date:20120112

sir,
Can you explain me
At a time no.of fax Recieve is possible ?

Best Regards,

Mahesh Katta
**


On Thu, Jan 12, 2012 at 11:27 PM, Ioan Indreias  wrote:

> On Thu, Jan 12, 2012 at 7:50 PM, mahesh katta 
> wrote:
> > I was search for free license but for this Digium require purchase any
> > Hardware then they can provide Free License.
> > But I have no Digium Device , I am using Grand stream FXO Gateway and
> > Asterisk.1.8.XX .
> > I was connected like
> > PSTN==>FXOGateway==>Asterisk(FXO configure through IP)
> >
> > If anything wrong please correct me.
>
> http://store.digium.com/products.php?category_id=94
>
> HTH,
> Ioan
>
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Re: [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1

2012-01-12 Thread Jim Dickenson

One good thing is now that you know what the problem is you should be able to 
work with zopier support and get them to fix zopier. They have been very 
responsive to a couple problems I have and I am running the free version.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jan 12, 2012, at 3:03 PM, Kevin P. Fleming wrote:

> On 01/12/2012 11:58 AM, Alex Villacís Lasso wrote:
> 
>> I have discovered the root cause of the issue. Due to a peculiarity of
>> Zoiper 2.18, this program will *not* send a ACCEPT or RINGING packet
>> back to Asterisk unless the NEW packet that announces the incoming call
>> contains an IAX_IE_CALLING_NUMBER information element. It does not
>> matter if the calling number is empty, but the corresponding IE must
>> exist. This behavior is a change between Asterisk 1.6 and Asterisk 1.8.
> 
> Well, I applaud your troubleshooting skills and analysis... well done!
> 
> Unfortunately, that IE is *not* mandatory in an IAX2 NEW packet, and thus 
> Zoiper failing to properly process such NEW packets is a bug in Zoiper. Yes, 
> Asterisk's behavior has changed (since Caller ID handling was overhauled in 
> Asterisk 1.8, while adding Connected ID support), but both the old and new 
> behavior are compliant with the IAX2 protocol.
> 
> -- 
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> 
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Re: [asterisk-users] Problems with codec translation when using Monitor and MixMonitor

2012-01-12 Thread Jim Dickenson

Here is a matrix we put together about g729 license needs:

 == 
= == ===  
Asterisk to SIP Provider SIP Client to Asterisk asterisk.conf sln 
defined record monitor encoders decoders
 == 
= == ===  
ulaw ulaw   yes 
  yesyes00
ulaw ulaw   yes 
  yesno 00
ulaw ulaw   yes 
  no no 00
ulaw ulaw   yes 
  no yes00

ulaw ulaw   no  
  yesyes00
ulaw ulaw   no  
  yesno 00
ulaw ulaw   no  
  no no 00
ulaw ulaw   no  
  no yes00

ulaw g729   yes 
  yesyes33
ulaw g729   yes 
  yesno 23
ulaw g729   yes 
  no no 11
ulaw g729   yes 
  no yes33

ulaw g729   no  
  yesyes33
ulaw g729   no  
  yesno 23
ulaw g729   no  
  no no 11
ulaw g729   no  
  no yes33

g729 ulaw   yes 
  yesyes25
g729 ulaw   yes 
  yesno 25
g729 ulaw   yes 
  no no 11
g729 ulaw   yes 
  no yes23

g729 ulaw   no  
  yesyes25
g729 ulaw   no  
  yesno 25
g729 ulaw   no  
  no no 11
g729 ulaw   no  
  no yes23

g729 g729   yes 
  yesyes47
g729 g729   yes 
  yesno 37
g729 g729   yes 
  no no 11
g729 g729   yes 
  no yes45

g729 g729   no  
  yesyes47
g729 g729   no  
  yesno 37
g729 g729   no  
  no no 11
g729 g729   no  
  no yes45

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

CfMC
http://www.cfmc.com/



On Jan 12, 2012, at 3:00 PM, Kevin P. Fleming wrote:

> On 01/12/2012 11:57 AM, Daniel - Asterisk wrote:
>> The simplest answer, I purchased one additional license and one
>> simultaneous call is being recorded now. I do not understand why the
>> ulaw codec (or format) is involved here (... No translator path from
>> alaw to unknown ...)
>> 
>> Any entry

Re: [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1


On 01/12/2012 11:58 AM, Alex Villacís Lasso wrote:


I have discovered the root cause of the issue. Due to a peculiarity of
Zoiper 2.18, this program will *not* send a ACCEPT or RINGING packet
back to Asterisk unless the NEW packet that announces the incoming call
contains an IAX_IE_CALLING_NUMBER information element. It does not
matter if the calling number is empty, but the corresponding IE must
exist. This behavior is a change between Asterisk 1.6 and Asterisk 1.8.


Well, I applaud your troubleshooting skills and analysis... well done!

Unfortunately, that IE is *not* mandatory in an IAX2 NEW packet, and 
thus Zoiper failing to properly process such NEW packets is a bug in 
Zoiper. Yes, Asterisk's behavior has changed (since Caller ID handling 
was overhauled in Asterisk 1.8, while adding Connected ID support), but 
both the old and new behavior are compliant with the IAX2 protocol.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Problems with codec translation when using Monitor and MixMonitor


On 01/12/2012 11:57 AM, Daniel - Asterisk wrote:

The simplest answer, I purchased one additional license and one
simultaneous call is being recorded now. I do not understand why the
ulaw codec (or format) is involved here (... No translator path from
alaw to unknown ...)

Any entry will be very appreciated.


When you say 'call', do you mean a call between two phones (endpoints)? 
If so, and both endpoints are using G.729 for audio, then yes, recording 
that call in any format other than G.729 will require *two* G.729 
decoders, one for each audio stream being received by Asterisk. Even in 
a case where you are only recording the combined audio from the two 
phones (MixMonitor), the audio must still be decoded in order to be mixed.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Questions on hardware or software-based echo cancellation


On 01/12/2012 06:39 AM, Olivier wrote:

Hi,

I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.

I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.

1. I'm correct thinking that it is then impossible to switch from
hardware to software echo can without removing the VPMOCT64 module
itself ?
2. Does the same also apply to HA8 and its VPMOCT032 module ?


With DAHDI 2.6 (and possibly 2.5), it is possible to override the 
configuration and apply a software echo canceller to a channel even if 
it has a hardware one. With prior versions, yes, the echo cancellation 
module would have to be physically removed (or disabled using a 
parameter to the kernel module).


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] ConfBridge no audio problem


On 01/03/2012 10:14 PM, Roi Stork wrote:

We're encountering no audio issues in ConfBridge.

Only the moderator and the 1st invited have audio.
When the 2nd invited number picks up the phone, only the announcement is
heard, then followed by silence.

This occurs frequently. Only 1 out of 5 test calls do not have this problem.

We were hoping that switching from MeetMe to ConfBridge will solve the
no audio issue, since there's no more audio mixing with DAHDI.
What else could be causing this problem?

By the way, our asterisk version is 1.6.2.14


ConfBridge in Asterisk 1.6.2 and Asterisk 1.8 was effectively 
'experimental'; it was there to exercise the new 'bridging API', but 
really was not stable enough for production usage.


ConfBridge in Asterisk 10 was nearly completely rewritten, and is now 
preferred over MeetMe in that version of Asterisk.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-12 Thread Lefteris Zafiris

On 01/12/2012 05:50 PM, Danny Nicholas wrote:
> Two more "offerings" - #1 - add DTMF parameter so function can be stopped by
> pressing a digit or digits other than * or #  - #2 - add an option to
> "silence" the beep.  If you were using this in an IVR and wanted to say
> "press 1 or say help for help",  silencing the beep before recording would
> (IMO) make the rendering sound more "professional"/less "mechanical".

Both features added:

-
Usage
-
agi(speech-recog.agi,[lang],[timeout],[intkey],[NOBEEP])
Records from the current channel untill the timeout (set to 10 seconds
by default, -1 for no timeout) is reached or the interrupt key (# by
default) is pressed.
If NOBEEP is set, no beep sound is played back to the user to indicate
the start of the recording.

