Re: [asterisk-users] Asterisk : CDR Analyzer Updated

[Mitul Limbani]

Hello Doug,

Maybe you can have it uploaded on GitHub.com as a repository ?
With a README.md file on how to install it for PHP7 ?

Regards,
Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422



On Mon, May 25, 2020 at 3:17 PM Doug Lytle  wrote:

> Everybody,
>
> I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a
> dozen years, it was easy to configure and didn't requite installing
> 'connectors' on anything or adding tables on the DB server.
>
> It's based off of PHP5 and the only reason I still keep around a Debian 7
> system, since it won't work with the newer PHP7.
>
> A friend of mine is learning PHP7 and offered to update Asterisk Stats to
> work with the PHP7 as a learning experience.
>
> I've currently got the updated Asterisk Stats running on Debian 10
> (Buster) without issues.
>
> Anybody wanting a copy, just reply to this email and I'll provide the
> updated archived install.
>
> Doug
>
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>
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Re: [asterisk-users] Using g729 now that patents have expired

[Mitul Limbani]

AFAIK g729 patent is expiring sometime in 2019-2020.

Mitul Limbani

On Feb 8, 2017 5:02 AM, "Victor Villarreal" <mefhigos...@gmail.com> wrote:

> Hi Steve,
>
> I understand your question and your point, but I use the g729 codec from
> the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13
> without a single problem.
>
> So, sory but I don't share your phrase "from a lesser know web site".
>
> About your question, I did not known that the patent has expired, so I
> expect and answer just like you.
>
> Cheers.
>
> El 7/2/2017 19:18, "Steve Edwards" <asterisk@sedwards.com> escribió:
>
>> On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards <asterisk@sedwards.com>
>>> wrote:
>>>
>>
>>   Now that the g729 patents have expired, how do we use g729 in
>>>  Asterisk?
>>>
>>>   Will Digium be releasing a g729 codec for 'free' use or do we
>>>   download the 'free' codec off the Internet now that we can use it
>>>   without moral or legal restrictions?
>>>
>>
>> On Tue, 7 Feb 2017, Carlos Rojas wrote:
>>
>> You can uses:
>>>
>>> http://asterisk.hosting.lv/
>>>
>>
>> I'm hoping Digium will do something so we can have an 'out of the box'
>> experience rather than downloading code from a lesser known web site.
>>
>> --
>> Thanks in advance,
>> -
>> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867
>> <+1%20760-468-3867> PST
>> https://www.linkedin.com/in/steve-edwards-4244281
>>
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>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

[Mitul Limbani]

If you want to use dahdi dummy driver inside asterisk for timer then this
is possible with openvz based container virtualization.

We have tested vicidial in this mode for 5-10 agents and it worked well.

Mitul Limbani
On Apr 8, 2016 8:52 AM, "Pete Mundy" <p...@fiberphone.co.nz> wrote:

> List,
>
> Might as well throw my hat in the ring!
>
> I can't say it's the 'best' way to do it, but I've been running Asterisk
> VMs inside the free 'VirtualBox' software for many years with nill issues
> (well, nill related to the hypervisor environment itself anyway!).
>
> https://www.virtualbox.org
>
> Pete
>
> On 8/04/2016, at 2:09 pm, Carlos Rojas <crt.ro...@gmail.com> wrote:
>
> I have tried with xen and kvm both are working fine.
>
> On Wed, Apr 6, 2016 at 3:44 PM, Loic Chabert <loic.chab...@voxity.fr>
> wrote:
>
>> Hello,
>>
>> Work well with kvm and centos 7.
>> Some ajustements has to be made with systemd.
>>
>> I'm using it in production since 1.5 year now, no issue to report.
>>
>> Regards.
>> Le 6 avr. 2016 21:13, "Yves" <yba...@dm2it.com> a écrit :
>>
>>
>> Le 06/04/2016 18:12, Markos Vakondios a écrit :
>>
>> Good evening,
>>
>> My English is limited but if I can help.
>>
>> We install Asterisk Version 13.1 on VmWare with Debian 8.2, no
>> problem since June 2015, currently I have tested on Unbutu 14.04 but problem
>> with network-manager (problem of stability with Asterisk 1.8.32 and
>> difficulty with routing network-manager).
>>
>> I also installed Asterisk on KVM (Debian 8.2) no problem (but not test
>> with dahdi) without particular problem.
>>
>> here is my little opinion
>>
>> Hello everyone
>>
>> Proxmox and KVM on Ubuntu
>>
>> On Wednesday, 6 April 2016, Ryan, Travis < <ry...@oscarwinski.com>
>> ry...@oscarwinski.com> wrote:
>>
>>> What is the best virtual server tech (and most stable, etc) to use for a
>>> asterisk virtual hosting environment?
>>>
>>>
>>> I have a client that wants to do virtual hosting of Asterisk (only SIP
>>> or IAX, no PRI, etc) and I’m wondering if Xen or something else would be
>>> best? We’d like to stay away from the costs of VMWare if possible.
>>>
>>>
>>
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Re: [asterisk-users] Asterisk Development Company in India

[Mitul Limbani]

Lol Jeff !!
On Mar 31, 2016 9:32 PM, "Jeff LaCoursiere"  wrote:

>
> And punctuation and grammar skills have we too!  Our english be VERY good
>
> On 03/31/2016 02:20 AM, ankur verma wrote:
>
>> Have you ever heard of Asterisk Development.There are only few companies
>> in
>> India which are providing this service and "Anticlock Technologies is one
>> of
>> them.it is dealing in this field from long time and We provide client
>> satisfaction, full support and long term services
>> .
>>
>>
>>
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Re: [asterisk-users] 1000 analogue lines with asterisk

[Mitul Limbani]

Use Sangoma 50 port FXS
On Feb 17, 2016 12:42 PM, "Goke Aruna" <gok...@gmail.com> wrote:

> Thanks Mitul,
> The server spec is okay but I need information on the fxs hardware to use.
> Regards
>
> On Wed, Feb 17, 2016 at 8:07 AM, Mitul Limbani <mi...@enterux.in> wrote:
>
>> Quad core Xeon with 4GB ram
>> On Feb 17, 2016 12:32 PM, "Goke Aruna" <gok...@gmail.com> wrote:
>>
>>> Hello all,
>>> Can someone recommend what hardware to use for a 1000 analogue line
>>> capacity asterisk PABX?
>>>
>>> Regards
>>>
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>
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Re: [asterisk-users] 1000 analogue lines with asterisk

[Mitul Limbani]

Quad core Xeon with 4GB ram
On Feb 17, 2016 12:32 PM, "Goke Aruna"  wrote:

> Hello all,
> Can someone recommend what hardware to use for a 1000 analogue line
> capacity asterisk PABX?
>
> Regards
>
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Re: [asterisk-users] No sound with internal calls depending on which phones

[Mitul Limbani]

You might have to disable srtp negotiations inside the phone web ui
options.

Mitul
On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" 
wrote:

> Dear all,
>
> I have a very strange problem :
>
>- external calls work perfectly,
>- internal calls between some phones too,
>- but internal call between two similar phones don't work !!! (Snom
>710)
>
> When we have sound, there are no errors in asterisk. When we do not have
> sound, there is the following error :
>
>- [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
>module loaded, can't setup SRTP session.
>
> This is a working internal call :
>
>   == Using SIP RTP CoS mark 5
> -- Executing [301@local:1] Dial("SIP/dbucher-", "SIP/phone1")
> in new stack
>   == Using SIP RTP CoS mark 5
> -- Called phone1
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 answered SIP/dbucher-
> -- Remotely bridging SIP/dbucher- and SIP/phone1-0001
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Got  RTP packet from192.168.128.99:49646 (type 126, seq 031575, ts
> 01, len 00)
> [Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read:
> Unknown RTP codec 126 received from '192.168.128.99:49646'
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
>   == Spawn extension (local, 301, 1) exited non-zero on
> 'SIP/dbucher-'
>
> This is a non-working call :
>
>   == Using SIP RTP CoS mark 5
> [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
> module loaded, can't setup SRTP session.
> -- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002",
> "SIP/phone1") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called phone1
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
> -- Remotely bridging SIP/hsolutionspf5-0002 and SIP/phone1-0003
> Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
>   == Spawn extension (local, 301, 1) exited non-zero on
> 'SIP/hsolutionspf5-0002'
>
> I tried many options to disable SRTP but without success :
>
>- canreinvite = no
>- canreinvite = nonat
>- srtpcapable=no
>- encryption=no
>- directmedia=nonat
>- ...or noload => res_srtp.so in modules.conf
>
>
> Any help would be GREATLY appreciated !
>
> Denis
>
> P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)
>
>
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Re: [asterisk-users] Fwd: Sublime Text License Key

[Mitul Limbani]

The company making sublime text gets few thousands of dollars of notional
loss :)
On Oct 5, 2015 8:45 PM, "Steve Howes"  wrote:

> Wonder what happens when an entire mailing list tries to use that key?...
>
> On 05/10/15 15:28, Optical Phoenix wrote:
>
> -- Forwarded message --
> From: *Sublime HQ Pty Ltd* 
> Date: Wednesday, July 25, 2012
> Subject: Sublime Text License Key
> To: "opticalphoe...@gmail.com" 
>
>
> Hello,
>
> Thanks for purchasing a copy of Sublime Text! Your license key is:
>
> - BEGIN LICENSE -
> Dennis Wright Jr
> Single User License
> EA7E-819939
> 356F68A3 BDE42447 A0B7E2C4 9429E490
> 1760A71B C59AF641 94066F0A 04146120
> 6F5FC041 A95B5175 139BB680 4EB40EFD
> C50C4829 806BCC12 E2C80B94 77474B29
> D1224F42 F916634C 68CE1BBB 96F1D6D0
> EA547ED4 2E695093 CC474A9B 755D3E9E
> 00CAF5FB 77AA4C22 12FC089C 17A0B891
> 61DDD391 808E58EE 2F9AA80E B04E344A
> -- END LICENSE --
>
>
> Entering the license details:
>
> 1. Open Sublime Text, and select Help/Enter License from the menu.
> 2. Copy the license above (including the BEGIN LICENSE and END LICENSE
> lines) and paste them into the license box.
> 3. Press the Use License button and Sublime Text will enter into licensed
> mode.
>
> Regards,
>
> Jon
> SUBLIME HQ PTY LTD
>
>
>
>
>
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Re: [asterisk-users] Tdm4010p 4 port card

[Mitul Limbani]

Try using older Asterisk version (1.8.x) and older dahdi (2.6.x)

It should work then.

Mitul Limbani
On 10-Jul-2015 9:07 PM, Tom Judge tvju...@gmail.com wrote:



 Hi running asterisk 13.x and dahdi-linux-complete-2.10.2+2.10.2. It looks
 like the kernel drivers lod but in asterisk console dahdi show anything not
 working. Trina to use a TDM410P pci card. Is this just too old and extinct
 card?
 Any suggestions gratefully apprecuated. We are a small non profit based on
 FOSS education. Learn Ubuntu Org inc. Low budget no budget any help greatly
 appreciated.

 Sent from my T-Mobile 4G LTE Device

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Re: [asterisk-users] Product CDR/Queue/Meetme

[Mitul Limbani]

Hey Helvio,

Would like to check it out as well.

Do email me,

Mitul
On 22-Jun-2015 9:05 AM, Helvio Junior helvio.lis...@gmail.com wrote:

  Gentleman,

 Moderators, i don't know if this topic if OFF-Topic, if yes, please tell
 me.

 I had some difficult looking for a Asterisk software that provide me some
 functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So
 i decided to develop than.

 In a few weeks i'll deploy a Beta version of this software and i'd like to
 know if is somebody available to try this beta and free version?

 If you don't want to try this version but would like to see/suggest any
 feature in this software, let me know.

 Forecast functions to Beta Version:

- Realtime view for:
   - Queues;
   - Peers (Similar as BLF);
   - Trunk calls/utilization;
- MeetMe
   - Create, modify, delete and schedule;
   - Real time view of members;
   - Delete members;
   - Mute/Unmute;
   - Send Invite by e-mail (with .VCS file)
- Dialer
   - Create dialer (by campaign with contacts)
- Monitoring of campaig, calls, and status;
   - Time control to retry failed call
- Control of day time to call (commercial time, full time, etc...)
- Charts and reports:
   - Trunk utilization;
   - CDR;
   - Queues (Most common reports and charts, distributions, times,
   etc...)
   - Export to Excel Spreadsheet and PDF File
- Report Scheduler
- Much more...
- REST API for 100% of functionalities;
 - Admin and User Console 100% Web HTML5;
- Developed in Windows with C#;
- Integrate with Asterisk using AMI only;
- Allow manage many Asterisk that you want using same instance of this
software (One software and one installation);


 Obs.: I'll provide a Full License for everybody that help me trying the
 Beta version.

 --

 Att,
 Hélvio Junior
 SafeId - Gestão de identidades e Acessos
 +55 41 | 9893-2694, single-sign-on.com.brhelvio.jun...@safetrend.com.br

  --

 Att,
 Hélvio Junior
 SafeId - Gestão de identidades e Acessos
 +55 41 | 9893-2694, single-sign-on.com.brhelvio.jun...@safetrend.com.br


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Re: [asterisk-users] German sounds on Asterisk

[Mitul Limbani]

http://lmgtfy.com/?q=say.conf+asterisk+german+digits
On 14-Jun-2015 1:06 PM, Luca Bertoncello lucab...@lucabert.de wrote:

 Hi again

 I'd like to configured my Asterisk to use german sounds for the
 Say-commands...

 I installed the sounds-files and I tried them with
 Playback(de/demo-echodone) and it works.

 Now I tried to add an extension to say the current time:

 exten = 24,1,Verbose(2,Time asked by ${CALLERID(num)})
 Exten = 24,n,Set(CHANNEL(language)=de)
 Exten = 24,n,SayUnixTime()
 Exten = 24,n,Hangup

 But if I call the 24, it says the time in English...
 On the CLI I see:

 -- Executing [24@default:2] Set(SIP/004935-0003,
 CHANNEL(language)=de) in new stack
 -- Executing [24@default:3]
 SayUnixTime(SIP/004935-0003, ) in new stack
 -- SIP/004935-0003 Playing 'digits/day-0.gsm' (language
 'de')
 -- SIP/004935-0003 Playing 'digits/h-14.gsm' (language
 'de')
 ...

 So, it seems, it would use the German sounds, but it doesn't...
 Has someone an explanation why it works so?

 Thanks
 Luca Bertoncello
 (lucab...@lucabert.de)

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Re: [asterisk-users] asterisk google contacts

[Mitul Limbani]

In that case you would have to take some professional help.

