumber NumberToSend
> > A 829 NPANXX
> > A 830 NPANXX
> > A 831 NPANXX
> > A 832 NPANXX
> > A 833 NPANXX
> > A 834 NPANXX
> > A 835 NPANXX
> > A 836 NPANXX
> > A PRI INFORMATION
> >
> >
> > A 837 NPANXX
> > A 838 NPANXX
> > A 839 NPANXX
> > A 840 NPANXX
> > A 841 NPANXX
> > A 842 NPANXX
> > A 843 NPANXX
> > A 844 NPANXX
> > A 845 NPANXX
> > A 846 NPANXX
> > A 847 NPANXX
> > A 848 NPANXX
> > A 849 NPANXX
> > A 850 NPANXX
> > A 851 NPANXX
> >
> > A Network Selection Table
> >
> > A Entry Number: 0 1 2 3
> > A Pattern to Match: 101 10***
> >
> > A Special Service Table
> >
> > A Entry Number: 0 1 2 3 4 5 6 7
> > A Pattern to Match: 011 010 01 00 1
> > A Operator: none OP OP OP/P OP none none none
> > A Type of Number: I I I N N N N N
> > A Digits to Delete: 3 0 0 0 0 0 0 0
> >
> > A Call-By-Call Service Table
> >
> > A Entry Number: 0 1 2 3 4
> > A Pattern 0: 0
> > A Pattern 1: 1
> > A Pattern 2: 2
> > A Pattern 3: 3
> > A Pattern 4: 4
> > A Pattern 5: 5
> > A Pattern 6: 6
> > A Pattern 7: 7
> > A Pattern 8: 8
> > A Pattern 9: 9
> > A Call Type: BOTH BOTH BOTH BOTH BOTH
> > A NtwkServ: No Service
> > A DeleteDigits: 0 0 0 0 0
> >
> > A Entry Number: 5 6 7 8 9
> > A Call Type: BOTH BOTH BOTH BOTH BOTH
> > A NtwkServ:
> > A DeleteDigits: 0 0 0 0 0
> >
> > A Dial Plan Routing Table
> >
> > A Entry Number: 0 1 2 3
> > A NtwkServ: Any service Any service
> > A PRI INFORMATION
> >
> >
> > A Expected Digits: 4 4 0 0
> > A Pattern to Match: 
> > A Digits to Delete: 0 4 0 0
> > A Digits to Add: 
> >
> > A Entry Number: 4 5 6 7
> > A NtwkServ:
> > A Expected Digits: 0 0 0 0
> > A Pattern to Match:
> > A Digits to Delete: 0 0 0

Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-06 Thread UIT DEVELOPMENT

Cool - thanks!

On Wed, Jan 6, 2010 at 12:14 PM, Ira  wrote:
> At 05:31 PM 1/5/2010, you wrote:
>>Ah, good idea.  :-)   Are you saying that if I got a number that was
>>in my parents area code then they could be making a "local" call to my
>>Asterisk, which is physically a 1000+ miles from them?   Now that is
>>cool.
>
>
> Once you have Asterisk set up you can essentially get local numbers
> anywhere, Some countries are harder than others but almost any area
> code in the US for $1.50/month + 1.4 cents/minute or less is
> possible.  The free one I pointed out will likely give you one in a
> useless place to you, but it's free, number and minutes, perfect for
> calling home from a cell.  Also, the cheapest ATT cell phone will let
> you call each other unlimited for free.
>
> Ira
>
>
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[asterisk-users] Asterisk 1.6.1.x SMDI MWI w/Fujitsu F9600 Problem

2010-01-06 Thread Will Szopko

We have recently pulled an ancient Fujitsu-branded Centigram voicemail
system out of production use and replaced it with an Asterisk box, which
is now serving as our enterprise voicemail system and automated attendant.
The Asterisk system is connected to a Fujitsu F9600 PBX and uses the
1.6.1.x SMDI module to communicate between the two systems.

Save for one issue, the replacement of the old system has been a
resounding success. The issue is that the integration of the MWI (message
waiting indicator) functionality works, but only about 50% of the time.

The problem is easy to see, but hard to track. Sometimes when a message is
left on the voicemail system, the MWI will fail to show. Similarly,
sometimes after all messages have been reviewed, a lit MWI will fail to
turn off. Thus, it's not a problem specific to either the OP or RMV
parameter in the MWI message. Both work and fail with about the same
frequency.

I have put some logging commands into the SMDI code and thus have been
able to verify that Asterisk itself appears to be sending the proper codes
at the proper time, with 100% accuracy, best I can tell. As such, I don't
believe that this is an Asterisk-specific problem. However, I am wondering
if there could be an issue of interplay between the two systems.

If any of you are using an arrangement such as that described above and
have solved this issue, I would appreciate hearing the solution.
Similarly, if any of you have experience with SMDI and MWI and know of
potential glitches that I have not taken into account, I would appreciate
hearing about that as well.

Thanks for any insights any of you can provide.

- Will


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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Tzafrir Cohen

On Wed, Jan 06, 2010 at 08:30:48PM +0530, Arun Sasidhar wrote:
> Hi,
> 
>Its a free service here and My ordinary phone displaying the Caller ID
> without any problem.
> I have done some modifications in zapata.conf
> Now it looks like this
> 
> *[channels]
> language=en
> hanguponpolarityswitch=yes
> answeronpolarityswitch=yes
> busydetect=yes
> busycount=6
> callprogress=yes

If hanguponpolarityswitch works for you, you don't need the hacks of
busydetect and callprogress .