There is now also the option to enable SSL for encrypted communication
between your pbx and the google voice server.

Updated code can be found here:
https://github.com/zaf/asterisk-speech-recog/tarball/master

Lefteris Zafiris

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[asterisk-users] t38modem v2, which version or patch of asterisk?

2012-01-12 Thread cyber.f...@infinito.it

Hi I'm developing for my company an Asterisk+t38modem+Hylafax solution. 
I'm struggling in asterisk<->t38modemv2 comunication. I've tried lot of 
asterisk version but noone seem to function well. In asterisk 1.8.8.8 
for example i can receive fax but I'm not able to send faxes due to a 
"SIP/2.0 488 Not acceptable here" in asterisk 1.6 and 1.4 T38 codec 
seems to be unsupported.

Asterisk 1.4.20 reports:
channel.c:3153 ast_channel_make_compatible: No path to translate from 
SIP/T38modem-0-084e8dd8(256) to SIP/audiocodes_mp114-084d7858(8)

Is there an Asterisk version that could go with t38modem?
thank a lot.
I hope that someone answer becouse I'm very not so far to be unenploied...

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Re: [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1

2012-01-12 Thread Alex Villacís Lasso

El 11/01/12 18:01, Alex Villacís Lasso escribió:

El 11/01/12 15:37, Alex Villacís Lasso escribió:
I am investigating an issue with IAX2 extensions in Asterisk 1.8.x. My application connects to Asterisk via AMI and attempts to run an Originate command between an extension (such as SIP/ or IAX2/) and an application (in my case it is
AgentLogin). This works correctly for SIP extensions, in all Asterisk versions. With IAX2 extensions, this worked correctly in Asterisk 1.6.2.20, but started failing sometimes in Asterisk 1.8.7, and now happens every time in Asterisk 1.8.8.1. I found
out that any application (not just AgentLogin) will trigger the issue. Instead of successfully ringing the IAX2 extension, as expected, the Originate attempt just sits there. The IAX2 extension does not receive any ringing indication. I can reproduce
the issue by running the following command from the Asterisk console:

originate IAX2/1099 application playback demo-congrats

This is supposed to ring the extension, and upon picking up, should play the audio file. Instead, the IAX2 extension sits idle. Also, the Asterisk console becomes unresponsive. If I try to execute any other command (such as "iax2 show threads", or even
"help"), I get a prompt back but no command output. Then, after some time (the ring timeout, maybe), I get the output of all commands I issued during the hang.

When my application connects to AMI and runs the Originate command, it
eventually gets a Hangup event, as if the extension never picked up the
ringing. But actually the ringing never made it to the IAX2 extension.

We have noticed that the IAX2 extension itself can place calls to a SIP extension normally during the Originate hang, but it cannot receive a call from another SIP extension (Busy Here). When not attempting the Originate call, the IAX2 extension appears
to behave normally.

This has been triggered in three machines to date: a big server with some 40
IAX2 extensions, and two test machines (one physical and one virtual machine).

Before I get into a bug hunt, I would like to know: Is this a known issue? Are
there any pointers on where to look first, or what to look for, based on my
symptoms?

Testing with Asterisk 1.8.8.1 x86_64 and Zoiper as an IAX2 client.
Some additional information - it is the Originate with IAX2 channel that has problems, not just applications. Given that 1099 is an IAX2 extension and 1065 is a SIP extension, with FreePBX contexts, i found that "originate IAX2/1099 extension
1065@from-internal" hangs, but "originate SIP/1065 extension 1099@from-internal" succeeds.

Some more information about this:

I have discovered the root cause of the issue. Due to a peculiarity of Zoiper 2.18, this program will *not* send a ACCEPT or RINGING packet back to Asterisk unless the NEW packet that announces the incoming call contains an IAX_IE_CALLING_NUMBER
information element. It does not matter if the calling number is empty, but the corresponding IE must exist. This behavior is a change between Asterisk 1.6 and Asterisk 1.8.

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Re: [asterisk-users] Problems with codec translation when using Monitor and MixMonitor

2012-01-12 Thread Daniel - Asterisk

The simplest answer, I purchased one additional license and one
simultaneous call is being recorded now. I do not understand why the ulaw
codec (or format) is involved here (... No translator path from alaw to
unknown ...)

Any entry will be very appreciated.

Elder D. Arohuanca

Lima - Peru

On Wed, Jan 11, 2012 at 6:10 PM, Daniel - Asterisk wrote:

> Hi folks,
>
> I'm having problems when I try to record my calls using MixMonitor or
> Monitor. Calls are working well and audio quality is good.
> But I just can't get recorded audio in one leg with both applications. It
> happens with internal calls too. As it seems, the problem is my g729
> licensing escheme. (just one license installed)
>
> What is the least number of licenses that are needed per recorded call?
> or what can I do to fix it?
>
> My asterisk version: Asterisk 1.8.7.1
>
> Logs with MixMonitor:
> [Jan 11 17:55:48] WARNING[19500]: translate.c:256
> ast_translator_build_path: No translator path from alaw to unknown
> [Jan 11 17:55:48] WARNING[19500]: translate.c:256
> ast_translator_build_path: No translator path from alaw to unknown
>
> testpbx*CLI> g729 show licenses
> 0/1 encoders/decoders of 1 licensed channels are currently in use
>
> Licenses Found:
> File: G729-... -- Key: G729-... -- Host-ID: ... -- Channels: 1 (Expires:
> ...) (OK)
>
>
> Logs with Monitor:
> [Jan 11 17:49:49] WARNING[19491]: translate.c:256
> ast_translator_build_path: No translator path from alaw to g723
> [Jan 11 17:49:49] WARNING[19491]: file.c:186 ast_writestream: Unable to
> translate to format wav49, source format g729
>
> testpbx*CLI> g729 show licenses
> 0/1 encoders/decoders of 1 licensed channels are currently in use
>
> Licenses Found:
> File: G729-...lic -- Key: G729-... -- Host-ID: ... -- Channels: 1
> (Expires: ...) (OK)
>
>
> I've searched for this on forums but can't find a complete answer yet.
>
> Thanks!
>
> Elder D. Arohuanca
>
> Lima - Peru
>
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Re: [asterisk-users] FAX Installation in Asterisk

2012-01-12 Thread Ioan Indreias

On Thu, Jan 12, 2012 at 7:50 PM, mahesh katta  wrote:
> I was search for free license but for this Digium require purchase any
> Hardware then they can provide Free License.
> But I have no Digium Device , I am using Grand stream FXO Gateway and
> Asterisk.1.8.XX .
> I was connected like
> PSTN==>FXOGateway==>Asterisk(FXO configure through IP)
>
> If anything wrong please correct me.

http://store.digium.com/products.php?category_id=94

HTH,
Ioan

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Re: [asterisk-users] FAX Installation in Asterisk

I was search for free license but for this Digium require purchase any
Hardware then they can provide Free License.
But I have no Digium Device , I am using Grand stream FXO Gateway and
Asterisk.1.8.XX .
I was connected like
PSTN==>FXOGateway==>Asterisk(FXO configure through IP)

If anything wrong please correct me.





Best Regards,

Mahesh Katta
**


On Thu, Jan 12, 2012 at 7:05 PM, Ruben Rögels <
ruben.roeg...@jumping-frog.org> wrote:

> Am 12.01.2012 14:09, schrieb mahesh katta:
> > WARNING[6982]: pbx.c:1851 pbx_extension_helper: No application
> > 'ReceiveFAX' for extension (macro-faxin, s,
> > 12)
> > [Jan 12 18:36:00]   == Spawn extension (macro-faxin, s, 12) exited
> > non-zero on 'SIP/gxw-000b'
> > Best Regards,
> >
> > Mahesh Katta
> > *BUZZ**WORKS*Business Services Private Limited
> > BANGALORE | CHENNAI | HYDERABAD |MUMBAI| DELHI
> > 222, Arunvihar,Sector-28, Noida 201301
> > GSM+91.3 45699 | Phone +91.12.0431.0581
> > Webhttp://www.buzzworks.com
> > 
>
> Hi Mahesh,
>
> I'm running asterisk 1.6.2.21 on Ubuntu 10.4.3 LTS
>
> On a Debian based linux distribution you need the following packets:
>
> libtiff-tools
> libtiff4
>
> To receive fax, you'll need the ReceiveFAX Application. You can get it
> for personal use directly from digium.com , just search for "FreeFAX for
> asterisk"
>
> (http://store.digium.com/productview.php?product_code=804-7)
>
> best regards,
> Ruben
>
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Re: [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)

2012-01-12 Thread Ishfaq Malik

Hi Jim

Not really, I looked at that earlier and it doesn't really apply to
this. Even with unanswered=no I should get 1 CDR entry for an abandoned
call and with unanswered=yes I would get a CDR entry for every interface
that was rung which I don't want.