Mitul

On Thursday, June 11, 2015, tux john a...@null.net wrote:

 I am afraid i do not know how to write that.




 *Sent:* Thursday, June 11, 2015 at 2:05 PM
 *From:* A J Stiles asterisk_l...@earthshod.co.uk
 javascript:_e(%7B%7D,'cvml','asterisk_l...@earthshod.co.uk');
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 javascript:_e(%7B%7D,'cvml','asterisk-users@lists.digium.com');
 *Subject:* Re: [asterisk-users] asterisk  google contacts
 On Thursday 11 Jun 2015, tux john wrote:
  Hello everyone. i am running an asterisk server and i would like to have
  the contacts from google. so every inbound call with fetch the caller ID
  from google contacts and present it to my screen.

 This is really three problems, as follows:

 (1) Accessing the Google Contacts API to retrieve someone's details based
 on
 their phone number.
 (2) Passing the incoming caller's number to an AGI script.
 (3) Displaying the details retrieved from Google on your screen.


 Presuming you already know how to write a program to look up a Google
 contact's details from their phone number, you just need to turn that into
 an
 AGI script. Then, in your dialplan, pass the incoming number to that
 script.

 If you want Asterisk itself to have anything to do with the Google data,
 you
 will have to return them by setting channel variables within the script. Or
 if you are going to use some external means to pass the data to the user,
 then
 you can have your script fork itself, detach and return straight away.


 --
 AJS

 Note: Originating address only accepts e-mail from list! If replying off-
 list, change address to asterisk1list at earthshod dot co dot uk .

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Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] Am I cracked?

[Mitul Limbani]

As a practice, by default all the extensions you expose on the allowguest
mode should lead inbound to your asterisk and should never pick any
outbound trunk and dial out.

Your best option is to remove all outbound extensions from the default
context, move them to default2 and set default extensions as honeypot to
play monkeys tts wave file or reject the call.

Mitul Limbani
 On 09-Jun-2015 2:05 AM, D'Arcy J.M. Cain da...@vex.net wrote:

 On Mon, 8 Jun 2015 22:24:33 +0200
 Luca Bertoncello lucab...@lucabert.de wrote:
  Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
   Basically, they are hoping that you are running the equivalent of a
   mail server open relay. They are trying to use you to dial out to
   another number. You don't want to pay for these calls.
 
  Of course, but how can I test, if I am an open relay?

 If you don't know how to do this I suggest that you shut down your
 Asterisk server until you find out.  Using your cell phone while you
 get it straight could save you some serious coin.

   Not sure what trunk pbxluca is, but if that is an outbound trunk,
   then this is very bad. The only reason it would fail then is if
   they have the
 
  This is one of my outbound trunk...

 Very, very bad then.

  On a Mail-Server I'd restrict outgoing calls to authenticated users.
  I was sure, that Asterisk already do that, but I'm not sure anymore...
  How can I restrict it?

 You need to make sure that only registered phones can connect to your
 outbound trunks.  Read the docs or hire someone but don't wait.  Shut
 down now, especially since this information is now on a public list.  I
 am sure that most people here are just looking out for you but it only
 takes one black hat.

 --
 D'Arcy J.M. Cain
 System Administrator, Vex.Net
 http://www.Vex.Net/ IM:da...@vex.net
 VoIP: sip:da...@vex.net

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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

[Mitul Limbani]

Read the config.log file to know for which dependency it failed.
On 13-Apr-2015 2:08 PM, akhilesh chand omakhileshch...@gmail.com wrote:

 yes I called

 On Mon, Apr 13, 2015 at 1:27 PM, jg webaccounts...@jgoettgens.de wrote:



 I'm not able to install asterisk whenever I hit make command I get below
 error:

 make[1]: *** No rule to make target `../main/modules.link', needed by
 `asterisk'.  Stop.
 make: *** [main] Error 2


 Just guessing. Did you call ./configure?

 jg

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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

[Mitul Limbani]

Did you do make menuselect and saved the options ?

Mitul
On 13-Apr-2015 5:43 PM, ajahar mohd azhar5...@gmail.com wrote:

 Hi Akhilesh,

 Here is another fix,

 getting the error, that: make[1]: *** No rule to make target
 `../main/modules.link’, needed by `asterisk’. Stop. make: *** [main] Error
 2 when compile asterisk

 To get around this, just delete following line in file makeopts.embed_rules

 EMBED_LDSCRIPTS+=../main/modules.link

 Source: http://showmyroutes.com/wordpress/?p=500

 Sincerely,

 M azhar

 http://www.nicacresults.com

 On Mon, Apr 13, 2015 at 1:10 PM, akhilesh chand omakhileshch...@gmail.com
  wrote:

 Hi folks,


 I'm not able to install asterisk whenever I hit make command I get below
 error:

 make[1]: *** No rule to make target `../main/modules.link', needed by
 `asterisk'.  Stop.
 make: *** [main] Error 2


 Regards
 Akhilesh

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Re: [asterisk-users] OpenVZ with asterisk 13

[Mitul Limbani]

Show him this freaking thread, or else ask him to prove it otherwise.

We all here have decades of exp dealing with asterisk.

Mitul
On 07-Apr-2015 7:27 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com wrote:

 Dear Mitul,

 I already told my boss about it, I really want a single box, no virtual,
 but my boss insist.
 He said that openvz use less resource then KVM (or other virtual for
 cloud).
 I really need a solid analysis to argue with him.

 On the other hand, dahdi   cannot be installed in openvz virtual server.

 I dont have any experience with openvz at all.

 Thx,

 On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com
 wrote:

 Dear all,

 Is anyone has experience making Asterisk server with virtual server
 OPEN-VZ (in proxmox 3.4 box) ?

 My boss want to build a production server with it, and it will have +/-
 300 sip user (concurrent call maybe  150 call)

 Is it good to go, or not ?

 I really hope someone who have experience with it  willing to share with
 me...

 Thanks in advance...


 Best Regards,


 Ikka - Jakarta, Indonesia





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Re: [asterisk-users] OpenVZ with asterisk 13

[Mitul Limbani]

Why not use just one single box and create 300 sip clients having 150 odd
con calls. OpenVZ might not be a good idea for this sort of volume.

Mitul
On 07-Apr-2015 7:12 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com wrote:

 Dear all,

 Is anyone has experience making Asterisk server with virtual server
 OPEN-VZ (in proxmox 3.4 box) ?

 My boss want to build a production server with it, and it will have +/-
 300 sip user (concurrent call maybe  150 call)

 Is it good to go, or not ?

 I really hope someone who have experience with it  willing to share with
 me...

 Thanks in advance...


 Best Regards,


 Ikka - Jakarta, Indonesia



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Re: [asterisk-users] OpenVZ with asterisk 13

[Mitul Limbani]

PBX =! CC my friend.

150 Conc Calls for CC agent is going to be far more expensive then running
200 extn PBX doing hardly 20 Conc Calls.

Load is way too diff.
On 07-Apr-2015 8:18 PM, Vinicius Fontes vinic...@aittelecom.com.br
wrote:

 I have several large customers (200+ extensions) running on vSphere
 without issue. Not sure about OpenVZ, thought.

 2015-04-07 11:36 GMT-03:00 Mitul Limbani mi...@enterux.in:

 Show him this freaking thread, or else ask him to prove it otherwise.

 We all here have decades of exp dealing with asterisk.

 Mitul
 On 07-Apr-2015 7:27 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com wrote:

 Dear Mitul,

 I already told my boss about it, I really want a single box, no virtual,
 but my boss insist.
 He said that openvz use less resource then KVM (or other virtual for
 cloud).
 I really need a solid analysis to argue with him.

 On the other hand, dahdi   cannot be installed in openvz virtual server.

 I dont have any experience with openvz at all.

 Thx,

 On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com
 wrote:

 Dear all,

 Is anyone has experience making Asterisk server with virtual server
 OPEN-VZ (in proxmox 3.4 box) ?

 My boss want to build a production server with it, and it will have +/-
 300 sip user (concurrent call maybe  150 call)

 Is it good to go, or not ?

 I really hope someone who have experience with it  willing to share
 with me...

 Thanks in advance...


 Best Regards,


 Ikka - Jakarta, Indonesia





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Re: [asterisk-users] OpenVZ with asterisk 13

[Mitul Limbani]

I guess best way for your boss to learn is to deploy a box once and get
bombed and then follow what ppl said here.

Both modes u should be the happy guy u see, u will get paid twice for same
work !!!

Mitul
On 07-Apr-2015 7:27 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com wrote:

 Dear Mitul,

 I already told my boss about it, I really want a single box, no virtual,
 but my boss insist.
 He said that openvz use less resource then KVM (or other virtual for
 cloud).
 I really need a solid analysis to argue with him.

 On the other hand, dahdi   cannot be installed in openvz virtual server.

 I dont have any experience with openvz at all.

 Thx,

 On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja ikka.ti...@gmail.com
 wrote:

 Dear all,

 Is anyone has experience making Asterisk server with virtual server
 OPEN-VZ (in proxmox 3.4 box) ?

 My boss want to build a production server with it, and it will have +/-
 300 sip user (concurrent call maybe  150 call)

 Is it good to go, or not ?

 I really hope someone who have experience with it  willing to share with
 me...

 Thanks in advance...


 Best Regards,


 Ikka - Jakarta, Indonesia





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Re: [asterisk-users] OpenVZ with asterisk 13

[Mitul Limbani]

With that kind of load, your users shall start complaining about choppy
audio or voice clarity on random occasions, and you wont have a clue where
to look for the problem.



Regards,
Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422


On Tue, Apr 7, 2015 at 9:57 PM, Jeff LaCoursiere j...@jeff.net wrote:

 On 04/07/2015 10:48 AM, Johan Wilfer wrote:

 Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev:

 Dear all,

 Is anyone has experience making Asterisk server with virtual server
 OPEN-VZ (in proxmox 3.4 box) ?

 My boss want to build a production server with it, and it will have +/-
 300 sip user (concurrent call maybe  150 call)


 As long as you don't overload the server it works great. I've used OpenVZ
 to separate Asterisk instances from each other. For my application (mostly
 conferencing) I can put ~ 350 concurrent calls on a single HP Xeon server.

 OpenVZ is not really like KVM but more like Solaris containers or BSD
 jails. Docker is mostly using the same Kernel api:s that OpenVZ uses, but
 OpenVZ also has some cusom stuff.

 If you need Dahdi you will need to give the VE's access to these devices,
 there are articles out there that explain how this is done.

 Good luck!


 We use LXC (what is under Docker) instead of OpenVZ to separate asterisk
 instances, and when Dahdi is needed I typically run an asterisk instance
 on the host and have SIP trunks between the container and the host
 instances.

 Cheers,


 j

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Re: [asterisk-users] WebRTC demo phones

[Mitul Limbani]

Sipml5 works. You need to have TLS enabled on asterisk web socket.

Mitul
On 12-Mar-2015 12:47 PM, David Cunningham dcunning...@voisonics.com
wrote:

 Hello,

 Can anyone recommend a particular online WebRTC phone for testing with
 Asterisk?

 We tried:

 - JsSIP, but even with the enable video checkbox disabled it sends video
 options in the INVITE SDP and Asterisk rejects it with Rejecting secure
 video stream without encryption details.

 - sipML5, but it won't register, perhaps something to do with not using
 the Asterisk Websocket server (which I don't see an option to choose)

 - Janus, but the INVITE SDP contains RTP/AVP not RTP/SAVP, and Asterisk
 rejects it with We are requesting SRTP for audio, but they responded
 without it!

 Thanks for any suggestions.

 --
 David Cunningham, Voisonics
 http://voisonics.com/
 USA: +1 213 221 1092
 UK: +44 (0) 20 3298 1642
 Australia: +61 (0) 2 8063 9019

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Re: [asterisk-users] Music on hold

[Mitul Limbani]

Just split the file into multiple files n have it all uploaded to the same
music on hold class.

Now every time a caller is put on hold they will hear the files randomly.
On 06-Mar-2015 8:32 AM, Kris Stark kris.st...@godataflow.com wrote:

 OK - so somebody just handed me the new music on hold file to use for the
 organization...

 Unfortunately, I was never asked about this to enough detail to be able to
 tell them how to set up the music, and as a result I have an eight minute
 file with several different messages all tied together into that one file.

 In general, we don't ever see a user being placed on hold for more than a
 minute, so using this file directly is of no use in general if I were to
 place it directly in to the server, as all users will only hear the first
 little bit of it.

 I suspect that when this was created, the producer assumed that the file
 would play in a loop, starting and stopping as callers were on hold.  I
 realize that the streaming category will do just that, but since this is a
 local file, the setup works differently.  (This is replacing a set of about
 10 previous files that worked perfectly.)

 Is there any way, other than splitting up the file and trying to make
 decent segues between the files, to get this to work on a current version?
 I realize that getting it redone would be the best way, but I don't know if
 that is going to be an easy possibility.

 Any recommendations?

 Thanks!

 Kris

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Re: [asterisk-users] dialplan contexts syntax and terminology

[Mitul Limbani]

This one specifically

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#asterisk-DP-Basics-SECT-3.1
On 22-Feb-2015 11:13 AM, thufir hawat.thu...@gmail.com wrote:

 On Sun, 22 Feb 2015 08:32:26 +0530, Mitul Limbani wrote:

  READ READ READ 


 I know, I have the 4th edition and I've been reading it.  Personally, I
 find it more general than specific, but I'll go back through that
 chapter, absolutely.


 thanks,

 Thufir


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Re: [asterisk-users] dialplan contexts syntax and terminology

[Mitul Limbani]

READ READ READ 

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html



Regards,
Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422


On Sun, Feb 22, 2015 at 8:25 AM, thufir hawat.thu...@gmail.com wrote:

 I'm looking into the dialplan specifics:

 tleilax:~ #
 tleilax:~ # cat /etc/asterisk/extensions.conf
 [general]
 static=yes
 writeprotect=no

 [globals]
 CONSOLE=Console/dsp ; Console interface for
 demo
 TRUNK=DAHDI/r1; Trunk interface
 TRUNKX=DAHDI/r2 ; 2nd trunk interface
 TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569; IAX trunk interface
 TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569   ; IAX trunk interface
 TRUNKBINFONE=IAX2/111222:passw...@iax.binfone.com   ; IAX trunk
 interface
 SIPtrunk=SIP/1234:passw...@sip.provider.net ; SIP trunk

 #include extensions-vicidial.conf



 Firstly, what language or format is this? Bash script?

 the line #include ... what is this called? An include statement?