> callerid=asreceived
> hidecallerid=no
> immediate=no
> cidsignalling=dtmf
> cidstart=polarity_IN

This is not supported in Asterisk 1.4 . Only as of 1.6.0, IIRC.

Generally the fixes from that bug report were not applied to Asterisk
1.4 .

> cid_rxgain=6
> useincomingcalleridonzaptransfer=yes

Renamed to "useincomingcalleridondahditransfer" (s/zap/dahdi/).

> rxgain=5.0
> txgain=2.0
> ;cidsignalling=bell
> ;cidsignalling=v23
> ;cidstart=polarity
> ;cidstart=ring
> ; include dahdi extensions defined in FreePBX
> #include chan_dahdi_additional.conf
> 
> ; XTDM20B Port #1,2 plugged into PSTN
> ;AMPLABEL:Channel %c - Button %n
> context=from-pstn
> signalling=fxs_ks
> faxdetect=incoming
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800
> group=0
> channel=1-2*

What is this '*'?

-- 
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http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Ira

At 07:00 AM 1/6/2010, you wrote:
>   Its a free service here and My ordinary phone displaying the 
> Caller ID without any problem.
>I have done some modifications in zapata.conf
>Now it looks like this

Make sure that there is between 1 and 5 seconds after the first ring 
before you answer the call. On my box I started with wait(2) and 
later changed it to wait(1). Without the wait() as the first line of 
my dialplan for analog lines I don't get caller id.

Ira 

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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-06 Thread Ira

At 05:31 PM 1/5/2010, you wrote:
>Ah, good idea.  :-)   Are you saying that if I got a number that was
>in my parents area code then they could be making a "local" call to my
>Asterisk, which is physically a 1000+ miles from them?   Now that is
>cool.

Once you have Asterisk set up you can essentially get local numbers 
anywhere, Some countries are harder than others but almost any area 
code in the US for $1.50/month + 1.4 cents/minute or less is 
possible.  The free one I pointed out will likely give you one in a 
useless place to you, but it's free, number and minutes, perfect for 
calling home from a cell.  Also, the cheapest ATT cell phone will let 
you call each other unlimited for free.

Ira

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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-06 Thread Jeff Brower

Allann-

> On Wed, Jan 6, 2010 at 8:50 AM, Allann Jones  wrote:
>> But jailbreaking increases the freedom to develop a application and
>
> Oh, I agree with you, but it's probably even better to make a decision
> to either buy into the constraints of Apple or find a better, free-er
> phone, which is what I hope a lot of people will be doing in the next
> few years. In a vain attempt to return to VoIP and Asterisk, let's
> hope that more future mobile OS will all allow multi-apps so that you
> can leave a SIP client running in the background.
>
> The big problem with jailbreaking is updates. If you have a lot of
> time and energy  to manage that problem when it comes up, jailbreaking
> is "fun" in a geeky sort of way.

I think that's very wise advice.  To offer a commercial perspective, our 
customers willing to pay for sophisticated
smart phone apps (currently gov/mil agencies and some mid-size telecoms) have 
very specific needs and care about
reliable operation, development duration, long-term support -- all the things 
you would expect in a normal project. 
Those aren't going to happen on a jailbreak system.  For example they're 
willing to use Droid and not iPhone if the
job can get done.

-Jeff

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Re: [asterisk-users] Skype for Asterisk

2010-01-06 Thread Yawar Hadi

Dear,
there is a problem in codec translation..so change the ulaw codec to
g729. .if problem persist then u must have same codex on asterisk server and
clients (skype)...

On Mon, Jan 4, 2010 at 11:24 AM, Tim Panton  wrote:

>
> On 30 Dec 2009, at 19:43, vijay.go...@alliance-infotech.com wrote:
>
>
> Hi Sir,
>
> We have integrated Skype with Asterisk (skype user id:- rexesbposolutions).
> Each call which is coming to skype account is getting transfered to Asterisk
> Queue. It has following two cases:
>
> case 1: When we call from normal skype account to skype account
> (rexesbposolutions), everything is working fine.
>
> case 2: This skype account (rexesbposolutions) has been assigned with a
> online virtual number (00 44 20  ). If somebody dial this number
> from their landline/cellphone, call is transfered to Asterisk queue but it
> shows some problem related to G729 codecs. following are Asterisk CLI log:
>
> Executing [...@skypeincoming:1]
> Answer("Skype/rexesbposolutions-084159e8", "") in new stack
> -- Executing [...@skypeincoming:2]
> Wait("Skype/rexesbposolutions-084159e8", "5") in new stack
> -- Executing [...@skypeincoming:3]
> GotoIfTime("Skype/rexesbposolutions-084159e8",
> "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack
> -- Goto (sky,s,1)
> -- Executing [...@sky:1] Playback("Skype/rexesbposolutions-084159e8",
> "enter") in new stack
> --  Playing 'enter' (language 'en')
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x4 (ulaw)
> [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to
> restore format back to 4
> -- Executing [...@sky:2] Queue("Skype/rexesbposolutions-084159e8",
> "markq|t|||900") in new stack
> -- Started music on hold, class 'default', on
> Skype/rexesbposolutions-084159e8
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x40 (slin)
> [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:251 ast_moh_files_next:
> Unable to open file '/var/lib/asterisk/moh/manolo_camp-morning_coffee': No
> such file or directory
> -- Stopped music on hold on Skype/rexesbposolutions-084159e8
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release:
> Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2'
> -- Playing periodic announcement
> [Dec 29 17:27:34] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x2 (gsm)
> --  Playing 'queue' (language 'en')
> [Dec 29 17:27:43] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to
> restore format back to 2
>   == Spawn extension (sky, s, 2) exited non-zero on
> 'Skype/rexesbposolutions-084159e8'
> [Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call
>
>
>
> following are output of some commands:-
>
> *CLI> core show translation
>
>   Translation times between formats (in milliseconds) for one second of
> data
>   Source Format (Rows) Destination Format (Columns)
>
>   g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726
> g722
>  g723-   ---- -- -- --
> --
>   gsm-   -222 21 26 --
> 2-
>  ulaw-   2-12 21 26 --
> 2-
>  alaw-   21-2 21 26 --
> 2-
> g726aal2-   222- 21 26 --
> 2-
> adpcm-   2222 -1 26 --
> 2-
>  slin-   1111 1- 15 --
> 1-
> lpc10-   2222 21 -6 --
> 2-
>  g729-   6666 65 6- --
> 6-
> speex-   ---- -- -- --
> --
>  ilbc-   ---- -- -- --
> --
>  g726-   2222 21 26 --
> --
>  g722-   ---- -- -- --
> --
>
>
> *CLI> help g729
>  g729 show hostid  Show G.729 Host-ID
>g729 show licenses  Show G.729 Licenses and Usage
> g729 show version  Show G.729 Module Version
>
> *CLI> g729 show hostid
> Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be
>
> *CLI> g729 show licenses
> 0/0 encoders/decoders of 1 licensed channels are currently in use
>
> Licenses Found:
> File: ***-*.lic -- Key:  ***-* --

Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Arun Sasidhar

Hi,

   Its a free service here and My ordinary phone displaying the Caller ID
without any problem.
I have done some modifications in zapata.conf
Now it looks like this

*[channels]
language=en
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
busydetect=yes
busycount=6
callprogress=yes
callerid=asreceived
hidecallerid=no
immediate=no
cidsignalling=dtmf
cidstart=polarity_IN
cid_rxgain=6
useincomingcalleridonzaptransfer=yes
rxgain=5.0
txgain=2.0
;cidsignalling=bell
;cidsignalling=v23
;cidstart=polarity
;cidstart=ring
; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf

; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c - Button %n
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-2*

Then Now my log showing
*
[Jan  6 20:22:57] DEBUG[2886] dsp.c: dsp busy pattern set to 0,0
[Jan  6 20:22:57] VERBOSE[2926] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan  6 20:22:58] NOTICE[2926] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan  6 20:22:59] NOTICE[2926] chan_dahdi.c: Got event 2 (Ring/Answered)...
[Jan  6 20:23:01] NOTICE[2926] chan_dahdi.c: Got event 18 (Ring Begin)...
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:1]
Set("DAHDI/1-1", "__FROM_DID=s") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:2]
Gosub("DAHDI/1-1", "app-blacklist-check|s|1") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@app-blacklist-check:1] LookupBlacklist("DAHDI/1-1", "") in new stack
[Jan  6 20:23:01] WARNING[2926] app_lookupblacklist.c: LookupBlacklist is
deprecated.  Please use ${BLACKLIST()} instead.
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@app-blacklist-check:2] GotoIf("DAHDI/1-1", "0?blacklisted") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@app-blacklist-check:3] Set("DAHDI/1-1", "CALLED_BLACKLIST=1") in new
stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@app-blacklist-check:4] Return("DAHDI/1-1", "") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:3]
ExecIf("DAHDI/1-1", "1 |Set|CALLERID(name)=") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:4]
Set("DAHDI/1-1", "FAX_RX=disabled") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:5]
Set("DAHDI/1-1", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:6]
SetCallerPres("DAHDI/1-1", "allowed_not_screened") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing [...@from-pstn:7]
Goto("DAHDI/1-1", "from-did-direct|104|1") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Goto
(from-did-direct,104,1)
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@from-did-direct:1] Macro("DAHDI/1-1", "exten-vm|104|104") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing 
[...@macro-exten-vm:1]
Macro("DAHDI/1-1", "user-callerid") in new stack
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@macro-user-callerid:1] Set("DAHDI/1-1", "AMPUSER=") in new stack
[Jan  6 20:23:01] DEBUG[2926] app_macro.c: Executed application: Set
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@macro-user-callerid:2] GotoIf("DAHDI/1-1", "0?report") in new stack
[Jan  6 20:23:01] DEBUG[2926] app_macro.c: Executed application: GotoIf
[Jan  6 20:23:01] VERBOSE[2926] logger.c: -- Executing
[...@macro-user-callerid:3] ExecIf("DAHDI/1-1", "1|Set|REALCALLERIDNUM=") in
new stack
[Jan  6 20:23:01] DEBUG[2926] app_macro.c: Executed application: ExecIf
[Jan  6 20:23:01] DEBUG[2926] app_macro.c: Last app: Set|REALCALLERIDNUM=
[Jan  6 20:23:01] DEBUG[2926] func_db.c: DB: DEVICE//user not found in
database.
*
But my phone display is showing unknown caller. Please help