Ish

On Thu, 2012-01-12 at 11:15 -0500, Jim DeVito wrote:
> I think in your cdr.conf you are looking for the unanswered=
> directive. 
> 
> Thanks!! 
> 
> Jim 
> 
> - Original message - 
> > Hi 
> > 
> > I'm using 1.8.7.0 with the RealTime architecture. 
> > 
> > If a call goes into application Queue and is abandoned by the
> caller, no 
> > entry is made in the CDR. Entries are made into the queue log. 
> > 
> > This cannot be correct behaviour, all calls should show in the CDR. 
> > 
> > Could anyone else try to reproduce this and if others get the same 
> > thing, I'll raise a bug on it. 
> > 
> > Thanks 
> > 
> > Ish 
> > -- 
> > Ishfaq Malik 
> > Software Developer 
> > PackNet Ltd 
> > 
> > Office: 0161 660 3062 
> > 
> > 
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> 
> 
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-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)

2012-01-12 Thread Jim DeVito

I think in your cdr.conf you are looking for the unanswered= directive.

Thanks!!

Jim

- Original message -
> Hi
> 
> I'm using 1.8.7.0 with the RealTime architecture.
> 
> If a call goes into application Queue and is abandoned by the caller, no
> entry is made in the CDR. Entries are made into the queue log. 
> 
> This cannot be correct behaviour, all calls should show in the CDR.
> 
> Could anyone else try to reproduce this and if others get the same
> thing, I'll raise a bug on it.
> 
> Thanks
> 
> Ish
> -- 
> Ishfaq Malik
> Software Developer
> PackNet Ltd
> 
> Office:     0161 660 3062
> 
> 
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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-12 Thread Danny Nicholas

Two more "offerings" - #1 - add DTMF parameter so function can be stopped by
pressing a digit or digits other than * or #  - #2 - add an option to
"silence" the beep.  If you were using this in an IVR and wanted to say
"press 1 or say help for help",  silencing the beep before recording would
(IMO) make the rendering sound more "professional"/less "mechanical".

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lefteris
Zafiris
Sent: Saturday, January 07, 2012 6:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Speech recognition in asterisk using google
voice API

On 01/07/2012 09:34 AM, Bruce B wrote:
> Added two new features to the script: Timeout value and speechdata type.
> 
> *exten => s,n,agi(speech-recog.agi,en-US,3000,phoneNumb)*
> - Will listen for 3 seconds and sanitize return as a single number 
> without any spaces in between. This helps when one reads phone number 
> in format
> 415-554-2323 and google returns, "415 554 2323" as result which is not 
> very usable.
> 
> *exten => s,n,agi(speech-recog.agi,en-US,2,string)*
> - Will listen for 20 second and return result as provided by Google 
> untouched.
> 
> It would be great to see them in future versions as I seem to need 
> them dearly in a real life scenario.
> 
> Updated script attached.
> 
> -Bruce

Thank you Bruce for the testing and the suggestions.
Both features added in the script. Timeout can now be set by the user, also
-1 means no timeout and the recording keeps going till # is pressed.
Space gets stripped between digits, this is now the default behavior and
there's no need to determine the 'speechdata' type.
The updated code can be found here:
https://github.com/zaf/asterisk-speech-recog/tarball/master

Next on my TODO list is to make use of the asterisk speech recognition API
(https://wiki.asterisk.org/wiki/display/AST/Speech+Recognition+API)
This will make the application actually usable for real case scenarios and
not a proof of concept as it is now.


Lefteris Zafiris

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Re: [asterisk-users] FAX Installation in Asterisk

2012-01-12 Thread Ruben Rögels

Am 12.01.2012 14:09, schrieb mahesh katta:
> WARNING[6982]: pbx.c:1851 pbx_extension_helper: No application
> 'ReceiveFAX' for extension (macro-faxin, s,
> 12)   
> [Jan 12 18:36:00]   == Spawn extension (macro-faxin, s, 12) exited
> non-zero on 'SIP/gxw-000b'
> Best Regards,
> 
> Mahesh Katta
> *BUZZ**WORKS*Business Services Private Limited
> BANGALORE | CHENNAI | HYDERABAD |MUMBAI| DELHI
> 222, Arunvihar,Sector-28, Noida 201301
> GSM+91.3 45699 | Phone +91.12.0431.0581
> Webhttp://www.buzzworks.com
> 

Hi Mahesh,

I'm running asterisk 1.6.2.21 on Ubuntu 10.4.3 LTS

On a Debian based linux distribution you need the following packets:

libtiff-tools
libtiff4

To receive fax, you'll need the ReceiveFAX Application. You can get it
for personal use directly from digium.com , just search for "FreeFAX for
asterisk"

(http://store.digium.com/productview.php?product_code=804-7)

best regards,
Ruben

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Re: [asterisk-users] FAX Installation in Asterisk

WARNING[6982]: pbx.c:1851 pbx_extension_helper: No application 'ReceiveFAX'
for extension (macro-faxin, s, 12)
[Jan 12 18:36:00]   == Spawn extension (macro-faxin, s, 12) exited non-zero
on 'SIP/gxw-000b'
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
222, Arunvihar,Sector-28, Noida 201301
GSM +91.3 45699 | Phone +91.12.0431.0581
Web http://www.buzzworks.com



On Thu, Jan 12, 2012 at 6:27 PM, mahesh katta wrote:

> Thank you for reply.
> Can I know which version of Asterisk and what supporting applications are
> you currently using.
>
> Best Regards,
>
> Mahesh Katta
> *
>
> *
>
> On Thu, Jan 12, 2012 at 6:15 PM, Ruben Rögels <
> ruben.roeg...@jumping-frog.org> wrote:
>
>> Am 12.01.2012 12:44, schrieb mahesh katta:
>> > Hi,
>> >
>> > Any one give me about FAX in Asterisk.
>> >
>> > PSTN>FXO GATEWAY>ASTERISK-1.4.27(OR)ASTERISK-1.8.X.X
>> >
>> > whenever some one is Fax to PSTN its convert into pdf format
>> >
>> > Help me any links or pdf .. for setup this. ?
>> >
>> >
>> > Best Regards,
>> >
>> > Mahesh Katta
>> > **
>>
>> Hi Mahesh,
>>
>> this is my macro in asterisk to handle fax:
>>
>> [macro-faxin]
>> ; Faxe
>> ; ARG1 = eMail-Adresse
>> exten => s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)})
>> exten => s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNELINFO})
>> ; nur verarbeiten, wenn B-Kanal frei ist
>> exten => s,n,GotoIf($[${BCHANNELINFO} = 2]?hangup:free)
>> exten => s,n(free),NoOp()
>> exten => s,n,Set(TO=${ARG1})
>> exten => s,n,Set(EXT=${MACRO_EXTEN})
>> exten => s,n,Verbose(1,${BOUNDARY} Eingehendes Fax ${CDR(uniqueid)})
>> exten => s,n,Set(FAXFILE=/tmp/fax-${TO}-${CDR(uniqueid)}.tif)
>> exten => s,n,Set(LOCALSTATIONID=jumping frog)
>> exten => s,n,Answer()
>> exten => s,n,Wait(3)
>> exten => s,n,ReceiveFAX(${FAXFILE},d)
>>
>>
>> This is an ugly work-around to handle fax properly becaus I can't catch
>> the hang-up event by the macro itself:
>>
>>
>> ;fax oder kein fax, das ist hier die Frage...
>> exten => h,1,Verbose(${BOUNDARY} ${EXT})
>> exten => h,n,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
>>
>> And this is the bash script to convert tif to pdf and send it via email
>> to my users:
>>
>> #!/bin/bash
>>
>> FAXFILE=$1
>> RECIPIENT=$2
>> SUBJECT="[Fax] Sie haben ein Fax erhalten"
>> BODYSUCCESS=/usr/local/bin/bodysuccess.txt
>> BODYFAILED=/usr/local/bin/bodyfailed.txt
>>
>> PDF=/tmp/fax-`date +"%s"`.pdf
>>
>>
>>
>> tiff2pdf $FAXFILE > $PDF
>>
>> # Konvertierung okay?
>> if [ $? == 0 ]; then
>>
>>mutt -s "$SUBJECT" -a $PDF -- $RECIPIENT < $BODYSUCCESS
>>
>># Hats geklappt?
>>if [ $? == 0 ]; then
>>
>>exit 0
>>
>>else
>>
>>exit 1
>>
>>fi
>>
>>
>>
>> else
>>
>>
>>mutt -s $RECIPIENT < $BODYFAILED
>>
>>exit 1
>> fi
>>
>>
>> I hope this helps!
>>
>> best regards,
>> Ruben
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Exceptionally long voice queue length