 The [globals] -- what's the terminology for this? It's a context?  And
 a context is a logical separation in the dialplan?  Is that, in any way,
 analogous to a function or method?

 Once you create your this logical separation, what's the syntax
 surrounding invoking a specific context?  For example:

 tleilax:~ #
 tleilax:~ # tail /etc/asterisk/extensions-vicidial.conf

 [vicidial-auto]
 exten = h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-
 NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-
 ${ANSWEREDTIME})

 include = vicidial-auto-internal
 include = vicidial-auto-phones
 include = vicidial-auto-external


 ; END OF FILELast Forced System Reload: 2015-02-20 16:49:28
 tleilax:~ #


 when the above contexts are included, these contexts are declared within
 the extensions-vicidial.conf, meaning that when they're declared, they're
 not actually used/invoked/called **until** the actual include = foo
 syntax?


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Re: [asterisk-users] One way audio internal

[Mitul Limbani]

You probably do not have enough g729 channels license.
On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote:

 On Friday 21 Nov 2014, Andrew Colin wrote:
  Hi All
 
  We have a strange issue with our hosted asterisk server running on Debian
  Internal calls btween extensions using g729 give one way audio
  As soon as we change the codec to ALAW the issues goes away.
 
  Any ideas how to fix this?
 
  Outbound calls via a trunk work fine with g729

 Unless you have serious bandwidth issues, just forget about g.729 and
 change
 to a-law throughout.  A-law is what the PSTN  (in civilised countries)
 uses
 anyway, so you won't need to transcode  (which chews up processor resources
 and risks compromising quality)  for calls to and from the outside world.

 If you really need to use g.729 and are outside the USA  (therefore, beyond
 the reach of software patents),  there is a free version that you can use
 --
 and this one, better than Digium's offering, comes with the Source Code so
 you
 can be sure it isn't doing anything nasty behind the scenes.

 But to be honest, you probably are better off just sticking with a-law.

 --
 AJS

 Note:  Originating address only accepts e-mail from list!  If replying off-
 list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] One way audio internal

[Mitul Limbani]

Then something to do with your codec selection priority.
On 21-Nov-2014 4:26 PM, Andrew Colin and...@convergedgroup.net wrote:

 I am using the free g729







 Kind Regards

 Andrew Colin

 *Converged Data (Pty) Ltd.*

 *Licensed Telecoms Operator :* (0258/IECS/JAN/09) (0258/IECNS/JAN/09)



 Direct: +27 (0)10 591 4607

 Mobile: +27 (0)82 310 3007
 Switchboard: +27 (0)10 591 4600
 Email: and...@convergedgroup.net

 Web: http://www.convergedgroup.net
 75 Witkoppen Road, Northriding, Johannesburg, 2169
 P O Box 7246, Weltevredenpark, 1715
 This communication is confidential and intended solely for the
 addressee(s). Any unauthorized review, use, disclosure or distribution is
 prohibited. If you believe this message has been sent to you in error,
 please notify the sender by replying to this transmission and delete the
 message without disclosing it. Thank you.E-mail including attachments is
 susceptible to data corruption, interception, unauthorized amendment,
 tampering and viruses, and we only send and receive emails on the basis
 that we are not liable for any such corruption, interception, amendment,
 tampering or viruses or any consequences thereof.



 *From:* Mitul Limbani [mailto:mi...@enterux.in]
 *Sent:* Friday, November 21, 2014 12:51 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Cc:* Andrew Colin
 *Subject:* Re: [asterisk-users] One way audio internal



 You probably do not have enough g729 channels license.

 On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk
 wrote:

 On Friday 21 Nov 2014, Andrew Colin wrote:
  Hi All
 
  We have a strange issue with our hosted asterisk server running on Debian
  Internal calls btween extensions using g729 give one way audio
  As soon as we change the codec to ALAW the issues goes away.
 
  Any ideas how to fix this?
 
  Outbound calls via a trunk work fine with g729

 Unless you have serious bandwidth issues, just forget about g.729 and
 change
 to a-law throughout.  A-law is what the PSTN  (in civilised countries)
 uses
 anyway, so you won't need to transcode  (which chews up processor resources
 and risks compromising quality)  for calls to and from the outside world.

 If you really need to use g.729 and are outside the USA  (therefore, beyond
 the reach of software patents),  there is a free version that you can use
 --
 and this one, better than Digium's offering, comes with the Source Code so
 you
 can be sure it isn't doing anything nasty behind the scenes.

 But to be honest, you probably are better off just sticking with a-law.

 --
 AJS

 Note:  Originating address only accepts e-mail from list!  If replying off-
 list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] ITSP Gateway Solution?

[Mitul Limbani]

Hey Tod,

Do message me offline, we might have few options to support your needs.

Mitul
On 12-Nov-2014 9:19 AM, Todd R. tjrl...@live.com wrote:

 Right now we I am using Asterisk boxes as a gateway between our Level 3
 SIP trunks and our customer PBXs.

 I love and understand Asterisk but the company I am working for is looking
 for a more Commercial type solution where we can go to a vendor for
 support etc. I know, we can get Asterisk support etc.. It's not my decision
 and I sort of get why they are leaning away from Asterisk, I just don't
 agree.

 I need to at least explore other options for more appliance products that
 will do the job the Asterisk boxes are doing now but, with a simple
 interface to add/remove trunks, DIDs etc. Integrated security and billing
 options/add-ons would be great.

 I know Digium offers appliance solutions but they don't seem to be
 anywhere near the power of what we are currently using.

 One big advantage I could see is going diskless but, I am really not sure
 whats out there, I am just kicking tires at the moment.

 The best of all worlds would be something with commercial support, a good
 GUI, billing and security built in but all based on the Asterisk core which
 I can understand :-)

 Again, just kicking tires as I can't just scream Asterisk and not be
 willing to look around to see what's out there.

 Everything I see out there seems to want to Transcode and such.. All we
 need is something to do SIP to SIP, no TDM here at all. Some codec support
 beyond G711 of course but that's it.

 I know there is every reason to do all this with Asterisk and that is my
 preference but in this case, I have lots of folks that lean more towards
 commercial products and I have not been able to completely sell them on the
 joy and flexibility of Asterisk.

 I don't want a Virtual PBX GUI solution, I want something that is built to
 be a work-horse, as a gateway only. No extensions, voicemail, ring groups
 or any of that. Just calls in/out to/from trunks, security and billing.

 Thanks!

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Re: [asterisk-users] Asterisk Phone ( Telecom feature )

[Mitul Limbani]

Hey Danni,

Having whale client means you ought to ask for paid consulting in first
place.

People over here have already told you that Analog phones are not capable
of doing paging.

Also so far you haven't indicated what model n which asterisk system your
client has. These are the most crucial elements to get desired answers. Do
some of these work before asking for help here (we all term these as
homework ) or have a paid consultant who would do the needful.

Hope this should set the tone right.

Mitul
On 08-Oct-2014 10:48 PM, Dania Asi da...@futuretrendsest.com wrote:

 Dear Mr. Adam,



 Thank you for you kind words and for judging me.



 I am a system integrator and I have a whale clients in UAE , I will not
 proceed further in dealing with Asterisk because of the lack of support and
 because of the rude emails.



 I have no idea what is wrong with you people. And I hope you get well soon
 from whatever is happening to you.





 *Best Wishes,*



 *Dania Abu Asi*

 Sales Executive Engineer

 [image: Description: cid:image002.png@01CE734B.028FC100]

 *Future Trends Establishment*

 Abu Dhabi - U.A.E.

 Mob : +971 50 4948363

 Off : +971 2 6730666

 Fax : +971 2 6734888



 *From:* Adam Goldberg [mailto:a...@agp-llc.com]
 *Sent:* Tuesday, October 7, 2014 8:51 PM
 *To:* Dania Asi
 *Subject:* FW: [asterisk-users] Asterisk Phone ( Telecom feature )



 I suggest that your question amounts to please do my homework for me.
 This may be understandable given that you are a recent grad and probably
 don't have much experience in business communication and/or Asterisk
 complexities.



 You cannot expect a mailing list to rush to answer vague, unanswerable
 questions -- nor emails that don't show that you've tried to answer the
 question first.



 Consider, if I asked:



 I don't understand how to set up Asterisk.  Can someone
 tell me how to do that?

 vs.



 I have a Dell R210-II with 32g of memory and two gigabit ethernet
 interfaces, I've installed Asterisk from the FreePBX Distro v9.99 and have
 an assortment of Polycom and Snom IP phones.  I've configured paging as
 described in http://wiki.snom.com/Interoperability/PBX/Asterisk and
 http://www.voip-info.org/wiki/view/Asterisk+cmd+Page, and it is working
 for the Snom phones but not the Polycom phones.  Can someone point me at
 what it's going to take to make the Polycom phones work?



 I’d expect to get attempts at an answer to the second one, but would
 expect snide and rude comments (at best) to the first one.



 Adam Goldberg

 AGP, LLC

 +1-202-507-9900



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [
 mailto:asterisk-users-boun...@lists.digium.com
 asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
 Sent: Tuesday, October 07, 2014 9:38 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )



 JG confirmed that it is possible, but it has not been defined.



 Without knowing what kind of instruments you are using, a possible it

 would be for a party to dial a 4-digit extension number to talk to someone
 internally, completing a call without using the PRI trunks.



   --Don



 -Original Message-

 From: asterisk-users-boun...@lists.digium.com

 [mailto:asterisk-users-boun...@lists.digium.com
 asterisk-users-boun...@lists.digium.com] On Behalf Of Dania Asi

 Sent: Tuesday, October 07, 2014 3:41 AM

 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

 Cc: 'Irene Galera'; 'Maysara Orabi'; moham...@futuretrendsest.com

 Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )



 Dear Mr. Mitual,



 Kindly check the attached mail where Mr. JG confirmed to me that is
 possible and I already informed my client of that.





 Dear Mr. Steve,



 I am not expecting a mailing list to do any work for me. All I was asking
 is for you to guide me because this is the first time we deal with Asterisk
 phones.





 Best Wishes,



 Dania Abu Asi

 Sales Executive Engineer



 Future Trends Establishment

 Abu Dhabi - U.A.E.

 Mob : +971 50 4948363

 Off : +971 2 6730666

 Fax : +971 2 6734888



 -Original Message-

 From: asterisk-users-boun...@lists.digium.com

 [mailto:asterisk-users-boun...@lists.digium.com
 asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes

 Sent: Tuesday, October 7, 2014 12:34 PM

 To: Asterisk Users Mailing List - Non-Commercial Discussion

 Cc: Irene Galera; Maysara Orabi; moham...@futuretrendsest.com

 Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )



 On 7 Oct 2014, at 09:24, Dania Asi da...@futuretrendsest.com wrote:

  Kindly note that I asked about the capability of the phones and now I

  am asking about the way I can do it to my client's phones, because he

  is asking for a demonstration.



 Yet you've not even told us the phones in use. You can't just expect a
 mailing list to do your work for you. You need to 

Re: [asterisk-users] SIP over 3G Mobile Network using NAT

[Mitul Limbani]

Oops its qualify= n not notify=

Also check if your asterisk sip server I available with ports on the public
ip that your phone is trying to register from 3G nw.
 On 09-Oct-2014 6:56 PM, mi...@enterux.in wrote:

 Remove Notify= setting in your sip.conf device section.
 On 09-Oct-2014 6:52 PM, Chirag Ajmera chi...@ncc.co.in wrote:

  Dear,

 Kindly guide with the 2 issues mentioned below

 *#1* - *Host unreachable 0 last qualify 0 (only in 3G**)*

 I am trying to use SIP client over 3G. It registers and call can be
 initiated from the client but it can't receive call; cause *asterisk
 sever *marks it as unreachable immediately after registration.

  [2014-10-08 14:32:47] NOTICE[1610]: chan_sip.c:29596
 sip_poke_noanswer: Peer '1007' is now UNREACHABLE! Last qualify: 0

 The above work well when I turn off 3g and switch over to my office wifi.
 Kindly guide if there are specific settings for 3G / mobile network.

 *#2* - *SIP retransmits - no reply to our critical packet*

 Issue occurs when dialing a call out from a remote wifi network ( in my
 case office wifi ), Its auto disconnected within 10s with SIP
 Retransmissions notice / warning message

 However if a call is initiated from home local network ( ipad ) to the
 phone ( registered with asterisk over office wifi )... all works well !

*Issues with call sequence in a table form*  *Local Device* Asterisk
 SIP Server  Router with NAT Port Forwards iPad with Bria SIP Client

   *Remote Device* iPhone with Bria SIP Client







   *CALL SEQUENCE*  *Remote Network* *From * *To* *Works/Failed* *Issue*
 *3G* Ipad ( local ) Iphone ( remote ) FAILED Host unreachable 0 last
 qualify 0 !

  Iphone ( remote ) Ipad ( local via NAT ) WORKS – both Audio  Video





   *Office Wifi / Broadband* Ipad ( local ) Iphone ( remote ) WORKS –
 both Audio  Video

  Iphone ( remote ) Ipad ( local via NAT ) FAILED SIP retransmits - no
 reply to our critical packet ! Call disconnects within 10s

 Kindly guide

 --
 Thank You
 Best,
 Chirag A.

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Re: [asterisk-users] SIP over 3G Mobile Network using NAT

[Mitul Limbani]

Remove Notify= setting in your sip.conf device section.
On 09-Oct-2014 6:52 PM, Chirag Ajmera chi...@ncc.co.in wrote:

  Dear,

 Kindly guide with the 2 issues mentioned below

 *#1* - *Host unreachable 0 last qualify 0 (only in 3G**)*

 I am trying to use SIP client over 3G. It registers and call can be
 initiated from the client but it can't receive call; cause *asterisk
 sever *marks it as unreachable immediately after registration.

  [2014-10-08 14:32:47] NOTICE[1610]: chan_sip.c:29596 sip_poke_noanswer:
 Peer '1007' is now UNREACHABLE! Last qualify: 0

 The above work well when I turn off 3g and switch over to my office wifi.
 Kindly guide if there are specific settings for 3G / mobile network.

 *#2* - *SIP retransmits - no reply to our critical packet*

 Issue occurs when dialing a call out from a remote wifi network ( in my
 case office wifi ), Its auto disconnected within 10s with SIP
 Retransmissions notice / warning message

 However if a call is initiated from home local network ( ipad ) to the
 phone ( registered with asterisk over office wifi )... all works well !