*Thanks,
Arun S*





Are you even paying for the service?
>
> Here in the US, on PSTN lines from the ILEC's, CallerID is a pay
> service, with 2 tiers. Number only, and number with name.
> Some CLEC's include this without extra charge, as do most/all VOIP
> providers.
>
> Do you have a box or phone, independent of the Asterisk box, that can
> display CallerID? Make sure first it is being delivered to you. You
> could also monitor the line on incoming calls and listen for the
> information being sent.
> You also need to determine what "standard"  or protocol is used to send
> the information, as worldwide there are several.
>
> John Novack
>
> Arun Sasidhar wrote:
> > Hi,
> >
> > I dont know the type of caller ID. What you mean by this?. I am
> > from India. I don't know more about this.
> > *
> > Thanks,
> > Arun S*
> >
> > On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen
> > mailto:tzafrir.co...@xorcom.com>> wrote:
> >
> > On Tue, Jan 05, 2010 at 06:54:18PM +0530,

Re: [asterisk-users] Some minor configuration issues with queues

2010-01-06 Thread jonas kellens

Please can someone help me with my queue-problems ?

When there is a member available for answering calls that entered the
queue, the caller does not hear the musiconhold as defined
"musicclass=default".

What happens is that when a second caller enters the queue, the caller
hears the ringtone for as long as the conversation of the first caller
takes.
It is as off there is nobody to pick up the phone, as in reality the
queue member is busy with an other conversation.

Jonas.


On Mon, 2010-01-04 at 09:51 +0100, jonas kellens wrote:

> Hello list !
> 
> I have some configuration issues with queues, but I'm sure they are
> minor and for someone who has already configured queues it could be
> trivial.
> 
> This is my queue configuration :
> 
> [VC_support_queue]
> musicclass = default
> strategy = ringall
> timeout = 20
> retry = 5
> wrapuptime=15
> autofill=yes
> autopause=no
> maxlen = 0
> setinterfacevar=yes
> announce-frequency = 0
> periodic-announce-frequency=0
> announce-holdtime = no
> ; announce-round-seconds = 10
> ; queue-thankyou=
> ;queue-youarenext = queue-youarenext
> ;queue-thereare = queue-thereare
> ;queue-callswaiting = queue-callswaiting
> ;queue-holdtime = queue-holdtime
> ;queue-minutes = queue-minutes
> ;queue-seconds = queue-seconds
> ;queue-thankyou = queue-thankyou
> ;queue-lessthan = queue-less-than
> ;queue-reporthold = queue-reporthold
> ;periodic-announce = queue-periodic-announce
> ; monitor-format = gsm|wav|wav49
> ; monitor-type = MixMonitor
> joinempty = strict
> ; leavewhenempty = yes
> eventwhencalled = no
> ; QueueMemberStatus
> eventmemberstatus = no
> reportholdtime = no
> ringinuse = no
> memberdelay = 0
> ; timeoutrestart = no
> member => SIP/VCsupport,1,Jonas
> member => Agent/VCjoeri,2,Joeri
> 
> First problem :
> 
> Although there is nobody in the queue, the caller is still inserted
> into the queue :
> 
> vps2301*CLI> queue show 
> VC_support_q has 0 calls (max unlimited) in 'ringall' strategy (0s
> holdtime), W:0, C:0, A:2, SL:0.0% within 0s
>Members: 
>   Jonas (SIP/VCsupport) with penalty 1 (Unavailable) has taken no
> calls yet
>   Joeri (Agent/VCjoeri) with penalty 2 (Unavailable) has taken no
> calls yet
>No Callers
> 
> [Jan  4 09:34:40] -- Executing [...@center:2]
> Queue("IAX2/zoiper-5307", "VC_support_queue|r") in new stack
> 
> 
> Second problem :
> 
> When a caller calls in and there is someone available in the queue,
> there is no music-on-hold while the caller waits. When the caller is
> directed to the agent there should be a ringtone indicating that the
> agent is called, but it stays silent. No music on hold, no ringtone...
> (this is a SIP-channel)
> 
> 
> Thank you for pointing out the misconfiguration.
> 
> 
> Kind regards,
> 
> Jonas.
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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread John Novack

Are you even paying for the service?

Here in the US, on PSTN lines from the ILEC's, CallerID is a pay 
service, with 2 tiers. Number only, and number with name.
Some CLEC's include this without extra charge, as do most/all VOIP 
providers.

Do you have a box or phone, independent of the Asterisk box, that can 
display CallerID? Make sure first it is being delivered to you. You 
could also monitor the line on incoming calls and listen for the 
information being sent.
You also need to determine what "standard"  or protocol is used to send 
the information, as worldwide there are several.