2012-01-12 Thread Vik Killa

Asterisk 1.6.1.22

On Thu, Jan 12, 2012 at 2:08 AM, Sammy Govind  wrote:
> which version of Asterisk are you using !. AFAIK this issue has been in
> asterisk for queue calls and I'm not sure if this has ever been resolved
> fully and stabilized. Not binding to Local channel only, I've seen this on
> SIP and IAX channels as well !

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Re: [asterisk-users] FAX Installation in Asterisk

Thank you for reply.
Can I know which version of Asterisk and what supporting applications are
you currently using.

Best Regards,

Mahesh Katta
*

*
On Thu, Jan 12, 2012 at 6:15 PM, Ruben Rögels <
ruben.roeg...@jumping-frog.org> wrote:

> Am 12.01.2012 12:44, schrieb mahesh katta:
> > Hi,
> >
> > Any one give me about FAX in Asterisk.
> >
> > PSTN>FXO GATEWAY>ASTERISK-1.4.27(OR)ASTERISK-1.8.X.X
> >
> > whenever some one is Fax to PSTN its convert into pdf format
> >
> > Help me any links or pdf .. for setup this. ?
> >
> >
> > Best Regards,
> >
> > Mahesh Katta
> > **
>
> Hi Mahesh,
>
> this is my macro in asterisk to handle fax:
>
> [macro-faxin]
> ; Faxe
> ; ARG1 = eMail-Adresse
> exten => s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)})
> exten => s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNELINFO})
> ; nur verarbeiten, wenn B-Kanal frei ist
> exten => s,n,GotoIf($[${BCHANNELINFO} = 2]?hangup:free)
> exten => s,n(free),NoOp()
> exten => s,n,Set(TO=${ARG1})
> exten => s,n,Set(EXT=${MACRO_EXTEN})
> exten => s,n,Verbose(1,${BOUNDARY} Eingehendes Fax ${CDR(uniqueid)})
> exten => s,n,Set(FAXFILE=/tmp/fax-${TO}-${CDR(uniqueid)}.tif)
> exten => s,n,Set(LOCALSTATIONID=jumping frog)
> exten => s,n,Answer()
> exten => s,n,Wait(3)
> exten => s,n,ReceiveFAX(${FAXFILE},d)
>
>
> This is an ugly work-around to handle fax properly becaus I can't catch
> the hang-up event by the macro itself:
>
>
> ;fax oder kein fax, das ist hier die Frage...
> exten => h,1,Verbose(${BOUNDARY} ${EXT})
> exten => h,n,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
>
> And this is the bash script to convert tif to pdf and send it via email
> to my users:
>
> #!/bin/bash
>
> FAXFILE=$1
> RECIPIENT=$2
> SUBJECT="[Fax] Sie haben ein Fax erhalten"
> BODYSUCCESS=/usr/local/bin/bodysuccess.txt
> BODYFAILED=/usr/local/bin/bodyfailed.txt
>
> PDF=/tmp/fax-`date +"%s"`.pdf
>
>
>
> tiff2pdf $FAXFILE > $PDF
>
> # Konvertierung okay?
> if [ $? == 0 ]; then
>
>mutt -s "$SUBJECT" -a $PDF -- $RECIPIENT < $BODYSUCCESS
>
># Hats geklappt?
>if [ $? == 0 ]; then
>
>exit 0
>
>else
>
>exit 1
>
>fi
>
>
>
> else
>
>
>mutt -s $RECIPIENT < $BODYFAILED
>
>exit 1
> fi
>
>
> I hope this helps!
>
> best regards,
> Ruben
>
>
>
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> _
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Re: [asterisk-users] FAX Installation in Asterisk

2012-01-12 Thread Ruben Rögels

Am 12.01.2012 12:44, schrieb mahesh katta:
> Hi,
> 
> Any one give me about FAX in Asterisk.
> 
> PSTN>FXO GATEWAY>ASTERISK-1.4.27(OR)ASTERISK-1.8.X.X
> 
> whenever some one is Fax to PSTN its convert into pdf format
> 
> Help me any links or pdf .. for setup this. ?
>  
> 
> Best Regards,
> 
> Mahesh Katta
> **

Hi Mahesh,

this is my macro in asterisk to handle fax:

[macro-faxin]
; Faxe
; ARG1 = eMail-Adresse
exten => s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)})
exten => s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNELINFO})
; nur verarbeiten, wenn B-Kanal frei ist
exten => s,n,GotoIf($[${BCHANNELINFO} = 2]?hangup:free)
exten => s,n(free),NoOp()
exten => s,n,Set(TO=${ARG1})
exten => s,n,Set(EXT=${MACRO_EXTEN})
exten => s,n,Verbose(1,${BOUNDARY} Eingehendes Fax ${CDR(uniqueid)})
exten => s,n,Set(FAXFILE=/tmp/fax-${TO}-${CDR(uniqueid)}.tif)
exten => s,n,Set(LOCALSTATIONID=jumping frog)
exten => s,n,Answer()
exten => s,n,Wait(3)
exten => s,n,ReceiveFAX(${FAXFILE},d)


This is an ugly work-around to handle fax properly becaus I can't catch
the hang-up event by the macro itself:


;fax oder kein fax, das ist hier die Frage...
exten => h,1,Verbose(${BOUNDARY} ${EXT})
exten => h,n,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})

And this is the bash script to convert tif to pdf and send it via email
to my users:

#!/bin/bash

FAXFILE=$1
RECIPIENT=$2
SUBJECT="[Fax] Sie haben ein Fax erhalten"
BODYSUCCESS=/usr/local/bin/bodysuccess.txt
BODYFAILED=/usr/local/bin/bodyfailed.txt

PDF=/tmp/fax-`date +"%s"`.pdf



tiff2pdf $FAXFILE > $PDF

# Konvertierung okay?
if [ $? == 0 ]; then

mutt -s "$SUBJECT" -a $PDF -- $RECIPIENT < $BODYSUCCESS

# Hats geklappt?
if [ $? == 0 ]; then

exit 0

else

exit 1

fi



else


mutt -s $RECIPIENT < $BODYFAILED

exit 1
fi


I hope this helps!

best regards,
Ruben



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[asterisk-users] Questions on hardware or software-based echo cancellation

Hi,

I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.

I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.

1. I'm correct thinking that it is then impossible to switch from
hardware to software echo can without removing the VPMOCT64 module
itself ?
2. Does the same also apply to HA8 and its VPMOCT032 module ?
3. Are the only options for OSLEC configuration the  echocancel=128 or
echocancel=256 values in chan_dahdi.conf ?
4. How could be compared user experience with oslec/256, mg2/256,
mg2/1024 on a HA8 without hardware module ? Which would you recommend
?