*Issues with call sequence in a table form*  *Local Device* Asterisk
 SIP Server  Router with NAT Port Forwards iPad with Bria SIP Client

   *Remote Device* iPhone with Bria SIP Client







   *CALL SEQUENCE*  *Remote Network* *From * *To* *Works/Failed* *Issue*
 *3G* Ipad ( local ) Iphone ( remote ) FAILED Host unreachable 0 last
 qualify 0 !

  Iphone ( remote ) Ipad ( local via NAT ) WORKS – both Audio  Video





   *Office Wifi / Broadband* Ipad ( local ) Iphone ( remote ) WORKS – both
 Audio  Video

  Iphone ( remote ) Ipad ( local via NAT ) FAILED SIP retransmits - no
 reply to our critical packet ! Call disconnects within 10s

 Kindly guide

 --
 Thank You
 Best,
 Chirag A.

 --
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Re: [asterisk-users] Asterisk Phone ( Telecom feature )

[Mitul Limbani]

It can't be done in analog phones.
On 07-Oct-2014 1:54 PM, Dania Asi da...@futuretrendsest.com wrote:

 Dear JG,

 Thank you for following up with me.

 Kindly note that I asked about the capability of the phones and now I am
 asking about the way I can do it to my client's phones, because he is
 asking
 for a demonstration.



 Best Wishes,

 Dania Abu Asi
 Sales Executive Engineer

 Future Trends Establishment
 Abu Dhabi - U.A.E.
 Mob : +971 50 4948363
 Off : +971 2 6730666
 Fax : +971 2 6734888

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
 Sent: Tuesday, October 7, 2014 12:21 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 You asked this question before and there was an already answer on September
 28.

 jg

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Re: [asterisk-users] Voice-Recognition / ASR / with barge in

[Mitul Limbani]

AFAIK, the generic asterisk speech API has an application : -
SpeechBackground

Not sure if it would work with your custom speech engine, you might have to
look at the Generic Speech API to make it work for ur engine.

Mitul

On Thursday, September 18, 2014, Thorsten Göllner t...@ovm-group.com wrote:

 Hi there,

 I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine
 :-) But I am wondering if there is a solution/application which will
 enable me to implement voice recognition while playing a voice file
 (barge in). So that the caller hears a voice file and can interrupt it
 with his voice.

 Currently (on our platform) the caller has to wait for the end of the
 voicefie. Then we play a beep. And then we record his voice and realize
 voice recognition with ispeech (it is an online service).

 Best regards
 -Thorsten-

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-- 
Regards,
Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] if statement recording - after hours

[Mitul Limbani]

Read GotoIfTime function.p
On 12-Sep-2014 3:13 AM, Joseph syscon...@gmail.com wrote:

 In my dial plan I have these two lines:

 exten = _NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${
 STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
 exten = _NXX,n,MixMonitor(${recordfilename},b)

 How to add if statement to execute these line only after let say 5pm.
 To record conversation only after 5pm.

 --
 Joseph

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Re: [asterisk-users] Special functionality for Secretary/Boss

[Mitul Limbani]

Kevin,

With your dialplan with g option on external trunk, if the call finishes
the boss's leg of call also gets disconnected. So the next instruction
would make a call to secratary, however with no one on other end.

Mitul
On 04-Sep-2014 11:44 PM, Kevin Larsen kevin.lar...@pioneerballoon.com
wrote:

 asterisk-users-boun...@lists.digium.com wrote on 09/04/2014 11:57:40 AM:
   We are currently migrating from a Nortel pbx to Asterisk and we
  have been able to convert most of the functions that people are used to
  but there is one I have no clear idea how to do.  The scenario is:
 
   Boss calls secretary from outside the office to get connected to
  another outside destination.  The secretary dials the destination and
  then trasfers call to the boss.  When boss finishes with that person
  they want to send the call back to the secretary in order to make
  another connection or simply to talk to the secretary.
 
   The first part is not a problem, but after the boss finishes his
  call how can we send the call back to the secretary?  I was thinking of
  using a conference room but how would the secretary know when the boss
  has finished?  Anyone know how to handle this scenario?

 I haven't tested this, but my initial thought would be to create a special
 context or extension that the secretary could route through when doing the
 call transfer. The Dial application could be called with the 'g' option to
 continue the dialplan at the next priority when the call hangs up.
 Something like a normal call transfer would just dial the number as normal,
 but for the special transfer, you could prepend the dialed number with a #.

 For example (using a local US dialstring, change to fit your needs):

 ; This is a normal external call.
 exten = _NXXNXXX,1,Dial(SIP/your_external_trunk/${EXTEN})
   same = n,Hangup()

 ; This is a call that should be transfered back to the secretary's
 extension when external call is finished
 exten = _#NXXNXXX,1,NoOp(Special Dial for Boss/Secretary Transfer)
   same = n,Dial(SIP/your_external_trunk/${EXTEN:1},,g)
 ; First call has ended, now we go back to the secretary)
   same = n,Dial(SIP/1234)
   same = n,Hangup()

 That's at least where I would start with my testing and then develop the
 solution from there.
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Re: [asterisk-users] packet2packet bridging

[Mitul Limbani]

Put sip debug on to know if reinvite packets are sent.
 On 09-Jul-2014 1:17 PM, Sameer Rathod sam...@hostnsoft.com wrote:

 Hi,

 Please clear me on this topic I am confused

 My log show switching to native rtp.
 Did this line means that the audio is not coming to the asterisk server
 any more and asterisk only send the re- invite packet to both the clients ?

 Am I right or wrong ?


 On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani mi...@enterux.in wrote:

 No way to avoid bw charges for any of the client if it is behind any sort
 of NAT.
 On 08-Jul-2014 8:52 PM, Sameer Rathod sam...@hostnsoft.com wrote:

 Hi Eric,


 I am behind nat

 Is there any solution for the same.

 My goal is to deduct the balance
 for the call but free my asterisk server from audio packet load.


 On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling ewiel...@nyigc.com wrote:

 I think you will find that direct audio between two endpoints does not
 work when NAT is involved.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sameer Rathod
 *Sent:* Tuesday, July 08, 2014 11:18 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] packet2packet bridging



 Hi Joshua,

 I had disabled

 ice support and remover encryption= yes

 Then also it is showing the same native_rtp in log

 Could you help me in bypassing asterisk server for audio?

 please help me I am struggling with it form a long time.





 On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod sam...@hostnsoft.com
 wrote:

  -- Channel SIP/1060-008e left 'native_rtp' basic-bridge
 3c12ca41-e180-4fc1-80cf-1339b96da42b
 -- Channel SIP/1061-008f left 'native_rtp' basic-bridge
 3c12ca41-e180-4fc1-80cf-1339b96da42b
   == Spawn extension (sameer, 1061, 1) exited non-zero on
 'SIP/1060-008e'

 here are more generated when I cut the call



 On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod sam...@hostnsoft.com
 wrote:

 so In this case If I disable ice support

 ie commented the icesuppot=yes from all files

 then also I am getting this output


 -- Executing [1061@sameer:1] Dial(SIP/1060-008e, SIP/1061) in
 new stack


   == Using SIP RTP CoS mark 5
 -- Called SIP/1061

 -- SIP/1061-008f is ringing
 -- SIP/1061-008f answered SIP/1060-008e
 -- Channel SIP/1061-008f joined 'simple_bridge' basic-bridge
 3c12ca41-e180-4fc1-80cf-1339b96da42b
 -- Channel SIP/1060-008e joined 'simple_bridge' basic-bridge
 3c12ca41-e180-4fc1-80cf-1339b96da42b
 Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from
 simple_bridge technology to native_rtp
 0x7f6800039020 -- Probation passed - setting RTP source
 address to 192.168.1.176:8000
 0x7f6780045810 -- Probation passed - setting RTP source
 address to 192.168.1.191:8000






 On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp jc...@digium.com wrote:

 Sameer Rathod wrote:

 yes I had configured

 icesupport=yes ;



 Asterisk does not support direct media establishment (with either
 chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use.



 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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 --

 Regards

 Sameer Rathod

 8109413462






 --

 Regards

 Sameer Rathod

 8109413462






 --

 Regards

 Sameer Rathod

 8109413462



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 Regards
 Sameer Rathod
 8109413462


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Re: [asterisk-users] packet2packet bridging

[Mitul Limbani]

No way to avoid bw charges for any of the client if it is behind any sort
of NAT.
On 08-Jul-2014 8:52 PM, Sameer Rathod sam...@hostnsoft.com wrote:

 Hi Eric,


 I am behind nat

 Is there any solution for the same.

 My goal is to deduct the balance
 for the call but free my asterisk server from audio packet load.


 On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling ewiel...@nyigc.com wrote:

 I think you will find that direct audio between two endpoints does not
 work when NAT is involved.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sameer Rathod
 *Sent:* Tuesday, July 08, 2014 11:18 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] packet2packet bridging



 Hi Joshua,

 I had disabled

 ice support and remover encryption= yes

 Then also it is showing the same native_rtp in log

 Could you help me in bypassing asterisk server for audio?

 please help me I am struggling with it form a long time.





 On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod sam...@hostnsoft.com
 wrote:

  -- Channel SIP/1060-008e left 'native_rtp' basic-bridge
 3c12ca41-e180-4fc1-80cf-1339b96da42b
 -- Channel SIP/1061-008f left 'native_rtp' basic-bridge
 3c12ca41-e180-4fc1-80cf-1339b96da42b
   == Spawn extension (sameer, 1061, 1) exited non-zero on
 'SIP/1060-008e'

 here are more generated when I cut the call



 On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod sam...@hostnsoft.com
 wrote:

 so In this case If I disable ice support

 ie commented the icesuppot=yes from all files

 then also I am getting this output


 -- Executing [1061@sameer:1] Dial(SIP/1060-008e, SIP/1061) in
 new stack


   == Using SIP RTP CoS mark 5
 -- Called SIP/1061

 -- SIP/1061-008f is ringing
 -- SIP/1061-008f answered SIP/1060-008e
 -- Channel SIP/1061-008f joined 'simple_bridge' basic-bridge
 3c12ca41-e180-4fc1-80cf-1339b96da42b
 -- Channel SIP/1060-008e joined 'simple_bridge' basic-bridge
 3c12ca41-e180-4fc1-80cf-1339b96da42b
 Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from
 simple_bridge technology to native_rtp
 0x7f6800039020 -- Probation passed - setting RTP source address
 to 192.168.1.176:8000
 0x7f6780045810 -- Probation passed - setting RTP source address
 to 192.168.1.191:8000






 On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp jc...@digium.com wrote:

 Sameer Rathod wrote:

 yes I had configured

 icesupport=yes ;



 Asterisk does not support direct media establishment (with either
 chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use.



 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



 --

 Regards

 Sameer Rathod

 8109413462






 --

 Regards

 Sameer Rathod

 8109413462






 --

 Regards

 Sameer Rathod

 8109413462



 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 --
 Regards
 Sameer Rathod
 8109413462


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Re: [asterisk-users] recording in mp3

[Mitul Limbani]

Move the .wav to diff server which has the processor to keep converting
files in runtime.

Asterisk would never have direct file save to mp3 due to patent
restrictions.

And pls Dont hijack the thread of packet filter. Open new email thread !!!
On 01-Jul-2014 9:43 PM, andrew Colin and...@vsave.co.za wrote:

 Problem with this is client needs to listen to the call recordings and my
 interface will only display .wav or .mp3 so they will moan if they have to
 wait until the next day for today's recordings


 Sent from Samsung Mobile


  Original message 
 From: binary
 Date:01/07/2014 6:09 PM (GMT+02:00)
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] recording in mp3

 i would go for recording into wav.
 then at regular intervals eg every night at 01:00 i would start a script
 to convert the wav to mp3 and then delete the wav files.
 it is really easy.



 On 30/6/2014 23:30, Scott Griepentrog wrote:

  ​You will not be able to able to save much space if any by using MP3
 instead of ulaw or wav -- at least not without expending a lot of CPU time
 to encode the file at a very low bitrate which sounds pretty bad even with
 just speech.  One of the better space savings options for recordings or
 voicemail is gsm.  Of course, using an MP3 format just because you ​prefer
 that is understandable.

  Additionally, I'm nearly 100% certain that Asterisk does not support
 encoding and directly writing MP3 files.



 On Mon, Jun 30, 2014 at 3:11 PM, andrew Colin and...@vsave.co.za wrote:

  Hey guys

  Is it possible to record with mixmonitor straight into mp3.

  I am trying to reduce disk space and want my calls to be recorded in
 mp3 Instead of wav.




  Sent from Samsung Mobile


  Original message 
 From: Sameer Rathod
 Date:30/06/2014 9:23 PM (GMT+02:00)
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Fwd: Regarding packet2packet bridging


Dear concern,


  I want to configure packet2packet bridging in asterisk.
  How could I do this any of the tutorial or instructions will help ?

  I found the setting the canreinvite=yes  will do the stuff but it is not
 working

  I am using asterisk 12.3 version

  I am very new to asterisk please help me in doing the same.

  Thanks in advance.

 --
 Regards
 Sameer Rathod
 8109413462




 --
 Regards
 Sameer Rathod
 8109413462


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  --
  [image: Digium logo]
 Scott Griepentrog
 Digium, Inc · Software Developer
 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
 direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
 Check us out at: http://digium.com · http://asterisk.org




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Re: [asterisk-users] Attack on Sip server.

[Mitul Limbani]

I think your asterisk server is behind firewall or some sort of NAT where
the out to in packets are getting masqueraded with local or DMZ  IP of your
firewall / gateway box.

Fix this first to get fail2ban detect the correct public IP.

Otherwise fail2ban will ban your local GW IP due to which you won't be able
to access the box even from your local network for ssh.

Hope u know how to fix the firewall snat.

Mitul
On 27-Jun-2014 9:51 PM, Jai Rangi jpra...@didforsale.com wrote:

 Anurag,

 Here is small script, that will check your logs and will block the IPs.
 http://www.didforsale.com/blog/is-your-asterisk-system-under-heavy-attack

 This is good if you dont expect any registration. If you do have some
 valid registration, you might want to add some counter to see how time IP
 need to fail or how many different users IP is trying to register on before
 blocking the IP.

 Jai Rangi
 www.didforslae.com



 On Fri, Jun 27, 2014 at 7:37 AM, Anurag Rana anuragrana31...@gmail.com
 wrote:


 Hi All.

 Someone is attacking on my SIP server.
 There are lot of requests coming in and I am not able to stop it because
 I am unable to detect the IP address.
 I used wireshark to capture the packets.

 Although I am using very strong password for my SIP users but still is
 there any way to drop these packets and stop this attack.