John Novack

Arun Sasidhar wrote:
> Hi,
>
> I dont know the type of caller ID. What you mean by this?. I am 
> from India. I don't know more about this.
> *
> Thanks,
> Arun S*
>
> On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen 
> mailto:tzafrir.co...@xorcom.com>> wrote:
>
> On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
> > Hi,
> >
> > I am using asterisknow 1.5.0 and Wildcard TDM410P card.
> Everything is
> > working fine except the caller ID of incoming call from PSTN
> line. The phone
> > display is showing "Unknown" when there is an incoming call. I
> think the
> > same problem listed here:
>  https://issues.asterisk.org/view.php?id=6683
> > There is one patch on this link but i don't know how to apply
> patch on
> > asterisknow. Is this patch will resolve my issue? Kindly help me
> to fix this
> > issue.
>
> What type of caller ID is used in that line?
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> 
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> 
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> -- 
> Thanks,
>
> Arun S
> System Administrator.
> Cabot Solutions
> www.cabotsolutions.com 
> 
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>
>
>
> Checked by AVG - www.avg.com 
> Version: 9.0.725 / Virus Database: 270.14.126/2602 - Release Date: 01/05/10 
> 14:35:00
>
>   

-- 
Dog is my co-pilot


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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Danny Nicholas

Another thing to consider (hopefully not covered in one of the 100 previous
replies) - Callerid is a "hit and miss" proposition;  to get "ABC Widgets
<205-555-1212>", The Telco has to have ABC Widgets in their database and
publish it.  If not, you get "Unknown <205-555-1212>".  Since a good chunk
of business get their calls from "known sources", an approach previously
discussed here is to try Telco first, then reference a database of frequent
callers if Unknown is return.  HTH.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Wednesday, January 06, 2010 8:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID on Indian PSTN is not working.

Are you even paying for the service?

Here in the US, on PSTN lines from the ILEC's, CallerID is a pay 
service, with 2 tiers. Number only, and number with name.
Some CLEC's include this without extra charge, as do most/all VOIP 
providers.

Do you have a box or phone, independent of the Asterisk box, that can 
display CallerID? Make sure first it is being delivered to you. You 
could also monitor the line on incoming calls and listen for the 
information being sent.
You also need to determine what "standard"  or protocol is used to send 
the information, as worldwide there are several.

John Novack

Arun Sasidhar wrote:
> Hi,
>
> I dont know the type of caller ID. What you mean by this?. I am 
> from India. I don't know more about this.
> *
> Thanks,
> Arun S*
>
> On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen 
> mailto:tzafrir.co...@xorcom.com>> wrote:
>
> On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
> > Hi,
> >
> > I am using asterisknow 1.5.0 and Wildcard TDM410P card.
> Everything is
> > working fine except the caller ID of incoming call from PSTN
> line. The phone
> > display is showing "Unknown" when there is an incoming call. I
> think the
> > same problem listed here:
>  https://issues.asterisk.org/view.php?id=6683
> > There is one patch on this link but i don't know how to apply
> patch on
> > asterisknow. Is this patch will resolve my issue? Kindly help me
> to fix this
> > issue.
>
> What type of caller ID is used in that line?
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> 
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> 
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> -- 
> Thanks,
>
> Arun S
> System Administrator.
> Cabot Solutions
> www.cabotsolutions.com 
> 
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>
>
>
> Checked by AVG - www.avg.com 
> Version: 9.0.725 / Virus Database: 270.14.126/2602 - Release Date:
01/05/10 14:35:00
>
>   

-- 
Dog is my co-pilot

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Re: [asterisk-users] Outgoing Calls Only -- Firewall Rules

2010-01-06 Thread Max McGraw

 Nicholas,

 Sorry I don't know, but are your calls working okay ?

 Depending on the verbosity level being set, I see warning
 msgs all the time, that I ignore.

 Frequently, an upgrade to the next release of the same
 major version also eliminates the warning msgs.

 If you are really concerned, I would find an unused machine,
 install Linux & Asterisk 1.6.x on it, try out your calls and
 see if the warnings still appear.

 If there are no warnings of this kind, it is an issue specific
 to a module in that 1.4.x release and likely to go away.

 Good luck !

--

  On Tue, Jan 5, 2010,   Nicholas Blasgenwrote:

> Asterisk 1.4.29 or so.
>
> access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any range
> 1 2
> access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any eq
> 5060
>
> But yes, all your feedback worked.  I didn't need to port-forward any
> incoming ports, only 5060/1-2 for outgoing UDP.  The only issue I'm
> now having is:
>
> <--- SIP read from 66.227.100.20:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 209.34.93.68:5060;branch=z9hG4bK3eb38bde;rport=51566
> 
> Warning: 392 66.227.100.20:5060 "Noisy feedback tells:  pid=9611
> req_src_ip=209.34.93.68 req_src_port=51566 in_uri=sip:sip.jnctn.net
> out_uri=sip:sip.jnctn.net via_cnt==1"
>
> 209.34.93.68 is my IP, 209.34.93.68 is Junction Networks (for this
> example).  I also get it from my backbone providers as well so it's likely
> something to do with that 51566 req_src_port thing.  Any idea what this is
> an how to configure it to a restricted range of IP addresses?
>
> Nicholas Blasgen
> Partner / Network Operations
> Refractive Dialer LLC
> (724) 252-7436
>
>
> On Sun, Jan 3, 2010 at 8:29 PM,  Max McGraw   wrote:
>>
>>  Nicholas,
>>
>>  you haven't specified which version, which does make
>>  a lot of difference.
>>
>>  1.6.x  can easily traverse NAT. If you are only making
>>  outbound calls, you shouldn't need to forward 5060.
>>
>>  Unless you have a special NAT that is blocking
>>  outbound connections, the  SIP.conf  settings below
>>  should work whether your provider uses SIP
>>  registrations or not. My codec related settings may
>>  not be applicable to your installation :
>>
>>  ; -
>>  [general]
>>  dtmfmode=rfc2833
>>  relaxdtmf=yess
>>  bandwidth=high
>>  disallow=all
>>  allow=ulaw
>>  ;
>>  ;   NAT stuff
>>  ;
>>  localnet=192.168.x.0/255.255.255.0
>>  externip=a.b.c.d:5060
>>  nat=yes
>>  ;
>>  ;   Media stuff
>>  ;
>>  canreinvite=no
>>  ;
>>  ;
>>  [your-voip-provider-para]
>>  ;
>>  context=default
>>  type=friend
>>  ;
>>  ;  your provider's outbound gateway
>>  ;
>>  host=w.x.y.z
>>  ;
>>  dtmfmode=rfc2833
>>  relaxdtmf=yess
>>  disallow=all
>>  allow=ulaw
>>  ;
>>  ; -
>>
>>
>>  On Sun, Jan 3, 2010,   Nicholas Blasgen    wrote:
>>
>> > I'm trying to move my Asterisk deployments under a Virtual IP address
>> > and
>> > now remember why I dislike this.  My primary Asterisk system is now
>> > behind a
>> > firewall in private address space.  My question is what ports are needed
>> > to
>> > be opened just for the purpose of placing outgoing calls.  I would have
>> > assumed none, but I can't even get replies on registration from any of
>> > my 3
>> > VoIP providers.  I tried defining the External IP and some other stuff,
>> > but
>> > I assume it's fully an issue with the firewall.  Do I really need 5060
>> > port
>> > forwarded just to register with remote hosts?
>> >
>> > Nicholas Blasgen
>> > Partner / Network Operations
>> > Refractive Dialer LLC
>> > (724) 252-7436
>> >
>> > __

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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Jeff LaCoursiere


On Wed, 6 Jan 2010, Arun Sasidhar wrote:

> Hi,
>
>I dont know the type of caller ID. What you mean by this?. I am from
> India. I don't know more about this.
> *
> Thanks,
> Arun S*

Hi Arun,

Just for fun I read over the bug id you quoted below, and it seems there 
are a number of settings you may need to try to get your particular 
situation working.  Have you done this?  You may need to open another 
issue on Mantis if not.  It seems India is not very consistent with its 
CID methods.  It also seems from the issue quoted that there are hardware 
dependancies.  If you are using hardware and a provider that is quoted by 
the issue as "resolved", you should be fine, as the changes were comitted 
to 1.4 a long time ago (assuming you set your cid options correctly).

j

>
> On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen wrote:
>
>> On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
>>> Hi,
>>>
>>> I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
>>> working fine except the caller ID of incoming call from PSTN line. The
>> phone
>>> display is showing "Unknown" when there is an incoming call. I think the
>>> same problem listed here:  https://issues.asterisk.org/view.php?id=6683
>>> There is one patch on this link but i don't know how to apply patch on
>>> asterisknow. Is this patch will resolve my issue? Kindly help me to fix
>> this
>>> issue.
>>
>> What type of caller ID is used in that line?
>>
>> --
>>Tzafrir Cohen
>> icq#16849755  
>> jabber:tzafrir.co...@xorcom.com
>> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
>> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> -- 
> Thanks,
>
> Arun S
> System Administrator.
> Cabot Solutions
> www.cabotsolutions.com
>

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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Arun Sasidhar

Hi,

I dont know the type of caller ID. What you mean by this?. I am from
India. I don't know more about this.
*
Thanks,
Arun S*

On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen wrote:

> On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
> > Hi,
> >
> > I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
> > working fine except the caller ID of incoming call from PSTN line. The
> phone
> > display is showing "Unknown" when there is an incoming call. I think the
> > same problem listed here:  https://issues.asterisk.org/view.php?id=6683
> > There is one patch on this link but i don't know how to apply patch on
> > asterisknow. Is this patch will resolve my issue? Kindly help me to fix
> this
> > issue.
>
> What type of caller ID is used in that line?
>
> --
>Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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Re: [asterisk-users] Inquiry:How to define incoming route for sip?

2010-01-06 Thread hadi motamedi

On Wed, Jan 6, 2010 at 11:55 AM, Tzafrir Cohen wrote:

> Hi,
>
> I noticed you always prefix 'Inquiry:' to your questions on the list.
> This is implied from the subject line itself, and wastes some space in
> the subject line, so I guess it is kind of pointless.
>
> Now to the question itself,
>
> On Wed, Jan 06, 2010 at 10:44:31AM +, hadi motamedi wrote:
>
> > Can you please let me know how can I define incoming route to accept
> > incoming calls from an external sip server?
>
> Just send them there?
>
> > I have defined the following profile for my sip phone :
> > Under sip.conf :
> > -
> > [osaka]
> > type=friend
> > context=sip-outgoing
> > host=192.168.0.139
> > disallow=all
> > allow=alaw
>
> This looks like a local phone, and you direct all the calls coming from
> it to the context 'sip-outgoing' .
>
> > [6672019]
> > type=friend
> > context=sip-outgoing
> > canreinvite=no
> > host=dynamic
> > nat=no
>
> Likewise this one (though it registers).
>
> >
> > Under extensions.conf :
> > 
> > [sip-outgoing]
> > include=sip_outgoing
> > [sip_outgoing]
> > exten => _XXX,1,Dial(SIP/osaka/${EXTEN})
> > [line-incoming]
> > exten => _6XX,1,Dial(SIP/${EXTEN})
>
> Could you explain what you actually want to do? Where do you expect
> those SIP calls will come from?
>
> >
> > Please be informed that the sip outbound toward the external sip server
> is
> > quite ok , but sip incoming is not working . Can you please let me know
> why
> > my incoming route is not working properly ?
>
> I would actually go the other way around. Please try to convince us
> (which also implies: convince yourself) that your setup should work.
> Please try to explain why an incoming call should work according to your
> configuration.
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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>