Regards

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Re: [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)

2012-01-12 Thread SIP IMS

2012/1/12 Ishfaq Malik 

> Hi
>
> I'm using 1.8.7.0 with the RealTime architecture.
>
> If a call goes into application Queue and is abandoned by the caller, no
> entry is made in the CDR. Entries are made into the queue log.
>
> This cannot be correct behaviour, all calls should show in the CDR.
>
> Could anyone else try to reproduce this and if others get the same
> thing, I'll raise a bug on it.
>
> Thanks
>
> Ish
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
>
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> _
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[asterisk-users] Call abandoned from queue not showing in CDR (possible bug)

2012-01-12 Thread Ishfaq Malik

Hi

I'm using 1.8.7.0 with the RealTime architecture.

If a call goes into application Queue and is abandoned by the caller, no
entry is made in the CDR. Entries are made into the queue log. 

This cannot be correct behaviour, all calls should show in the CDR.

Could anyone else try to reproduce this and if others get the same
thing, I'll raise a bug on it.

Thanks

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] FAX Installation in Asterisk

Hi,

Any one give me about FAX in Asterisk.

PSTN>FXO GATEWAY>ASTERISK-1.4.27(OR)ASTERISK-1.8.X.X

whenever some one is Fax to PSTN its convert into pdf format

Help me any links or pdf .. for setup this. ?


Best Regards,

Mahesh Katta
**
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Re: [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ? [SOLVED]

2012/1/11, José Pablo Méndez Soto :
> Im using the one that comes with Ubuntu Server 10.10 (0.0.6~pre12-1):
>
> http://packages.ubuntu.com/search?keywords=libspandsp&searchon=names&suite=maverick§ion=all
>
> And having a sweet time with T.38 gateway. Oneiric already offers latest
> pre18.

T.38/T.30 gatewaying can tricky enough to configure so moving this
library version choice out of the equation is a great step forward for
me.

I'll go with pre18, as also suggested elsewhere.

Thanks for sharing this !

>
>
>  *José Pablo Méndez
> *
>
>
> On Wed, Jan 11, 2012 at 12:39 AM, Olivier  wrote:
>
>> Hi,
>>
>> Maybe I missed it while checking it, but which spandsp version is
>> recommended to play with  Asterisk 10 and T.38/T.30 gatewaying ?
>>
>> I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here
>> (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a
>> changelog documenting differences between them.
>> So I prefer to double check ask for recommendations.
>>
>> Regards
>>
>> --
>> _
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>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>

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Re: [asterisk-users] Anyone have a reliable T.38 Solution

I didn't know spandsp could do 14.4k : that's great !
So, now I'm realizing how strange my previous question could resonate.

Thanks for clarifying this.

2012/1/11, Steve Underwood :
> On 01/11/2012 11:16 PM, Olivier wrote:
>> 2012/1/11, Steve Underwood:
>>> On 01/11/2012 03:01 PM, Olivier wrote:
 2012/1/5, Kevin P. Fleming:
> On 01/04/2012 12:25 AM, Matt Darnell wrote:
>> Aloha,
>>
>> We are looking to roll a solution that will have the following network
>> layout:
>>
>> ISDN-PRI<-->Asterisk<-->T.38<-->ATA<-->Fax
>>
>> Does version 1.8 with the Digium fax driver have this capability?  I
>> like 1.8 because it is a long term support version.
>>
>> What ATA's are people using?
>>
>> Any working solutions would be great!
> What you are looking for is T.38 gateway mode (converting between T.30
> over modems on a TDM circuit and T.38 over UDPTL), and the answer is
> no:
> Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it
> is supported using SpanDSP and res_fax_spandsp. It is not yet supported
> by Digium's Fax for Asterisk commercial FAX module.
 Do you have any idea when  Digium's Fax for Asterisk commercial FAX
 module could roughly become supported ?
>>> Are you really desperate to pay for functionality you can get for free?
>> Not yet ;-)))
>> But the increased fax sending speed (14.4 kbs/s says the datasheet but
>> I must be too naive to still read datasheets) may be a feature
>> interesting for some.
>>
>> By the way, which spandsp version would recommend for asterisk 10 ?
>> spandsp-0.0.6pre18.tgz ?
>>
> How is 14.4k an increase? Both spandsp and the Digium modules do 14.4k.
> There is nothing the Digium module does which spandsp does not do, and
> the file handling in spandsp is more flexible.
>
> spandsp-0.0.6pre18.tgz is currently the right version to use?
>
> Steve
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to set callerid in php AGI file.

Phpagi also has predefined method

$agi -> set_callerid("");

Regards,
Zohair Raza



On Thu, Jan 12, 2012 at 1:02 PM, Zohair Raza
wrote:

> Any variable can be set and get from agi
> CDR(clid) is a CDR variable
>
> Regards,
> Zohair Raza
>
>
> On Thu, Jan 12, 2012 at 12:51 PM, virendra bhati wrote:
>
>> How to used it in AGI ? I think it's Dialplan apps.
>>
>>
>> On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza <
>> engineerzuhairr...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Try setting CDR(clid)
>>>
>>> Regards,
>>> Zohair Raza
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati wrote:
>>>
 Hi,
 I am using phpagi for agi scripting. and want to update callerid number
 but didn't get any success. please help me how to update PHPAGI is new for
 me. Below is the code which I write.

 #!/usr/bin/php -q
 >>> set_time_limit(30);
 //require(.phpagi.php.);
 include("phpagi.php");
 $agi = new AGI();

 //answer the call
 $agi-> answer();
 $agi->verbose("--");
 $agi-> exec('Set',"CALLERID(num)=01133200274");

 $ani = $agi->request['agi_callerid'];
 $agi->noop("My CalleID: <<<=".$ani);

 $agi->set_variable("CALLERID(num)","01133200274");
 $ani = $agi->request['agi_callerid'];
 $agi->noop("My CalleID: <<<=".$ani);

 $agi-> exec('Dial',"SIP/00918885268...@sip.trunk.gradwell.com,60,r");
 //$agi-> exec('Dial',"SIP/00918885268942@voipon,60,r");
 ?>

 And CLI>

  == Using SIP RTP CoS mark 5
 -- Executing [101@outbound:1] Answer("SIP/2209-26d3", "") in
 new stack
 -- Executing [101@outbound:2] AGI("SIP/2209-26d3",
 "/home/virendra.bhati/outdial.php") in new stack
 -- Launched AGI Script /home/virendra.bhati/outdial.php
 AGI Tx >> agi_request:
 /home/virendra.bhati/outdial.php
 AGI Tx >> agi_channel: SIP/2209-26d3
 AGI Tx >> agi_language: en
 AGI Tx >> agi_type: SIP
 AGI Tx >> agi_uniqueid: 1326357644.10070
 AGI Tx >> agi_version: 1.6.2.20
 AGI Tx >> agi_callerid: 2209
 AGI Tx >> agi_calleridname: unknown
 AGI Tx >> agi_callingpres: 0
 AGI Tx >> agi_callingani2: 0
 AGI Tx >> agi_callington: 0
 AGI Tx >> agi_callingtns: 0
 AGI Tx >> agi_dnid: 101
 AGI Tx >> agi_rdnis: unknown
 AGI Tx >> agi_context: outbound
 AGI Tx >> agi_extension: 101
 AGI Tx >> agi_priority: 2
 AGI Tx >> agi_enhanced: 0.0
 AGI Tx >> agi_accountcode:
 AGI Tx >> agi_threadid: 1386719552
 AGI Tx >>
 AGI Rx << ANSWER
 AGI Tx >> 200 result=0
 AGI Rx << VERBOSE
 "--" 1
  /home/virendra.bhati/outdial.php:
 --
 AGI Tx >> 200 result=1
 AGI Rx << EXEC Set CALLERID(num)=01133200274
 -- AGI Script Executing Application: (Set) Options: (CALLERID(num)=
 01133200274)
 AGI Tx >> 200 result=0
 AGI Rx << NOOP "My CalleID: <<<=2209"
 AGI Tx >> 200 result=0
 AGI Rx << SET VARIABLE CALLERID(num) "01133200274"
 AGI Tx >> 200 result=1
 AGI Rx << NOOP "My CalleID: <<<=2209"
 AGI Tx >> 200 result=0
 AGI Rx << EXEC Dial SIP/
 00918885268...@sip.trunk.gradwell.com,60,r
 -- AGI Script Executing Application: (Dial) Options: (SIP/
 00918885268...@sip.trunk.gradwell.com,60,r)
   == Using SIP RTP CoS mark 5
> ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
 mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
 -- Called 00918885268...@sip.trunk.gradwell.com
 [Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463
 handle_response_invite: Received response: "Forbidden" from '"
 01133200274" ;tag=as76229e88'
 -- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
   == Everyone is busy/congested at this time (1:0/1/0)
 AGI Tx >> 200 result=0
 -- AGI Script /home/virendra.bhati/outdial.php
 completed, returning 0
 -- Executing [101@outbound:3] Hangup("SIP/2209-26d3", "") in
 new stack

 --

 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

Re: [asterisk-users] how to set callerid in php AGI file.