 I tried dropping packet after matching some string (most of the packets
 from attacker contains string 'VaxSIPUserAgent/3.1' ) but it failed.
 Packets are still flowing in.

 iptables -I INPUT 1 -p tcp --dport 5060 -m string --string VaxSIPUserAgent 
 --algo bm -j DROP


 ​Its something like this

 Registration from '30 sp:30@my_public_ip:5060 failed for
 '192.168.xxx.xxx:6373' - Wrong Password​

 ​and there are approx 10 request per minute of this type.

 Please suggest some way to stop this.​


 --
 Anurag Rana
 http://newbie42.blogspot.in/
 On the trampoline of life's experiences, Striving towards a saintly life
 in the midst of these materialistic turbulences.



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Re: [asterisk-users] Attack on Sip server.

[Mitul Limbani]

No way out. Fix ur gateway which is masquerading out to in traffic.

And do some research as others mentioned instead of expecting quick fix.

Mitul
On 27-Jun-2014 10:45 PM, Anurag Rana anuragrana31...@gmail.com wrote:

 Can't use anything which block IP addresses because my system is behind a
 gateway and attacker gets the address of that gateway. In this way I will
 end up blocking myself.

 Please suggest something else.


 On Fri, Jun 27, 2014 at 10:24 PM, Anurag Rana anuragrana31...@gmail.com
 wrote:

 Right Mitul. System is behind some gateway.


 On Fri, Jun 27, 2014 at 10:06 PM, Mitul Limbani mi...@enterux.in wrote:

 I think your asterisk server is behind firewall or some sort of NAT
 where the out to in packets are getting masqueraded with local or DMZ  IP
 of your firewall / gateway box.

 Fix this first to get fail2ban detect the correct public IP.

 Otherwise fail2ban will ban your local GW IP due to which you won't be
 able to access the box even from your local network for ssh.

 Hope u know how to fix the firewall snat.

 Mitul
 On 27-Jun-2014 9:51 PM, Jai Rangi jpra...@didforsale.com wrote:

 Anurag,

 Here is small script, that will check your logs and will block the IPs.

 http://www.didforsale.com/blog/is-your-asterisk-system-under-heavy-attack

 This is good if you dont expect any registration. If you do have some
 valid registration, you might want to add some counter to see how time IP
 need to fail or how many different users IP is trying to register on before
 blocking the IP.

 Jai Rangi
 www.didforslae.com



 On Fri, Jun 27, 2014 at 7:37 AM, Anurag Rana anuragrana31...@gmail.com
  wrote:


 Hi All.

 Someone is attacking on my SIP server.
 There are lot of requests coming in and I am not able to stop it
 because I am unable to detect the IP address.
 I used wireshark to capture the packets.

 Although I am using very strong password for my SIP users but still is
 there any way to drop these packets and stop this attack.

 I tried dropping packet after matching some string (most of the
 packets from attacker contains string 'VaxSIPUserAgent/3.1' ) but it
 failed. Packets are still flowing in.

 iptables -I INPUT 1 -p tcp --dport 5060 -m string --string 
 VaxSIPUserAgent --algo bm -j DROP


 ​Its something like this

 Registration from '30 sp:30@my_public_ip:5060 failed for
 '192.168.xxx.xxx:6373' - Wrong Password​

 ​and there are approx 10 request per minute of this type.

 Please suggest some way to stop this.​


 --
 Anurag Rana
 http://newbie42.blogspot.in/
 On the trampoline of life's experiences, Striving towards a saintly
 life in the midst of these materialistic turbulences.



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 --
 Anurag Rana
 http://newbie42.blogspot.in/
 On the trampoline of life's experiences, Striving towards a saintly life
 in the midst of these materialistic turbulences.





 --
 Anurag Rana
 http://newbie42.blogspot.in/
 On the trampoline of life's experiences, Striving towards a saintly life
 in the midst of these materialistic turbulences.



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Re: [asterisk-users] Echo Cancellation when calling from softphone to mobile.

[Mitul Limbani]

Put line side echo cancelation chip on ur PRI card.
On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.com wrote:

 Hi,

 I am using Twinkle to call mobile phone but there is too much noise on the
 mobile user's end. Mobile user's voice is echoed back to user. While on
 twinkle end everything is fine.

 Using Asterisk 11.

 Please suggest some way to mitigate the problem.

 Thanks.



 --
 Anurag Rana
 http://newbie42.blogspot.in/
 On the trampoline of life's experiences, Striving towards a saintly life
 in the midst of these materialistic turbulences.



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Re: [asterisk-users] Redfone FoneBridge2 Quad T1/E1 Alternative

[Mitul Limbani]

Hello,

Do respond back Offline ..

Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422



On Tue, Jun 24, 2014 at 11:35 PM, tirveni yadav yadav.tirv...@gmail.com
wrote:



 We have been using Red-fone foneBridge2 Quad T1/E1 for last few years.

 As these devices are not available anymore, we are looking for
 alternatives.

 Are there any similar devices available ?



 --
 Regards,

 Tirveni Yadav
 www.udyansh.org

 What is this Universe ? From what it arises ? Into what does it go?
 In freedom it arises, In freedom it rests and into freedom it melts away.
 Upanishads.

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Re: [asterisk-users] Open Source Asterisk Polling Solution

[Mitul Limbani]

Use vicidial for achieving the same.

Mitul

On Monday, April 21, 2014, Nick Cameo sym...@gmail.com wrote:

 Hello Everyone,

 We are looking for a simple open source auto dialer with polling
 capabilities. What we would like is a program that we can upload
 leads to, and have asterisk:

 i) Dial numbers
 ii) Play pre-recorded
 iii) If user presses one, forward the call to an agent

 There are so many solutions out there it's hard to make a decision on what
 works, what has just a limited free version etc Something that can
 support
 10 channels, and is stable would be greatly appreciated.

 If this can be simply implemented using asterisk and call folder, even
 better

 PS Our preferred version of * is 1.8.x

 Kind Regards,

 Nick from Toronto.



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] Webrtc and adventures with Asterisk 11

[Mitul Limbani]

Hello,

I was able to use webrtc2sip and connect audio calls in g729 passthrough
and ulaw modes over a callus webpage js.

However not tested Video.

and it worked good even on AST 1.8.XX


Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422



On Mon, Apr 14, 2014 at 2:26 PM, Johan Wilfer li...@jttech.se wrote:

 Hi,

 I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 +
 opus/vb8 codec patch. This is interesting technology and I try to find out
 how to connect all the moving parts.

 Firefox:
 Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't
 matter.
 WARNING[977][C-0005] chan_sip.c: Rejecting secure audio stream without
 encryption details: audio 35684 RTP/SAVPF 109 0 8 101
 -- Asterisk sends SIP/2.0 488 Not acceptable here

 Chrome:
 I've tried both sipml5 and jssip softphones and they both work. Even video
 + confbridge works with some minor quirks (lost connections sometimes, I
 guess plain old nat issues).
 Just relaying audio+video with confbridge to a handful of participants
 seems to use quite a bit of cpu thought.

 Screen-share:
 This works, but Confbridge is not very happy about a channel with video
 (vp8) and not audio and is printing this 80 times a second:

 WARNING[8919][C-] channel.c: Unable to find a codec translation
 path from (vp8) to (slin)
 WARNING[8919][C-] chan_sip.c: Asked to transmit frame type slin,
 while native formats is (vp8) read/write = unknown/unknown
 WARNING[8919][C-] channel.c: Don't know any of (vp8) formats


 How do you think about adding webrtc to a existing Asterisk/Kamailio
 environment? Do you use kamailio (websockets) as a front, a dedicated
 webrtc asterisk or something like webrtc2sip?

 How do you use / plan to implement webrtc in your environment?

 Any feedback is welcome. Thanks!

 --
 Johan Wilfer


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Re: [asterisk-users] SIP fraud IP blacklist

[Mitul Limbani]

Looks nice, might start using it Stefan :)

Thanks.

Mitul

On Friday, April 11, 2014, Stefan Gofferje li...@home.gofferje.net wrote:

 Hi,

 in case, anyone is interested...
 I have started compiling a blacklist of hosts and networks from which
 SIP fraud attempts occur.
 My criteria currently are:

 To block an IP:
 - Minimum 3 attacks within one week from the same IP
 To block a network:
 - Attacks from minimum 3 IPs from that network within 2 weeks
 Common criteria:
 - Provider does not react to complaints OR
 - Provider sends autoreply but attacks don't stop within a week

 Definition of attack:
 - Minimum 5 attempts to make an unauthorized phone call to a
 non-PBX-internal number OR
 - Minimum 10 attempts to make an unauthorized phone call to a
 PBX-internal number OR
 - Minimum 10 failed authentication attempts

 If this happens, the IP gets auto-banned (iptables) for 24 hours and
 goes to my watch list. The watch list is the base for my further decisions.

 Currently, I don't remove IPs or networks from the list. If I have time
 and/or motivation I might create some kind of removal process later -
 also, depending on how big the list gets and how many people use it.

 The list is yet pretty short but for me, it has reduced the noise on my
 PBX from 20-30 attacks per day to about 2 or 3 per week, especially
 after most of the Palestinian networks ended up on the list.

 You're free to use the list - own your own responsibility and risk. It's
 in the ipdeny.com format, so a simple script can be used to CURL the
 list and create iptables rules from it. A sample script for something
 like that is also on my website (check the Linux section).

 That's the website for the list:
 http://stefan.gofferje.net/it-stuff/sipfraud/sip-attacker-blacklist

 And that's the download URL:
 http://stefan.gofferje.net/sipblocklist.zone

 Note that the list is updated every 6h so polling it more often doesn't
 help anything. Please limit polling to once a day or so.

 -S

 --
  (o_   Stefan Gofferje| SCLT, MCP, CCSA
  //\   Reg'd Linux User #247167   | VCP #2263
  V_/_  Heckler  Koch - the original point and click interface




-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] Regarding SIP-T/SIP-I support in Asterisk.

[Mitul Limbani]

You can achieve this by setting relevant sip flags in the dialplan back and
forth.

Mitul
On Mar 12, 2014 11:18 PM, DHAVAL INDRODIYA dhaval.it01...@gmail.com
wrote:

 Thanks Amit,

 I want following scenario.

 INCOMINGCALL --- MSC (SIP-T)   PBX (Asterisk)

 OUTGOINGCALL ---  PBX (Asterisk) (SIP) to (SIP-T) --- Aircel MSC

 I understood that via Dial-plan we can achieve and get extra parameters
values. But what about RTP fields as per my analysis ISUP packets are not
sending RTP/AVP they are sending multipart data.

 please correct me if can achieve this functionality.

 Thanks
 Dhaval


 On Wed, Mar 12, 2014 at 6:15 PM, Amit a...@avhan.com wrote:

 Hi Dhaval,

 Theoretically, Asterisk can support SIP-I / SIP-T. Since protocols
provide additional information and controls, you will not get those
benefits. You will have to write dial plan functions to extract addition
information exposed by SIP-I / SIP-T.
 Though, I have not tested it with Asterisk, I have successfully deployed
application on other SIP platforms and interoperability with SIP-I/SIP-T
was not an issue.

 Regards,
 Amit Patkar



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Re: [asterisk-users] High Availability with Asterisk

[Mitul Limbani]

Hello,

Using Single Server with multiple VMs essentially kills the purpose, coz it
doesnt protect against physical hardware failures.

To save costs, use low end box as failover, to keep u in business, till
primary box goes live.

Mitul
On Mar 6, 2014 8:51 PM, Thorolf Godawa nos...@godawa.de wrote:
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Re: [asterisk-users] Enterprise VoIP Trunk

[Mitul Limbani]

Gopal,

We can do it. Tell ur requirements offline, mebbe we could help you out.

Mitul

On Thursday, March 6, 2014, Gopalakrishnan N gopalakrishnan...@gmail.com
wrote:

 Am looking for a service provider who can provide enterprise SIP trunk
 with 100 channels concurrent sessions.

 I see some like Inphonex, Broadvoice... and etc

 Is there any suggestions for the service providers.

 Regards



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] Dialer software for Asterisk...

[Mitul Limbani]

Have a look at vicidial it has alternate number dialing capability.

Mituo

On Saturday, February 15, 2014, Carlos Chavez cur...@telecomabmex.com
wrote:

 I have a customer with a more or less unique need.  Right now we are
 using Wombat as a dialer software so they can contact clients for QA
 purposes.  Everything is working very well and their contact center
 productivity is way up from the old manual dialing method.

 The only thing we are having a problem with is that they have up to 5
 phone numbers to contact a single customer.  Obviously we cannot load all
 numbers into the dialer because we do not want to contact the same customer
 5 times.  Does anyone know of a dialer for Asterisk that can take several
 phone numbers for the same contact and if any of those answers it will not
 try the other numbers?  Most of the dialers I have looked at cannot relate
 information for different numbers so there is no way to tell if you have
 already contacted a specific customer with a different number.

 I really do not want to develop a new dialer software (well, while the
 dialer is not that difficult the interfaces, reports and backends are a
 pain to maintain).  Anyone know of a commercial or open source software
 that can handle this kind of dialing?

 --
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Carlos Chávez
 +52 (55)9116-91161


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Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] Connect to remote GW

[Mitul Limbani]

Make two trunk entries in sip.conf for both IPs and do the failover routing
in extensions.conf

Mitul

On Wednesday, February 5, 2014, Meadows Hoa meadows_...@yahoo.com wrote:

 If SIP channel driver needs to connect to a remote GW over a dedicated SIP
 trunk BUT the remote GW has a 'standby' in case of failure, how can the sip
 configuration file be configured for the remote GW when there are actually
 two IP addresses. If the main remote GW fails control automatically
 switches to the standby GW, so how could the SIP configuration file hande
 this switch and support both host IP addresses. There is no DNS so straight
 IP addressing is used.



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] Asterisk on Windows

[Mitul Limbani]

As per that theory 3CX should have been public by now !!

Mitul
On Dec 4, 2013 8:49 PM, CDR vene...@gmail.com wrote:

 Digium is 100% lost in the map. If they would come up with a Paid
 version of Asterisk, one that would use the .NET framework in Windows,
 something simple to install, they could go public on the product.
 Linux has a very steep learning curve. A Windows application that
 would do exactly the same would be a home run. Note: I am a Linux
 expert user, but it took me years to get here. And still, moving from
 regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. The .NET
 framework and Windows server 2012 are miles away in terms of
 friendliness and on equal footing on performance. I don´t mean another
 slow cygwin port, I man a native Asterisk for windows. In fact, I
 would invest on the project if somebody wants to do it.

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Re: [asterisk-users] Asterisk on Windows

[Mitul Limbani]

Use FreeSWITCH !! Thats what you want on your winblows system, so suit
yourself my friend.