Thank you for your reply . I want to correctly route the incoming calls
coming from external sip (named in my profile as osaka) to the destination
(that is my Asterisk subscriber sip phone) . To this end , I defined the
osaka profile in my sip.conf and my Asterisk subscriber phone is at 6672019
that I have defined his profile in my sip.conf as well (as you saw it) .
Then , I tried to define the [sip-outgoing] route in my extensions.conf for
rourting my Asterisk sip subscriber outgoing calls toward the external sip
server (named osaka) and it works here . But my [line-incoming] route for
accepting incoming sip calls from external sip server (osaka) toward my
Asterisk subscriber sip phone at 6672019 fails . Can you please let me know
what is wrong in my configuration ?
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Re: [asterisk-users] Inquiry:How to define incoming route for sip?

2010-01-06 Thread Tzafrir Cohen

Hi,

I noticed you always prefix 'Inquiry:' to your questions on the list.
This is implied from the subject line itself, and wastes some space in
the subject line, so I guess it is kind of pointless.

Now to the question itself,

On Wed, Jan 06, 2010 at 10:44:31AM +, hadi motamedi wrote:

> Can you please let me know how can I define incoming route to accept
> incoming calls from an external sip server?

Just send them there?

> I have defined the following profile for my sip phone :
> Under sip.conf :
> -
> [osaka]
> type=friend
> context=sip-outgoing
> host=192.168.0.139
> disallow=all
> allow=alaw

This looks like a local phone, and you direct all the calls coming from
it to the context 'sip-outgoing' .

> [6672019]
> type=friend
> context=sip-outgoing
> canreinvite=no
> host=dynamic
> nat=no

Likewise this one (though it registers).

> 
> Under extensions.conf :
> 
> [sip-outgoing]
> include=sip_outgoing
> [sip_outgoing]
> exten => _XXX,1,Dial(SIP/osaka/${EXTEN})
> [line-incoming]
> exten => _6XX,1,Dial(SIP/${EXTEN})

Could you explain what you actually want to do? Where do you expect
those SIP calls will come from?

> 
> Please be informed that the sip outbound toward the external sip server is
> quite ok , but sip incoming is not working . Can you please let me know why
> my incoming route is not working properly ?

I would actually go the other way around. Please try to convince us
(which also implies: convince yourself) that your setup should work.
Please try to explain why an incoming call should work according to your
configuration.

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

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[asterisk-users] Zaptel compilation problems: Data Mode!!

2010-01-06 Thread mosleh

Hi all,
I need Help. I want to compile zaptel in data mode but i got this errors:

/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_xmitâ:
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618: error: implicit
declaration of function âhdlc_statsâ
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618: warning:
initialization makes pointer from integer without a cast
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_ppp_xmitâ:
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1722: warning: comparison of
distinct pointer types lacks a cast
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1785: warning: comparison of
distinct pointer types lacks a cast
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function
â__putbuf_chunkâ:
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6806: warning:
initialization makes pointer from integer without a cast
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6819: warning:
initialization makes pointer from integer without a cast
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6912: warning:
initialization makes pointer from integer without a cast
make[3]: *** [/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.o] Error 1
make[2]: *** [_module_/usr/src/zaptel-1.4.12.1/kernel] Error 2
make[2]: Leaving directory `/usr/src/linux-2.6.27.7'
make[1]: *** [modules] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4.12.1'
make: *** [all] Error 2





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Re: [asterisk-users] Fastagi-mapping problem

2010-01-06 Thread ABBAS SHAKEEL

You can try this
[agi_test]
exten => 123,1,Answer();
exten => 123,n,noop(${CALLERID(num)})
exten => 123,n,set(IP_FOR_AGI=192.168.127.58)
exten =>123,n,Agi(agi://${IP_FOR_AGI}/com.package.ClassName)


On Wed, Jan 6, 2010 at 2:28 PM, ABBAS SHAKEEL
wrote:

> Hi,
>
> You can directly call that class like  AGI(com.abc.cde.Hello) . Hello is
> class name.
>
> Hope this helps
>
> On Wed, Jan 6, 2010 at 2:16 PM, ahmed magdy wrote:
>
>> Hello
>>
>> I am new  in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started
>> the first example Hello AGI in this tutorial
>> http://asterisk-java.org/development/tutorial.html  I put the jar file
>> and the proparty file in folder
>> i wrote in extensions.conf this  exten => 1300,1,AGI(agi://
>> 192.168.50.127/hello.agi,${EXTEN},${UNIQUEID},${CALLERID(name)})
>>
>> I started AGI server ,then when i call extension 1300 i am facing this
>> error
>>  Resource bundle 'fastagi-mapping' is missing
>> and  No script configured for URL 'agi://192.168.50.127/hello.agi'
>> (script 'hello.agi')
>> please can anyone help me ?
>> Thanks
>>
>> --
>> Ahmed Magdy Mahmoud
>>
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best Regards
> Shakeel Abbas
>
>


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Shakeel Abbas
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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Tzafrir Cohen

On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
> Hi,
> 
> I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
> working fine except the caller ID of incoming call from PSTN line. The phone
> display is showing "Unknown" when there is an incoming call. I think the
> same problem listed here:  https://issues.asterisk.org/view.php?id=6683
> There is one patch on this link but i don't know how to apply patch on
> asterisknow. Is this patch will resolve my issue? Kindly help me to fix this
> issue.