Any variable can be set and get from agi
CDR(clid) is a CDR variable

Regards,
Zohair Raza


On Thu, Jan 12, 2012 at 12:51 PM, virendra bhati  wrote:

> How to used it in AGI ? I think it's Dialplan apps.
>
>
> On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza  > wrote:
>
>> Hi,
>>
>> Try setting CDR(clid)
>>
>> Regards,
>> Zohair Raza
>>
>>
>>
>>
>>
>> On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati wrote:
>>
>>> Hi,
>>> I am using phpagi for agi scripting. and want to update callerid number
>>> but didn't get any success. please help me how to update PHPAGI is new for
>>> me. Below is the code which I write.
>>>
>>> #!/usr/bin/php -q
>>> >> set_time_limit(30);
>>> //require(.phpagi.php.);
>>> include("phpagi.php");
>>> $agi = new AGI();
>>>
>>> //answer the call
>>> $agi-> answer();
>>> $agi->verbose("--");
>>> $agi-> exec('Set',"CALLERID(num)=01133200274");
>>>
>>> $ani = $agi->request['agi_callerid'];
>>> $agi->noop("My CalleID: <<<=".$ani);
>>>
>>> $agi->set_variable("CALLERID(num)","01133200274");
>>> $ani = $agi->request['agi_callerid'];
>>> $agi->noop("My CalleID: <<<=".$ani);
>>>
>>> $agi-> exec('Dial',"SIP/00918885268...@sip.trunk.gradwell.com,60,r");
>>> //$agi-> exec('Dial',"SIP/00918885268942@voipon,60,r");
>>> ?>
>>>
>>> And CLI>
>>>
>>>  == Using SIP RTP CoS mark 5
>>> -- Executing [101@outbound:1] Answer("SIP/2209-26d3", "") in
>>> new stack
>>> -- Executing [101@outbound:2] AGI("SIP/2209-26d3",
>>> "/home/virendra.bhati/outdial.php") in new stack
>>> -- Launched AGI Script /home/virendra.bhati/outdial.php
>>> AGI Tx >> agi_request:
>>> /home/virendra.bhati/outdial.php
>>> AGI Tx >> agi_channel: SIP/2209-26d3
>>> AGI Tx >> agi_language: en
>>> AGI Tx >> agi_type: SIP
>>> AGI Tx >> agi_uniqueid: 1326357644.10070
>>> AGI Tx >> agi_version: 1.6.2.20
>>> AGI Tx >> agi_callerid: 2209
>>> AGI Tx >> agi_calleridname: unknown
>>> AGI Tx >> agi_callingpres: 0
>>> AGI Tx >> agi_callingani2: 0
>>> AGI Tx >> agi_callington: 0
>>> AGI Tx >> agi_callingtns: 0
>>> AGI Tx >> agi_dnid: 101
>>> AGI Tx >> agi_rdnis: unknown
>>> AGI Tx >> agi_context: outbound
>>> AGI Tx >> agi_extension: 101
>>> AGI Tx >> agi_priority: 2
>>> AGI Tx >> agi_enhanced: 0.0
>>> AGI Tx >> agi_accountcode:
>>> AGI Tx >> agi_threadid: 1386719552
>>> AGI Tx >>
>>> AGI Rx << ANSWER
>>> AGI Tx >> 200 result=0
>>> AGI Rx << VERBOSE
>>> "--" 1
>>>  /home/virendra.bhati/outdial.php:
>>> --
>>> AGI Tx >> 200 result=1
>>> AGI Rx << EXEC Set CALLERID(num)=01133200274
>>> -- AGI Script Executing Application: (Set) Options: (CALLERID(num)=
>>> 01133200274)
>>> AGI Tx >> 200 result=0
>>> AGI Rx << NOOP "My CalleID: <<<=2209"
>>> AGI Tx >> 200 result=0
>>> AGI Rx << SET VARIABLE CALLERID(num) "01133200274"
>>> AGI Tx >> 200 result=1
>>> AGI Rx << NOOP "My CalleID: <<<=2209"
>>> AGI Tx >> 200 result=0
>>> AGI Rx << EXEC Dial SIP/
>>> 00918885268...@sip.trunk.gradwell.com,60,r
>>> -- AGI Script Executing Application: (Dial) Options: (SIP/
>>> 00918885268...@sip.trunk.gradwell.com,60,r)
>>>   == Using SIP RTP CoS mark 5
>>>> ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
>>> mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
>>> -- Called 00918885268...@sip.trunk.gradwell.com
>>> [Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463
>>> handle_response_invite: Received response: "Forbidden" from '"
>>> 01133200274" ;tag=as76229e88'
>>> -- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
>>>   == Everyone is busy/congested at this time (1:0/1/0)
>>> AGI Tx >> 200 result=0
>>> -- AGI Script /home/virendra.bhati/outdial.php
>>> completed, returning 0
>>> -- Executing [101@outbound:3] Hangup("SIP/2209-26d3", "") in
>>> new stack
>>>
>>> --
>>>
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-8885268942
>>> Software Engineer
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- Bandw

Re: [asterisk-users] how to set callerid in php AGI file.

In phpagi

$agi->set_variable("CDR(clid) ")
and to get it
 $agi->get_variable("CDR(clid)")

Regards,
Zohair Raza

www.zuhair.info

*http://ae.linkedin.com/in/zuhairraza**  ***




On Thu, Jan 12, 2012 at 12:51 PM, virendra bhati  wrote:

> How to used it in AGI ? I think it's Dialplan apps.
>
>
> On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza  > wrote:
>
>> Hi,
>>
>> Try setting CDR(clid)
>>
>> Regards,
>> Zohair Raza
>>
>>
>>
>>
>>
>> On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati wrote:
>>
>>> Hi,
>>> I am using phpagi for agi scripting. and want to update callerid number
>>> but didn't get any success. please help me how to update PHPAGI is new for
>>> me. Below is the code which I write.
>>>
>>> #!/usr/bin/php -q
>>> >> set_time_limit(30);
>>> //require(.phpagi.php.);
>>> include("phpagi.php");
>>> $agi = new AGI();
>>>
>>> //answer the call
>>> $agi-> answer();
>>> $agi->verbose("--");
>>> $agi-> exec('Set',"CALLERID(num)=01133200274");
>>>
>>> $ani = $agi->request['agi_callerid'];
>>> $agi->noop("My CalleID: <<<=".$ani);
>>>
>>> $agi->set_variable("CALLERID(num)","01133200274");
>>> $ani = $agi->request['agi_callerid'];
>>> $agi->noop("My CalleID: <<<=".$ani);
>>>
>>> $agi-> exec('Dial',"SIP/00918885268...@sip.trunk.gradwell.com,60,r");
>>> //$agi-> exec('Dial',"SIP/00918885268942@voipon,60,r");
>>> ?>
>>>
>>> And CLI>
>>>
>>>  == Using SIP RTP CoS mark 5
>>> -- Executing [101@outbound:1] Answer("SIP/2209-26d3", "") in
>>> new stack
>>> -- Executing [101@outbound:2] AGI("SIP/2209-26d3",
>>> "/home/virendra.bhati/outdial.php") in new stack
>>> -- Launched AGI Script /home/virendra.bhati/outdial.php
>>> AGI Tx >> agi_request:
>>> /home/virendra.bhati/outdial.php
>>> AGI Tx >> agi_channel: SIP/2209-26d3
>>> AGI Tx >> agi_language: en
>>> AGI Tx >> agi_type: SIP
>>> AGI Tx >> agi_uniqueid: 1326357644.10070
>>> AGI Tx >> agi_version: 1.6.2.20
>>> AGI Tx >> agi_callerid: 2209
>>> AGI Tx >> agi_calleridname: unknown
>>> AGI Tx >> agi_callingpres: 0
>>> AGI Tx >> agi_callingani2: 0
>>> AGI Tx >> agi_callington: 0
>>> AGI Tx >> agi_callingtns: 0
>>> AGI Tx >> agi_dnid: 101
>>> AGI Tx >> agi_rdnis: unknown
>>> AGI Tx >> agi_context: outbound
>>> AGI Tx >> agi_extension: 101
>>> AGI Tx >> agi_priority: 2
>>> AGI Tx >> agi_enhanced: 0.0
>>> AGI Tx >> agi_accountcode:
>>> AGI Tx >> agi_threadid: 1386719552
>>> AGI Tx >>
>>> AGI Rx << ANSWER
>>> AGI Tx >> 200 result=0
>>> AGI Rx << VERBOSE
>>> "--" 1
>>>  /home/virendra.bhati/outdial.php:
>>> --
>>> AGI Tx >> 200 result=1
>>> AGI Rx << EXEC Set CALLERID(num)=01133200274
>>> -- AGI Script Executing Application: (Set) Options: (CALLERID(num)=
>>> 01133200274)
>>> AGI Tx >> 200 result=0
>>> AGI Rx << NOOP "My CalleID: <<<=2209"
>>> AGI Tx >> 200 result=0
>>> AGI Rx << SET VARIABLE CALLERID(num) "01133200274"
>>> AGI Tx >> 200 result=1
>>> AGI Rx << NOOP "My CalleID: <<<=2209"
>>> AGI Tx >> 200 result=0
>>> AGI Rx << EXEC Dial SIP/
>>> 00918885268...@sip.trunk.gradwell.com,60,r
>>> -- AGI Script Executing Application: (Dial) Options: (SIP/
>>> 00918885268...@sip.trunk.gradwell.com,60,r)
>>>   == Using SIP RTP CoS mark 5
>>>> ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
>>> mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
>>> -- Called 00918885268...@sip.trunk.gradwell.com
>>> [Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463
>>> handle_response_invite: Received response: "Forbidden" from '"
>>> 01133200274" ;tag=as76229e88'
>>> -- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
>>>   == Everyone is busy/congested at this time (1:0/1/0)
>>> AGI Tx >> 200 result=0
>>> -- AGI Script /home/virendra.bhati/outdial.php
>>> completed, returning 0
>>> -- Executing [101@outbound:3] Hangup("SIP/2209-26d3", "") in
>>> new stack
>>>
>>> --
>>>
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-8885268942
>>> Software Engineer
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engin

Re: [asterisk-users] how to set callerid in php AGI file.

2012-01-12 Thread virendra bhati

How to used it in AGI ? I think it's Dialplan apps.

On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza
wrote:

> Hi,
>
> Try setting CDR(clid)
>
> Regards,
> Zohair Raza
>
>
>
>
>
> On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati wrote:
>
>> Hi,
>> I am using phpagi for agi scripting. and want to update callerid number
>> but didn't get any success. please help me how to update PHPAGI is new for
>> me. Below is the code which I write.
>>
>> #!/usr/bin/php -q
>> > set_time_limit(30);
>> //require(.phpagi.php.);
>> include("phpagi.php");
>> $agi = new AGI();
>>
>> //answer the call
>> $agi-> answer();
>> $agi->verbose("--");
>> $agi-> exec('Set',"CALLERID(num)=01133200274");
>>
>> $ani = $agi->request['agi_callerid'];
>> $agi->noop("My CalleID: <<<=".$ani);
>>
>> $agi->set_variable("CALLERID(num)","01133200274");
>> $ani = $agi->request['agi_callerid'];
>> $agi->noop("My CalleID: <<<=".$ani);
>>
>> $agi-> exec('Dial',"SIP/00918885268...@sip.trunk.gradwell.com,60,r");
>> //$agi-> exec('Dial',"SIP/00918885268942@voipon,60,r");
>> ?>
>>
>> And CLI>
>>
>>  == Using SIP RTP CoS mark 5
>> -- Executing [101@outbound:1] Answer("SIP/2209-26d3", "") in new
>> stack
>> -- Executing [101@outbound:2] AGI("SIP/2209-26d3",
>> "/home/virendra.bhati/outdial.php") in new stack
>> -- Launched AGI Script /home/virendra.bhati/outdial.php
>> AGI Tx >> agi_request: /home/virendra.bhati/outdial.php
>> AGI Tx >> agi_channel: SIP/2209-26d3
>> AGI Tx >> agi_language: en
>> AGI Tx >> agi_type: SIP
>> AGI Tx >> agi_uniqueid: 1326357644.10070
>> AGI Tx >> agi_version: 1.6.2.20
>> AGI Tx >> agi_callerid: 2209
>> AGI Tx >> agi_calleridname: unknown
>> AGI Tx >> agi_callingpres: 0
>> AGI Tx >> agi_callingani2: 0
>> AGI Tx >> agi_callington: 0
>> AGI Tx >> agi_callingtns: 0
>> AGI Tx >> agi_dnid: 101
>> AGI Tx >> agi_rdnis: unknown
>> AGI Tx >> agi_context: outbound
>> AGI Tx >> agi_extension: 101
>> AGI Tx >> agi_priority: 2
>> AGI Tx >> agi_enhanced: 0.0
>> AGI Tx >> agi_accountcode:
>> AGI Tx >> agi_threadid: 1386719552
>> AGI Tx >>
>> AGI Rx << ANSWER
>> AGI Tx >> 200 result=0
>> AGI Rx << VERBOSE
>> "--" 1
>>  /home/virendra.bhati/outdial.php:
>> --
>> AGI Tx >> 200 result=1
>> AGI Rx << EXEC Set CALLERID(num)=01133200274
>> -- AGI Script Executing Application: (Set) Options: (CALLERID(num)=
>> 01133200274)
>> AGI Tx >> 200 result=0
>> AGI Rx << NOOP "My CalleID: <<<=2209"
>> AGI Tx >> 200 result=0
>> AGI Rx << SET VARIABLE CALLERID(num) "01133200274"
>> AGI Tx >> 200 result=1
>> AGI Rx << NOOP "My CalleID: <<<=2209"
>> AGI Tx >> 200 result=0
>> AGI Rx << EXEC Dial SIP/
>> 00918885268...@sip.trunk.gradwell.com,60,r
>> -- AGI Script Executing Application: (Dial) Options: (SIP/
>> 00918885268...@sip.trunk.gradwell.com,60,r)
>>   == Using SIP RTP CoS mark 5
>>> ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
>> mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
>> -- Called 00918885268...@sip.trunk.gradwell.com
>> [Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463
>> handle_response_invite: Received response: "Forbidden" from '"01133200274"
>> ;tag=as76229e88'
>> -- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
>>   == Everyone is busy/congested at this time (1:0/1/0)
>> AGI Tx >> 200 result=0
>> -- AGI Script /home/virendra.bhati/outdial.php
>> completed, returning 0
>> -- Executing [101@outbound:3] Hangup("SIP/2209-26d3", "") in new
>> stack
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-8885268942
>> Software Engineer
>>
>>
>> --
>> _
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Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] how to set callerid in php AGI file.