Mitul
On Dec 5, 2013 12:43 AM, Ruddy Gbaguidi plugwo...@micnes.com wrote:

 I never tought this is become a Linux vs Windows fight.
 We have been using asterisk on linux from a long time now and happy with
 it.
 But some of our customers who has windows in their environment want to use
 our call center software we developed on top of asterisk.
 So, the question was :
 Did anybody ever tried to isolate the asterisk SIP server/module and make
 it run under Windows ?
 Since, asterisk 12 is using pjsip (which is cross platform already), I
 tought it may be possible and wanted advices.

 I would love that every single customer switch to Linux and Ubuntu
 tomorrow morning but at the moment, that's not the case.

 Thanks.

 Le 2013-12-04 11:31, Patrick Lists a écrit :

 Probably feeding the trolls but here it goes.

 On 12/04/2013 04:19 PM, CDR wrote:
 Digium is 100% lost in the map. If they would come up with a Paid
 version of Asterisk, one that would use the .NET framework in Windows,
 something simple to install, they could go public on the product.

 IIRC Microsoft no longer invests in the .Net framework which makes it a
 bad idea for a product that would live for up to 10 years. Do you really
 want to bet your business/company that .Net will be there in 5 to 10
 years?

 Linux has a very steep learning curve. A Windows application that
 would do exactly the same would be a home run.

 I find Linux easier than Windows. Installing a package on Linux or
 Windows is not the issue. How is a simple 'yum install asterisk' any
 more difficult than double clicking on it in Windows? It's what you do
 afterwards with the OS and package. Asterisk has a much steeper learning
 curve than either. It's easy to mess up the config and suffer the
 consequences if the box is Internet facing. Also, Windows has a terrible
 reputation when it comes to security. Why would anyone want to use
 Windows for an Internet facing service? There's a reason that Google,
 Facebook, Twitter and pretty much the rest of the world are powered by
 Linux and it's not only because it's cheaper.

 Just because you find Windows easier does not make it a good idea.

 Note: I am a Linux
 expert user, but it took me years to get here. And still, moving from
 regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck.

 There is probably a saying about people calling themselves experts and
 then complain about a move from EL6 to F20 which is puzzling by itself.

 The .NET
 framework and Windows server 2012 are miles away in terms of
 friendliness and on equal footing on performance.

 I have yet to see a large Telco or ITSP deploy their services on
 Windows. A while back I have seen some attempts. It was hilarious to
 hear that the servers had to be restarted every few hours. Performance
 totally sucked, components would crash and the solution was, even by
 telco standards, ridiculously expensive. So no, they are not on equal
 footing when it comes to performance (and other aspects).

 I don´t mean another
 slow cygwin port, I man a native Asterisk for windows. In fact, I
 would invest on the project if somebody wants to do it.

 If you really want to use Windows then have a look at FreeSWITCH as it's
 available on Windows too. Then there is also Lync and 3CX. Good luck
 keeping your Windows boxes from getting hacked with all the financial
 and other damage it would cause.

 Regards,
 Patrick


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Re: [asterisk-users] issue with speech in IVR

[Mitul Limbani]

Try following in chan_dahdi

immediate = yes
echocancel = no
dtmfmode = auto

Mitul
On Nov 29, 2013 1:42 PM, isr...@gmail.com wrote:

 Are you using a mp3 file?
 I have noticed that using control playback with a mp3 file I cannot use
 the keypad to control the playback

 -Original Message-
 From: Salaheddine Elharit salah.elharit...@gmail.com
 Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013
 08:05:16
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] issue with speech in IVR

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Re: [asterisk-users] issue with speech in IVR

[Mitul Limbani]

Sounds cool, I suspected the echo cancel situation, these are usually issue
even for FAX communication on dahdi.

Mitul

On Friday, November 29, 2013, Salaheddine Elharit wrote:

 hello

 i add the following in chan_dahdi and the issue has been solved  thanks a
 lot for your help and support now ican stop the speech and go to my context

 i really appreciate your help and support

 immediate = yes
 echocancel = no
 dtmfmode = auto

 -- Forwarded message --
 From: Salaheddine Elharit salah.elharit...@gmail.com javascript:_e({},
 'cvml', 'salah.elharit...@gmail.com');
 Date: 2013/11/29
 Subject: Re: [asterisk-users] issue with speech in IVR
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com javascript:_e({}, 'cvml',
 'asterisk-users@lists.digium.com');


 hello

 i add the following in chan_dahdi and the issue has been solved  thanks a
 lot for your help and support now ican stop the speech and go to my context

 i really appreciate your help and support


  2013/11/29 Mitul Limbani mi...@enterux.in javascript:_e({}, 'cvml',
 'mi...@enterux.in');

 Try following in chan_dahdi

 immediate = yes
 echocancel = no
 dtmfmode = auto

 Mitul
 On Nov 29, 2013 1:42 PM, isr...@gmail.com javascript:_e({}, 'cvml',
 'isr...@gmail.com'); wrote:

 Are you using a mp3 file?
 I have noticed that using control playback with a mp3 file I cannot use
 the keypad to control the playback

 -Original Message-
 From: Salaheddine Elharit salah.elharit...@gmail.comjavascript:_e({}, 
 'cvml', 'salah.elharit...@gmail.com');
 
 Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013
 08:05:16
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com javascript:_e({}, 'cvml',
 'asterisk-users@lists.digium.com');
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com javascript:_e({}, 'cvml',
 'asterisk-users@lists.digium.com');
 Subject: Re: [asterisk-users] issue with speech in IVR

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-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] Bulk forwarding to another Asterisk

[Mitul Limbani]

If using IAX then I would recommend setting up DUNDi or Switch statement in
dialplan.

Mitul
On Nov 17, 2013 12:50 PM, Steve Edwards asterisk@sedwards.com wrote:

 On Sat, 16 Nov 2013, Doug wrote:

  I want to be able to pass any number (variable length) to a context and
 then forward that to another asterisk server for processing by that servers
 dial plan.

 If I use a  _x!  it just stops at the first character.


 How about '_x.' or '_!.'

 Note that '!' will match any alphanumeric -- including T, h, i, s, t, etc.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
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Re: [asterisk-users] Asterisk 1.8.22

[Mitul Limbani]

Buy SIP Channel from Skype and you can configure it as sip trunk on
asterisk box.

Mitul

On Friday, November 8, 2013, motty cruz wrote:

 Hello, I have a fully functional Asterisk Server, I want to configure this
 server to be able to process call from Skype, can someone point me to a
 howto? or if there are suggestions on best way to approach this problem.

 Thanks,



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] TE420, is it possible do disable span (red blinking)?

[Mitul Limbani]

Just dont configure those spans n related channels inside chan_dahdi.conf

Mitul
On Nov 1, 2013 3:38 PM, Dmitry Melekhov d...@belkam.com wrote:

 Hello!

 Just got new server with TE420.
 Not all four spans will be used immediately, but spans not configured or
 not connected blink red light.
 Is it possible to turn span off, so my colleagues will not eventally tell
 me that something is wrong with asterisk? :-)

 Thank you!


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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Mitul Limbani]

Asterisk is a swiss army knife, you should either know how to use it or
rely on ready made software which control routing of calls through variable
bit rates (skype does that very effectively)

So the key here for you to research upon from those several hundred results
is variable bit rate codec negotiations

Mitul Limbani
www.facebook.com/enterux
www.facebook.com/entvoice
 On Oct 29, 2013 1:30 AM, Ron Wheeler rwhee...@artifact-software.com
wrote:

  I am reaching the same level of frustration.
 I have tried to find the source of the problems.
 We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk
 - No analogue.
 We have a very lightly loaded 60 Mbs cable link to the Internet that tests
 pretty close to that most of the time.

 I have not found any good tools to track down the causes of poor voice
 quality.
 In my case, I have good incoming quality and terrible quality going out.
 That is, I can hear people perfectly well but they complain that my voice
 drops out and is garbled regardless of who places the call.
 As a result,  I use Skype for all of my calls and if someone calls me, I
 call them back on Skype if they have any problems.
 I don't understand why Skype works so well and Asterisk works so poorly on
 the same environment.

 Googling Asterisk poor audio quality return several hundred thousand
 references

 Ron
 On 28/10/2013 2:29 PM, Eddie Mikell wrote:

 All,

  The users in our organization are well, quite frankly, sick of phone
 service that is being provided.  The choppy phone calls, and drop outs are
 detrimental to our sales force.

  I've tried about everything I can think of.

   Moved the asterisk server from VM machine to dedicated machine

  More than enough bandwidth

  Setting 802.1p = 7

  Set Dedicated voice traffic 35% of bandwidth.

  Not sure what option would be the best


   Put analog lines in the conference room to avoid the dropouts - leave
 the sip lines in place for day to day use

  Hire a consultant

  Ditch the system and buy a pre-packaged system - RingCentral or some
 such.

  There are no local asterisk professionals who can help, and we are a
 little leery of opening up our system to outside consultants.

  Anyone else face the above, and finally abandoned Asterisk for a
 commercial system?

  We have 167 users.
 I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
 conference rooms.

  Suggestions welcome.

  Best

  Eddie
 --
 Eddie H. Mikell
 Senior Systems Engineer
 RKG

  Office: 434.970.1010 x 124
 Email: emik...@rimmkaufman.com

  http://www.rimmkaufman.com
 http://twitter.com/rimmkaufman  http://www.linkedin.com/company/85385 
 http://plus.google.com/104980442218952272663/posts
   http://www.facebook.com/rimmkaufman  http://www.RKGblog.com






 --
 Ron Wheeler
 President
 Artifact Software Inc
 email: rwhee...@artifact-software.com
 skype: ronaldmwheeler
 phone: 866-970-2435, ext 102


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Re: [asterisk-users] Problem with call transfer from one server to another server

[Mitul Limbani]

Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd
link here.

Mitul
On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com
wrote:

 Dear All,

 I have pri with E1 facility that have 30 line and 100 pri number which is
 provided by service provider.Number started like 23568561,23568562,23568563
 and so on. Service provider provide last four digit number for did mapping
 like 4561,4562,4563.


 exten = 8561,1,Dial(SIP/4001@192.168.14.110,120,tT)
 exten = 8561,n,hangup()

 exten = 8562,1,Dial(SIP/4001@192.168.14.110,120,tT)
 exten = 8562,n,hangup()

 Call comes into first server successful.But problem with second server
 when call came into second server i got following error:

 * chan_sip.c:20063 handle_request_invite: Call from '' to extension
 '4001' rejected because extension not found.*

 In one more scenario:

 when i create one extension and call forwarding with this extension that
 time I'm able to transfer call successful the code is given below:

 exten = 5001,1,Dial(SIP/4001@192.168.14.110,120,tT)
 exten = 5001,n,hangup()


 Regards
 Akhilesh

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Re: [asterisk-users] Capture Media IP in CDR (CDR)

[Mitul Limbani]

Nailed it to the point Matt +1 on.this entire philosophy of open source.

Mitul
On Oct 14, 2013 7:19 PM, Matthew Jordan mjor...@digium.com wrote:




 On Sun, Oct 13, 2013 at 2:06 PM, CDR vene...@gmail.com wrote:

 snip


 I need Digium to store this IP in the CDR. I will be honest with the
 government and let them know that my tool is incapable of saving lives
 or safeguarding our national security because nobody thought about
 this.
 PD: I am not paying for a patch, since this is huge burden on a small
 company like mine, with a single employee, and also because the whole
 world will enjoy the benefit. It is not fair that I would have to hire
 somebody to patch Asterisk.
 I appeal to Digium to patch Asterisk.


 I won't comment any further on the technical aspects of what you are
 looking for; others have already pointed out how various portions of SIP
 messages can be stored in CDRs and how these portions of the SIP messages
 are (a) actually of more use than the media IP address in the SDP and (b)
 meet the requirements being levied by your use case.

 That aside, I do think it is important to note here that Asterisk does
 not, by default, have a warranty. This is clearly enumerated in sections 10
 and 11 of the GPLv2 license included with Asterisk [1]:

 NO WARRANTY

   11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
 FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW.  EXCEPT WHEN
 OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
 PROVIDE THE PROGRAM AS IS WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
 OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
 MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE.  THE ENTIRE RISK AS
 TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU.  SHOULD THE
 PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
 REPAIR OR CORRECTION.

   12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
 WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
 REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
 INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
 OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED
 TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY
 YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER
 PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE
 POSSIBILITY OF SUCH DAMAGES.

  END OF TERMS AND CONDITIONS

 As you are using software licensed free of charge under the GPLv2, there
 is no obligation by anyone in the community or at Digium to provide you
 with a patch. If you require assistance, there are many avenues you can
 choose to pursue to gain such assistance. Just as you profit by running
 Asterisk, others profit by customizing and supporting the Asterisk project.
 Asterisk is lucky to have many such talented developers who can assist you
 with such a development effort. If you really require this functionality, I
 highly suggest that you look to hire said developers to help you with this
 feature request [2].

 [1] http://svn.asterisk.org/svn/asterisk/branches/11/COPYING

 [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

 Matt

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] is g729 codec free? or under license???

[Mitul Limbani]

Are these end points Hard IP Phones having g729 codec?

If yes then you dont need any license. Just download passthrough g729
license.

Mitul
On Oct 2, 2013 1:17 PM, Frederic Van Espen frederic...@gmail.com wrote:

 On 10/02/2013 09:33 AM, s m wrote:

   and the last question is how many license key should i buy? i read
 that license for g729 is per-channel but i don't understand what channel
 exactly means here. this is my scenario :

 10endpointspbx181...**pbx182...pbx183...10endpoints

 pbx181 and pbx183 has 10 endpoints connected to them. the call between
 these endpoints are established by pbx182. if i want to buy a license
 for pbx182, how many license key do i need? just one because i have just
 one connection on it?  or two, because two trunks is defined on it? or
 as many as endpoints which are connected to each other via pbx182?


 AFAIK, you need one license for each channel that is transcoding from one
 given codec to g729 (or the other way around).

 So if at any given time on an asterisk box you would have a maximum of 3
 simultaneous calls that are g729 at one end and ulaw at the other, you
 would need a license key for 3 transcoding channels.

 Anyone, please correct me if I am wrong on this.

 Frederic

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Re: [asterisk-users] MDL-ERROR

[Mitul Limbani]

Something very similar happened on our boxes and we did nothing special
apart from changing DAHDI and Asterisk versions to higher stable (2.6.1 n
1.8.14 respectively) and it went off.

Don't really know exactly why it happens, it would be worthwhile to
investigate if you see it again on CLI or logs.