What type of caller ID is used in that line?

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

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[asterisk-users] Inquiry:How to define incoming route for sip?

2010-01-06 Thread hadi motamedi

Dear All
Can you please let me know how can I define incoming route to accept
incoming calls from an external sip server?
I have defined the following profile for my sip phone :
Under sip.conf :
-
[osaka]
type=friend
context=sip-outgoing
host=192.168.0.139
disallow=all
allow=alaw
[6672019]
type=friend
context=sip-outgoing
canreinvite=no
host=dynamic
nat=no

Under extensions.conf :

[sip-outgoing]
include=sip_outgoing
[sip_outgoing]
exten => _XXX,1,Dial(SIP/osaka/${EXTEN})
[line-incoming]
exten => _6XX,1,Dial(SIP/${EXTEN})

Please be informed that the sip outbound toward the external sip server is
quite ok , but sip incoming is not working . Can you please let me know why
my incoming route is not working properly ?
Thank you
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[asterisk-users] MITEL

2010-01-06 Thread phiroc

Hello,

can any of the Asterisk API be used to dial a number on MITEL telephones?

Many thanks.

phiroc

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Re: [asterisk-users] Fastagi-mapping problem

2010-01-06 Thread ABBAS SHAKEEL

Hi,

You can directly call that class like  AGI(com.abc.cde.Hello) . Hello is
class name.

Hope this helps

On Wed, Jan 6, 2010 at 2:16 PM, ahmed magdy wrote:

> Hello
>
> I am new  in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started
> the first example Hello AGI in this tutorial
> http://asterisk-java.org/development/tutorial.html  I put the jar file and
> the proparty file in folder
> i wrote in extensions.conf this  exten => 1300,1,AGI(agi://
> 192.168.50.127/hello.agi,${EXTEN},${UNIQUEID},${CALLERID(name)})
>
> I started AGI server ,then when i call extension 1300 i am facing this
> error
>  Resource bundle 'fastagi-mapping' is missing
> and  No script configured for URL 'agi://192.168.50.127/hello.agi' (script
> 'hello.agi')
> please can anyone help me ?
> Thanks
>
> --
> Ahmed Magdy Mahmoud
>
>
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Shakeel Abbas
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[asterisk-users] Fastagi-mapping problem

2010-01-06 Thread ahmed magdy

Hello

I am new  in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started the
first example Hello AGI in this tutorial
http://asterisk-java.org/development/tutorial.html  I put the jar file and
the proparty file in folder
i wrote in extensions.conf this  exten => 1300,1,AGI(agi://
192.168.50.127/hello.agi,${EXTEN},${UNIQUEID},${CALLERID(name)})
I started AGI server ,then when i call extension 1300 i am facing this error
 Resource bundle 'fastagi-mapping' is missing
and  No script configured for URL 'agi://192.168.50.127/hello.agi' (script
'hello.agi')
please can anyone help me ?
Thanks

-- 
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Re: [asterisk-users] CallerID on Indian PSTN is not working.

2010-01-06 Thread Arun Sasidhar

Hi,

  But the caller ID function is still not working my system.

Please Help.

Thanks,
Arun S

On Wed, Jan 6, 2010 at 11:13 AM, Kyle Kienapfel wrote:

> On Tue, Jan 5, 2010 at 5:24 AM, Arun Sasidhar
>  wrote:
> > Hi,
> >
> > I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
> > working fine except the caller ID of incoming call from PSTN line. The
> phone
> > display is showing "Unknown" when there is an incoming call. I think the
> > same problem listed here:  https://issues.asterisk.org/view.php?id=6683
> > There is one patch on this link but i don't know how to apply patch on
> > asterisknow. Is this patch will resolve my issue? Kindly help me to fix
> this
> > issue.
> >
>
> Hello,
> The last comment on that page you linked says the patch was applied to
> the source in June of 2007.
>
> Cheers
>
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-- 
Thanks,

Arun S
System Administrator.
Cabot Solutions
www.cabotsolutions.com
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Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-06 Thread Randy R

On Wed, Jan 6, 2010 at 8:50 AM, Allann Jones  wrote:
> But jailbreaking increases the freedom to develop a application and

Oh, I agree with you, but it's probably even better to make a decision
to either buy into the constraints of Apple or find a better, free-er
phone, which is what I hope a lot of people will be doing in the next
few years. In a vain attempt to return to VoIP and Asterisk, let's
hope that more future mobile OS will all allow multi-apps so that you
can leave a SIP client running in the background.

The big problem with jailbreaking is updates. If you have a lot of
time and energy  to manage that problem when it comes up, jailbreaking
is "fun" in a geeky sort of way.

/r

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