Hi,

Try setting CDR(clid)

Regards,
Zohair Raza





On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati  wrote:

> Hi,
> I am using phpagi for agi scripting. and want to update callerid number
> but didn't get any success. please help me how to update PHPAGI is new for
> me. Below is the code which I write.
>
> #!/usr/bin/php -q
>  set_time_limit(30);
> //require(.phpagi.php.);
> include("phpagi.php");
> $agi = new AGI();
>
> //answer the call
> $agi-> answer();
> $agi->verbose("--");
> $agi-> exec('Set',"CALLERID(num)=01133200274");
>
> $ani = $agi->request['agi_callerid'];
> $agi->noop("My CalleID: <<<=".$ani);
>
> $agi->set_variable("CALLERID(num)","01133200274");
> $ani = $agi->request['agi_callerid'];
> $agi->noop("My CalleID: <<<=".$ani);
>
> $agi-> exec('Dial',"SIP/00918885268...@sip.trunk.gradwell.com,60,r");
> //$agi-> exec('Dial',"SIP/00918885268942@voipon,60,r");
> ?>
>
> And CLI>
>
>  == Using SIP RTP CoS mark 5
> -- Executing [101@outbound:1] Answer("SIP/2209-26d3", "") in new
> stack
> -- Executing [101@outbound:2] AGI("SIP/2209-26d3",
> "/home/virendra.bhati/outdial.php") in new stack
> -- Launched AGI Script /home/virendra.bhati/outdial.php
> AGI Tx >> agi_request: /home/virendra.bhati/outdial.php
> AGI Tx >> agi_channel: SIP/2209-26d3
> AGI Tx >> agi_language: en
> AGI Tx >> agi_type: SIP
> AGI Tx >> agi_uniqueid: 1326357644.10070
> AGI Tx >> agi_version: 1.6.2.20
> AGI Tx >> agi_callerid: 2209
> AGI Tx >> agi_calleridname: unknown
> AGI Tx >> agi_callingpres: 0
> AGI Tx >> agi_callingani2: 0
> AGI Tx >> agi_callington: 0
> AGI Tx >> agi_callingtns: 0
> AGI Tx >> agi_dnid: 101
> AGI Tx >> agi_rdnis: unknown
> AGI Tx >> agi_context: outbound
> AGI Tx >> agi_extension: 101
> AGI Tx >> agi_priority: 2
> AGI Tx >> agi_enhanced: 0.0
> AGI Tx >> agi_accountcode:
> AGI Tx >> agi_threadid: 1386719552
> AGI Tx >>
> AGI Rx << ANSWER
> AGI Tx >> 200 result=0
> AGI Rx << VERBOSE
> "--" 1
>  /home/virendra.bhati/outdial.php:
> --
> AGI Tx >> 200 result=1
> AGI Rx << EXEC Set CALLERID(num)=01133200274
> -- AGI Script Executing Application: (Set) Options:
> (CALLERID(num)=01133200274)
> AGI Tx >> 200 result=0
> AGI Rx << NOOP "My CalleID: <<<=2209"
> AGI Tx >> 200 result=0
> AGI Rx << SET VARIABLE CALLERID(num) "01133200274"
> AGI Tx >> 200 result=1
> AGI Rx << NOOP "My CalleID: <<<=2209"
> AGI Tx >> 200 result=0
> AGI Rx << EXEC Dial SIP/
> 00918885268...@sip.trunk.gradwell.com,60,r
> -- AGI Script Executing Application: (Dial) Options: (SIP/
> 00918885268...@sip.trunk.gradwell.com,60,r)
>   == Using SIP RTP CoS mark 5
>> ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
> mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
> -- Called 00918885268...@sip.trunk.gradwell.com
> [Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463 handle_response_invite:
> Received response: "Forbidden" from '"01133200274" <
> sip:01133200274@10.10.10.181>;tag=as76229e88'
> -- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
> AGI Tx >> 200 result=0
> -- AGI Script /home/virendra.bhati/outdial.php
> completed, returning 0
> -- Executing [101@outbound:3] Hangup("SIP/2209-26d3", "") in new
> stack
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] how to set callerid in php AGI file.

2012-01-12 Thread virendra bhati

Hi,
I am using phpagi for agi scripting. and want to update callerid number but
didn't get any success. please help me how to update PHPAGI is new for me.
Below is the code which I write.

#!/usr/bin/php -q
 answer();
$agi->verbose("--");
$agi-> exec('Set',"CALLERID(num)=01133200274");

$ani = $agi->request['agi_callerid'];
$agi->noop("My CalleID: <<<=".$ani);

$agi->set_variable("CALLERID(num)","01133200274");
$ani = $agi->request['agi_callerid'];
$agi->noop("My CalleID: <<<=".$ani);

$agi-> exec('Dial',"SIP/00918885268...@sip.trunk.gradwell.com,60,r");
//$agi-> exec('Dial',"SIP/00918885268942@voipon,60,r");
?>

And CLI>

 == Using SIP RTP CoS mark 5
-- Executing [101@outbound:1] Answer("SIP/2209-26d3", "") in new
stack
-- Executing [101@outbound:2] AGI("SIP/2209-26d3",
"/home/virendra.bhati/outdial.php") in new stack
-- Launched AGI Script /home/virendra.bhati/outdial.php
AGI Tx >> agi_request: /home/virendra.bhati/outdial.php
AGI Tx >> agi_channel: SIP/2209-26d3
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1326357644.10070
AGI Tx >> agi_version: 1.6.2.20
AGI Tx >> agi_callerid: 2209
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 101
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: outbound
AGI Tx >> agi_extension: 101
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> agi_threadid: 1386719552
AGI Tx >>
AGI Rx << ANSWER
AGI Tx >> 200 result=0
AGI Rx << VERBOSE
"--" 1
 /home/virendra.bhati/outdial.php:
--
AGI Tx >> 200 result=1
AGI Rx << EXEC Set CALLERID(num)=01133200274
-- AGI Script Executing Application: (Set) Options:
(CALLERID(num)=01133200274)
AGI Tx >> 200 result=0
AGI Rx << NOOP "My CalleID: <<<=2209"
AGI Tx >> 200 result=0
AGI Rx << SET VARIABLE CALLERID(num) "01133200274"
AGI Tx >> 200 result=1
AGI Rx << NOOP "My CalleID: <<<=2209"
AGI Tx >> 200 result=0
AGI Rx << EXEC Dial SIP/
00918885268...@sip.trunk.gradwell.com,60,r
-- AGI Script Executing Application: (Dial) Options: (SIP/
00918885268...@sip.trunk.gradwell.com,60,r)
  == Using SIP RTP CoS mark 5
   > ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com'
mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060
-- Called 00918885268...@sip.trunk.gradwell.com
[Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463 handle_response_invite:
Received response: "Forbidden" from '"01133200274" <
sip:01133200274@10.10.10.181>;tag=as76229e88'
-- SIP/sip.trunk.gradwell.com-26d4 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
AGI Tx >> 200 result=0
-- AGI Script /home/virendra.bhati/outdial.php
completed, returning 0
-- Executing [101@outbound:3] Hangup("SIP/2209-26d3", "") in new
stack

-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
--
_
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Re: [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]

2012/1/11, A J Stiles :
> On Wednesday 11 January 2012, Olivier wrote:
>> At the time I first wrote my question, libcurl4-openssl-dev was
>> missing from my system so func_CURL was not available, which lead me
>> check with wiki.asterisk.org.
>
> It's *always* a -dev  (or -devel if you're into RPMs)  package missing.
> Always!
>
> Frankly, why distributions still insist to separate out "development" files
> in
> 2012 is a mystery to me.  Ubuntu especially have *no* excuse; user-
> friendliness is supposed to be their USP, and compiling a package from
> Source
> Code is something everyone has to do at some stage.
>
>> Now, I added the missing library and I can see CURL function available.
>> I will open a ticket to let concerned people know about the missing
>> entry in wiki.asterisk.org
>
> Isn't the point of a Wiki, so that anybody can edit it without raising a
> support ticket?
Yes and no: you're right pointing a wiki let people correct things but
I think in this specific topic, function list is automatically
extracted from source code, so a missing entry show either a bug in
source code or in the script extracting it.

>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?