Mitul

On Friday, September 6, 2013, Richard Mudgett wrote:




 On Thu, Sep 5, 2013 at 1:02 PM, jg 
 webaccou...@jgoettgens.dejavascript:_e({}, 'cvml', 
 'webaccou...@jgoettgens.de');
  wrote:

 I have 2 ISDN BRI boxes, each with 4 spans, where the first one is
 configured as CPE, the second one as NET(so I don't need real lines for
 developing and testing).

 Once in a while I do see the following libpri error messages
 simultaneously on both boxes:

 PRI Span: 1 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 2 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 2 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 4 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 2 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 4 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 2 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 4 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 4 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 4 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 4 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 PRI Span: 4 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)
 ...
 PRI Span: 4 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state
 7(Multi-frame established)

 It does not seem to be a severe error, because everything seems to work.
 I have not yet figured out how to produce these error messages reliably.
 They seem to appear after booting both machines before any calls have been
 made.

 Does anybody know what to do with these messages?


 Q.921 just says to log the condition and carry on as if it did not
 happen.  It does not say what to do
 about it.  The condition is likely to happen when the Q.921 peers are out
 of sync with each other.  When
 you reboot your machines, they will be out of sync with each other for
 awhile.  I cannot think of a
 particular sequence of events off-hand where it would happen though.

 Richard



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] using E1 PRI lines

[Mitul Limbani]

Operators are unnecessarily confusing you by talking tech Lang which you
are not well versed with. Are you trying to create prod / services which
they don't want u to launch but they have to provide lines under some sort
of regulatory obligations ?

Just go ahead n plug the wires on the E1 card ports.

Mitul

On Tuesday, July 30, 2013, Duncan Turnbull wrote:



 On 30/07/2013, at 4:22 PM, Akib Sayyed 
 akibsay...@gmail.comjavascript:_e({}, 'cvml', 'akibsay...@gmail.com');
 wrote:

 I didnt understand what you were saying.can you please explain

 I am using digium cards

 sent from android

 E1 PSTN line interfaces are either unbalanced 75 ohm( and used to use BNC
 connectors ) or a 120 ohm balanced twisted pair.

 The other standard is T1 and digium cards can let you choose between T1 
 E1 and definitely do 120 ohm

 Telco's will usually provide 120ohm twisted pair interfaces as it travels
 further and has less interference from noise.


 On Jul 30, 2013 6:55 AM, James zhu zhulizh...@live.comjavascript:_e({}, 
 'cvml', 'zhulizh...@live.com');
 wrote:

 hello:
 you can add T1_E1 by load card drivers

 Best regards,
 James.zhu
 website: www.hiastar.com

 --
 From: akibsay...@gmail.com javascript:_e({}, 'cvml',
 'akibsay...@gmail.com');
 Date: Mon, 29 Jul 2013 21:48:19 +0530
 To: asterisk-users@lists.digium.com javascript:_e({}, 'cvml',
 'asterisk-users@lists.digium.com');
 Subject: Re: [asterisk-users] using E1 PRI lines




 On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades 
 mailinglist+aster...@dns99.co.uk javascript:_e({}, 'cvml',
 'mailinglist+aster...@dns99.co.uk'); wrote:

 On 29/07/13 16:28, Akib Sayyed wrote:

 Dear asterisk users


 I wanted to use E1 pri lines on my asterisk box but my provider support
 only 120ohm on E1 line. I dont know how to set those values.

 Please help me

  Its done on whatever interface cards you have. Some may have a jumper
 setting. I know Sangoma has it in their configuration file (wanpipe).

 I am using digium card TE410P. can anyone help me how to change jumper
 settings


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 --
 Akib Sayyed
 Matrix-Shell
 akibsay...@gmail.com javascript:_e({}, 'cvml', 'akibsay...@gmail.com');
 akibsay...@matrixshell.com javascript:_e({}, 'cvml',
 'akibsay...@matrixshell.com');
 Mob:- +91-966-514-2243


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-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6

[Mitul Limbani]

Chan_zap has been deprecated more then 2-3 yrs back. You might have to ping
ipcortex helpdesk to get fix.

Mitul
On Jul 11, 2013 4:32 PM, Xavier Singer - EcuTek xav...@ecutek.com wrote:

 We use an IPcortex PABX running Asterisk 1.2.39-BRIstuffed-0.3.0-PRE-1y-y.
 We have recently implemented Call Queuing for our main incoming line with
 hold music. The call queue type is: Ring all - One call at a time (no
 position announcement).

 Since implementing this feature we've been receiving the below error daily
 in the mornings and lunchtime when the queue will jump to the next
 available phone, as the main reception phone is in Do Not Disturb mode:

 Jul 11 08:30:54 WARNING[23444] chan_zap.c: 1 Cause code 17 not
 allowed when disconnecting an active call. Changing to cause 16.
 Jul 11 08:30:54 ERROR[23444] chan_zap.c: You cannot use cause 17
 number when in state 6! Corrected.
 Jul 11 08:30:54 WARNING[7133] chan_zap.c: Call specified, but not
 found?
 Jul 11 08:30:54 NOTICE[7133] chan_zap.c: Hangup, did not find cref
 1, tei 127
 Jul 11 08:30:54 WARNING[7133] chan_zap.c: Hangup on bad channel
 0/1 on span 1
 Jul 11 08:30:58 WARNING[7133] chan_zap.c: Call specified, but not
 found?
 Jul 11 08:30:58 NOTICE[7133] chan_zap.c: Hangup, did not find cref
 1, tei 127
 Jul 11 08:30:58 WARNING[7133] chan_zap.c: Hangup on bad channel
 0/1 on span 1
 Jul 11 08:47:04 WARNING[7133] chan_zap.c: 1 received SETUP message
 for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:08 WARNING[7133] chan_zap.c: 1 received SETUP message
 for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:19 WARNING[7133] chan_zap.c: 1 received SETUP message
 for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:23 WARNING[7133] chan_zap.c: 1 received SETUP message
 for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,

 The ERROR happens when the call is ended. I can't replicate the error
 either...

 I suspect that the chan_zap driver has a bug and is possibly trying to
 hang up the call on the first phone in the queue, rather than the phone
 that answered the call.

 I have investigated the different state and causes listed in the above log
 file, and this is what I think they mean (please correct me if I got it
 wrong):
 ISDN State 6 = not initialised
 Cause 16 = normal call clearing
 Cause 17 = user busy
 TEI 127 = reserved as the broadcast TEI


 So my questions are:
 1. What could be causing the error and any suggestions on how to
 troubleshoot this issue?
 2. Can I upgrade the chan_zap driver for the ISDN card without breaking
 the IPcortex frontend (we have root access)?
 3. Should I supply any config files?


 Thanks!
 Xavier



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Re: [asterisk-users] SIP Trunking Mantra (Origination)

[Mitul Limbani]

Interesting.
You might want to consider paying some expert for consulting ?

Mitul
On Jun 22, 2013 7:21 PM, Nick Khamis sym...@gmail.com wrote:

 Hello Everyone,

 We are currently having talks with various service providers, and
 trying to determine what the best way is to interconnect in order to
 have access to the PSTN network. As you know there are two ways of
 doing this:

 Traditional PRI: Have trunks grouped into a transport layer such as
 OC3/12. With DIDs attached to the group. As you many know, this
 approach would also require a POP near the CO of the exchange we want
 to service etc.. We could also have the service provider backhaul some
 of the NXX in areas we do not have a POP, to a location near by.

 SIP Trunking: SIP traffic coming through the end of transport layer
 such as OC3 or ethernet connection directly connected to the service
 provider, with DID that can come from anywhere. No need for a POP in
 Chicago, for example, when we are located in Kansas.

 The benefits of one over the other are known, and not the topic of
 this message. What we are trying to determine are:

 When talking market price, a virtual PRI/SIP Trunk interconnect
 costs about 500-550 per 24 channel virtual pri. This compared to a
 true ISDN/PRI which can costs between 200-500 dollars depending
 who you talk to. We also have to take into consideration the hardware
 needed for either setup i.e.:

 * Option 1: SIP Proxy
 * Option 2: media gatweays, multiplexers, media server

 Even though it was natural to talk about pricing, this is still not
 what we are interested in knowing. What we are interested in finding
 out is:

 * How are service providers that offer virtual pris interconnected
 with their suppliers? I would imagine that some (non-CLECS), are
 renting a connection from the LECs, and grouping PRI/ISDN trunks
 (option 2). And others (CLECS), have a A-Link/ISUP trunk interconnect
 to the CO.
   - Which brings up a second question. How does a PRI trunk group
 differ from an ISUP
 trunk. I don't know much about and ISUP trunks and would *really*
 appreciate having
someone educate us on (i) the concept, (ii) what type of equipment
 would be needed,
(iii) how it differs from ISDN trunk groups. (iv) is it only
 available for LECS

 I do have more questions, however for the sake of brevity will stop
 right here. And before anyone asks the it depends what you want to
 do, I will mention that we are trying to establish an interconnection
 that will sustain 2016 channels or 84 T1s, and 5000 DIDs. We are not
 trying to become a CLEC however, still feel that option 2 would be the
 better choice for reasons covered here, and some that are left
 implicit (i.e, quality, reliability of managing our own
 networks..).

 Your insights are greatly appreciated!

 Nick.

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Re: [asterisk-users] SIGTRAN Integration

[Mitul Limbani]

I think you need a SIGTRAN stack from Netfors or LeibICT.

Mitul

On Friday, June 14, 2013, Nick Khamis wrote:

 Hello Everyone,

 I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP
 model.
 We are looking to interconnect with the PSTN world, and our supplier
 has given us
 a few options. We can either do this over traditional PRIs, A-Links or
 the SS7IP new.

 I am really interested in SIGTRAN, and was wondering how some of you
 have integrated
 it into your architecture. Can Asterisk handle SS70IP or do we have to
 put a yate or squire
 server at the end of that connection.

 Kind Regards,

 Nick.

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Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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Re: [asterisk-users] SIGTRAN Integration

[Mitul Limbani]

There is no open source solution for SIGTRAN yet.

If you come across one, do let everyone here know about it.

You can however request some time restricted demo from Netfors or LeibICT.

Mitul

On Friday, June 14, 2013, Nick Khamis wrote:

 Hello Mitul,

 Thank you so much for your response. During the testing phase
 we would like to employ an open source solution, and wanted
 to know what people have had success with, given the different
 user part etc..

 On a side note, anyone know of service providers offering SIGTRAN?

 Kind Regards,

 Nick.

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Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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Re: [asterisk-users] Not able to build the chan_sip.c module

[Mitul Limbani]

Without posting exact error messages, dont expect help !!

Mitul
On May 28, 2013 1:01 PM, upendra uppi...@gmail.com wrote:

 hi,


 anyone can help me to debug this ??


 --
 upendar


 On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote:

 hi,

 chan_local and res_crypto are building but the chan_sip is not building .
 installed openssl also but still the chan_sip not building.

 --
 Upendra


 On Mon, May 27, 2013 at 2:59 PM, Alec Davis siva...@paradise.net.nzwrote:

 
  i am trying to install asterisk newer version on the Elastix
  machine, but while installing the chan_sip,c module is not
  building while make. when i see  in make menuselect options
  it showing XXX -- extended , please let me know how to
  enable it and make build chan_sip module.
  --
  Upendra

 from makeselect you'll find chan_sip depends on the following
 Depends on: chan_local(M), res_crypto(M), res_http_websocket(M)

 then you'll find res_crytpo is dependant on open_ssl
  Depends on: openssl(E)

 which for me on debian wheezy is
 libssl-dev

 Alec Davis


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Re: [asterisk-users] Not able to build the chan_sip.c module

[Mitul Limbani]

Why not install the updated rpm version?

Mitul
On May 28, 2013 1:12 PM, upendra uppi...@gmail.com wrote:

 hi,

 there is no build errors , but the thing is that on Elastix Machine i want
 to install asterisk1.8.11.0 , while make the chan_sip module is not
 building, and when i see in the memuselect the chan_sip module driver
 showing as XXX to enable for building.

 --
 Upendra.


 On Tue, May 28, 2013 at 1:03 PM, Mitul Limbani mi...@enterux.in wrote:

 Without posting exact error messages, dont expect help !!

 Mitul
 On May 28, 2013 1:01 PM, upendra uppi...@gmail.com wrote:

 hi,


 anyone can help me to debug this ??


 --
 upendar


 On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote:

 hi,

 chan_local and res_crypto are building but the chan_sip is not building
 . installed openssl also but still the chan_sip not building.

 --
 Upendra


 On Mon, May 27, 2013 at 2:59 PM, Alec Davis siva...@paradise.net.nzwrote:

 
  i am trying to install asterisk newer version on the Elastix
  machine, but while installing the chan_sip,c module is not
  building while make. when i see  in make menuselect options
  it showing XXX -- extended , please let me know how to
  enable it and make build chan_sip module.
  --
  Upendra

 from makeselect you'll find chan_sip depends on the following
 Depends on: chan_local(M), res_crypto(M), res_http_websocket(M)

 then you'll find res_crytpo is dependant on open_ssl
  Depends on: openssl(E)

 which for me on debian wheezy is
 libssl-dev

 Alec Davis


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Re: [asterisk-users] Stress testing Asterisk

[Mitul Limbani]

I have a question here.

How can we test the quality of voice upon increasing the call load?

Can we try passing a voice file using sipp and record the same in dial plan
record application ? Is this reliable enough to simulate near real world
scenario?

Mitul

On Wednesday, May 22, 2013, Tommy Cooper wrote:

 Thank you for your help I finally solved this issue. Is it possible that
 my setup can achieve 212 concurrent calls, I am running Asterisk on just 1
 core using 3.5 GHz, and 1Gb of RAM?

  - Forwarded Message -
 *From:* Marie Fischer ma...@vtl.ee javascript:_e({}, 'cvml',
 'ma...@vtl.ee');
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com javascript:_e({}, 'cvml',
 'asterisk-users@lists.digium.com');
 *Sent:* Wednesday, May 22, 2013 1:16 PM
 *Subject:* Re: [asterisk-users] Stress testing Asterisk


 On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.comjavascript:_e({}, 
 'cvml', 'tomcoope...@yahoo.com');
 wrote:

  Hi,
  I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is
 generating are failing. I am trying to run Sipp on the same machine as
 Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.

 Do you have a peer and extension configured for SIPP in your Asterisk
 configuration? You also needat least the -s extension_to_dial option on
 your sipp command line.

 http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has
 some simple instructions which should get you started.
 If the calls still fail, Asterisk console output would be helpful.



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-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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Re: [asterisk-users] Performance Asterisk large installation on Vmware/Xen

[Mitul Limbani]

Not recommended to run Asterisk on Virualization.

Mitul
On May 18, 2013 11:33 PM, Rafael dos Santos Saraiva rafaels...@gmail.com
wrote:

 Hi

 I would like the opinion of you and if anyone has a similar scenario. I
 have a project for installation of a Asterisk server in a client with about
 400 extensions. My question is whether this scenario carry an Asterisk
 virtualized. Will be used only extensions and trunks sip sip, 1 queue with
 2 agents, without call recording. It is best to use XEN or VMware? Which
 best version of Asterisk for this scenario?

 Thank you.

 Att,
 *Rafael dos Santos Saraiva*
 Tel: (51) 8174-7956 | (51) 3205-1504
 http://www.astdocs.com | 
 http://br.linkedin.com/pub/rafael-saraiva/52/aab/230

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Re: [asterisk-users] dial and bridge

[Mitul Limbani]

The dial n bridge might work, but there ain't indefinite wait in that
scenario.
Direct calls to parking you might try Local(70X@from-internal) but I m not
sure if this method works reliably.

The method I mentioned is used by vicidial and it works flawlessly, yes it
comes with some computing load, however you can try the newer ConfBridge
app to see if its cheaper.

Mitul

On Wednesday, May 15, 2013, Lenz Emilitri wrote:

 Hi Mitul,
 I agree that the dialplan way is easier, but it's a client requirement to
 avoid using it. I was wondering if there was a way to send a call directly
 to a parking slot right from the originate, because that is cheaper than
 running conferences, and then joining the second call right to the parked
 call, so that all we have to do is two originates.
 l.


 2013/5/14 Mitul Limbani mi...@enterux.in javascript:_e({}, 'cvml',
 'mi...@enterux.in');

 Dial first call and put it into a conference, then dial second call and
 put him into same conference to bridge both.

 However dial plan way is much more simpler.

 Mitul


 On Tuesday, May 14, 2013, Lenz Emilitri wrote:


 Hi all,
 I need some advice - I have been working on originating multiple calls
 using AMI and then joining them.
 What I want to do is:
 - dial call 1 (where the caller is in a channel format, like SIp/1234
 or Local/1234@ext) and park it somehow
 - dial call 2 (where again the caller is in channel format) and join it
 to the previous call.

 As a requirement, I cannot use the dialplan as an end-point (as I cannot
 change it) but need to use the AMI only.

 I tried doing something like:

 Action: Originate
 Channel: Local/300@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 So that the call goes to 300 and then basically stays there forever, and
 then I dial again:

 Action: Originate
 Channel: Local/500@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 And then try to bridge the results, but it does not seem to work.
 What I would like to do would be more on the lines of:

 Originate call 1 and park it (using a park or waiting)
 Originate call 2 and bridge it immediately to call1 (using the
 Application part)

 But maybe I am missing something? is there anybody who has better
 suggestions?

 Thanks
 l.






 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com



 --
 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel,
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in javascript:_e({}, 'cvml', 'mi...@enterux.in');
 DID: +91-22-71967121
 Cell: +91-9820332422



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 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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Re: [asterisk-users] dial and bridge

[Mitul Limbani]

Dial first call and put it into a conference, then dial second call and put
him into same conference to bridge both.

However dial plan way is much more simpler.

Mitul

On Tuesday, May 14, 2013, Lenz Emilitri wrote:


 Hi all,
 I need some advice - I have been working on originating multiple calls
 using AMI and then joining them.
 What I want to do is:
 - dial call 1 (where the caller is in a channel format, like SIp/1234 or
 Local/1234@ext) and park it somehow
 - dial call 2 (where again the caller is in channel format) and join it to
 the previous call.

 As a requirement, I cannot use the dialplan as an end-point (as I cannot
 change it) but need to use the AMI only.

 I tried doing something like:

 Action: Originate
 Channel: Local/300@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 So that the call goes to 300 and then basically stays there forever, and
 then I dial again:

 Action: Originate
 Channel: Local/500@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 And then try to bridge the results, but it does not seem to work.
 What I would like to do would be more on the lines of:

 Originate call 1 and park it (using a park or waiting)
 Originate call 2 and bridge it immediately to call1 (using the Application
 part)

 But maybe I am missing something? is there anybody who has better
 suggestions?

 Thanks
 l.






 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] dial and bridge

[Mitul Limbani]

Dial first call and put it into a conference, then dial second call and put
him into same conference to bridge both.

However dial plan way is much more simpler.

Mitul

On Tuesday, May 14, 2013, Lenz Emilitri wrote:


 Hi all,
 I need some advice - I have been working on originating multiple calls
 using AMI and then joining them.
 What I want to do is:
 - dial call 1 (where the caller is in a channel format, like SIp/1234 or
 Local/1234@ext) and park it somehow
 - dial call 2 (where again the caller is in channel format) and join it to
 the previous call.

 As a requirement, I cannot use the dialplan as an end-point (as I cannot
 change it) but need to use the AMI only.

 I tried doing something like:

 Action: Originate
 Channel: Local/300@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 So that the call goes to 300 and then basically stays there forever, and
 then I dial again:

 Action: Originate
 Channel: Local/500@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 And then try to bridge the results, but it does not seem to work.
 What I would like to do would be more on the lines of:

 Originate call 1 and park it (using a park or waiting)
 Originate call 2 and bridge it immediately to call1 (using the Application
 part)

 But maybe I am missing something? is there anybody who has better
 suggestions?

 Thanks
 l.






 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

[Mitul Limbani]

Why dont u run a reverse dialer on the admin contacts phone number. Leave
him clueless as well.

Mitul
On Apr 5, 2013 1:25 AM, Joseph syscon...@gmail.com wrote:

 I receive several calls from this scamer: Senior SafeAlert
 It is an automated call and they keep rotating their caller ID so it is
 harder to block them.

 Does asterisk have a fax sound tone? If I block their number and play
 fax tone/sound maybe they will remove me from their calling list.
 I've tried to call them but nothing helps.
 Any better ideas? They keep calling sometimes few times a day.


 Their web site is 
 http://www.seniorsafealert.**com/http://www.seniorsafealert.com/

 Registered through: GoDaddy.com, LLC (Domain Names | The World's Largest
 Domain Name Registrar - Go Daddy)
 Domain Name: SENIORSAFEALERT.COM
 Created on: 28-Aug-12
 Expires on: 28-Aug-13
 Last Updated on: 28-Aug-12

 Registrant:
 Sam Alure
 8832 Thornbury Ln
 Huntersville, North Carolina 28078
 United States

 Administrative Contact:
 Alure, Sam seniorsafeal...@gmail.com
 8832 Thornbury Ln
 Huntersville, North Carolina 28078
 United States
 +1.7044972383

 --
 Joseph

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Re: [asterisk-users] ISDN- E1 PRI module in network side signaling

[Mitul Limbani]

Use pri_net as signalling mode inside chan_dadhi.conf in /etc/asterisk/
folder.

You can set this up using any pri card thats supported on Asterisk.

Mitul
On Mar 31, 2013 12:25 PM, Dimitar Dimitrov ddimit...@consult.bg wrote:

  Hello everyone.
 I am looking for a E1 PRI card which supports network side signaling not
 CPE. The main idea is to connect an plain old E1 compliant PBX which
 doesn't have an VoIP module to the newly created VoIP infrastructure.
 Could we use a Digium TE122P or something other to resolve this situation?

 Thanks in advance.
 Dimitar


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Re: [asterisk-users] app_rtsp.c ported to Asterisk 11.x

[Mitul Limbani]

Hey,

Can you send me URL to download the tar ball pls?

Mitul Limbani

On Saturday, March 16, 2013, Robert Krakora wrote:

 Hi,

 If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x.  I
 have tested it with GStreamer RTSP server and a C920 webcam streaming H264
 SVC video from one machine to another machine running Linphone.  Contact me
 at this e-mail address robkrak...@messagenetsystems.comjavascript:_e({}, 
 'cvml', 'robkrak...@messagenetsystems.com');for source code.

 Best Regards,

 --
 Rob Krakora
 MessageNet Systems
 101 East Carmel Dr. Suite 105
 Carmel, IN 46032
 (317)566-1677 Ext 212
 (317)663-0808 Fax



-- 
Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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Re: [asterisk-users] asterisk sizing for play and dtmf detection

[Mitul Limbani]

If you accept calls on.g711 and static ivr dialplan you should be able to
do around 300-400 concurrent on the box config that you provided.

And If you pay some expert consultant, he may be to fine tune it to be able
to handle 500 concurrent as well.

Which version of asterisk are you planning to use?
Any DB integration layer inside IVR?

Mitul Limbani
On Mar 8, 2013 5:20 PM, nik600 nik...@gmail.com wrote:

 Dear all

 i'm planning a migration to asterisk for a high volume IVR service
 (from 1000 to 1500 concurrent call)

 The IVR service is based only on DTMF tones so the features required is

 - play feature
 - dtmf detection

 Asterisk will receive calls via VOIP (SIP with g711 codec)

 The IVR service wil be a static service based on Asterisk dialplan
 with some prompt (from 0 to 5, play of files in the same codec of the
 received call) and some dtmf detections.

 How many simultaneous call can i handle per server? each server will have:

 4 core 3.0 Ghz
 4 GB of RAM

 I need an aproximate sizing:

 0-100 calls per server ?
 100-200 calls per server ?
 200-300 calls per server ?
 300-400 calls per server?
 400-500 calls per server?

 Thanks to all in advance

 --
 /*/
 nik600
 http://www.kumbe.it

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Re: [asterisk-users] digium card and virualbox

[Mitul Limbani]

convert the calls from PRI to SIP and throw it inside the VirtualBox
Asterisk, thats the ONLY WAY OUT

Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422



On Sat, Mar 9, 2013 at 2:51 AM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hi All;

 How to let the virualbox (ubuntu OS) to be able to see the digium card?
 Because when I install elastix or asterisk with dahdi, it is not able to
 see the digium card if the installation though the virualbox .. What is the
 solution?

 Regards
 Bilal

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Re: [asterisk-users] AEL Macro are evil :-)

[Mitul Limbani]

Hi,

You might want to use ${MACRO_EXTEN} variable inside to preserve exten
variable of the original dialplan exten variable.

Mitul
On Feb 24, 2013 4:04 PM, Leandro Dardini ldard...@gmail.com wrote:

 I just discover an hidden problem with AEL macro I want to have your
 feedback. If you use a macro to dial out, like dialout(${EXTEN}), the leg
 extension will became s and if it happens you transfer the call,
 that will be the callerid appearing on the other phone display.
 I am just rewriting all the dialplan getting rid of the macro and using
 gosub, even if asterisk is complaining about  application call to gosub
 affects flow of control, and needs to be re-written using AEL if, while,
 goto, etc. keywords instead!, but I am not seeing any other way...

 Leandro



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Re: [asterisk-users] Annoying delay after main server goes down

[Mitul Limbani]

I would suggest to use linux ha and use same ip, which can failover to
second standby server using heartbeat.

This activity takes less then 5secs.

Mitul
On Jan 18, 2013 9:42 PM, Ishfaq Malik i...@pack-net.co.uk wrote:

 On Fri, 2013-01-18 at 18:06 +0200, Onur Cem Çelebi wrote:
  Hello,
 
 
  we have distributed lots of cisco spa303 IP phones and get them work
  with Asterisk. I have configured proxy and alternate proxy and enabled
  dual registration features in provisioning files(xml files). All
  phones are able to subscribe to both of servers. But the problem is,
  if main server goes down, i am obliged to wait nearly 20 second in
  order to place a call over second server. How to get alternate
  proxy work immediately after first server fails ? Thanks in advance.
 

 I think the failover time will be related to the the registration expiry
 setting in the phones.
 
 
 


 --
 Ishfaq Malik i...@pack-net.co.uk
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
 NORTH, MANCHESTER
 SCIENCE PARK, MANCHESTER, M156SE
 COMPANY REG NO. 04920552


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Re: [asterisk-users] IVR platform for a mobile operator

[Mitul Limbani]

On Jan 9, 2013 8:38 PM, Danny Nicholas da...@debsinc.com wrote:

 From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon
 Sent: Wednesday, January 09, 2013 9:06 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] IVR platform for a mobile operator



 Hi Friends ,



 I want to setup a IVR platform using asterisk to a mobile operator.



 Can somebody give me some guides with recommended hardware types ?



 Thank you

 Luke.



 IMO you will be happiest with a SIP trunk handling this as there can be
horrible latency in DAHDI/Mobile connections.

Latency on DAHDI - heard this for first time 

TDM networks have zero latency we face latency only on IP (SIP) networks.

Mitul Limbani
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Re: [asterisk-users] DIDForSale spam

[Mitul Limbani]

+1 here.
On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com
wrote:

 On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote:
  On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote:
  What were the senders IP(s)?
 
  Will have to look it up when I get home.
 
  Doug
 
  --
  Ben Franklin quote:
 
  Those who would give up Essential Liberty to purchase a little
 Temporary Safety, deserve neither Liberty nor Safety.
 
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  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  I have gotten hit with this twice so far. in March and Today:
 
  Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com
  3/8/12
 
  DIDForSale donotre...@didforsale.com via mail.bingotelecom.com
  1/9/13
 
  UGH, when I asked in March where he got my email he said:
 
  Hi Chris,
  We got your contact from the Internet. Let me know the good time to
  talk about this in detail.
  Thank you,
  -Rohit Dhaka
 

 Obviously by harvesting these lists.  I received 2 myself.

 Thanks,
 Steve T

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Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

[Mitul Limbani]

Mebbe you guys should try snom m9 dect ip phone, i have been using it since
over 3 years now without any of these issues.

Mitul
On Dec 12, 2012 4:25 AM, Kai-Uwe Jensen kujen...@gmail.com wrote:

 Using a Gigaset C610IP here, and am very happy with the features. The base
 station can handle two concurrent SIP calls, and another internal one at
 that. It does it with a single SIP registration to each server. You can
 setup multiple servers if you want to and define dial patterns/plans that
 determine which server gets used. After some playing around with it, I'm
 now using my setup connected to a single asterisk only. (Let asterisk make
 call routing decisions based on cost, using an AGI)

 Call transfer is working fine, the handsets have a Flash/R key to
 accomplish this. Using the Flash lets you start a second call, and once
 answered you can easily conference the second party in (softkey right on
 the screen), or transfer the call to the other party (via menu, then
 transfer). Using this capability, someone on a call can easily confer with
 another party, and bridge them into the call. AFAIK it is not possible for
 someone to join an existing call easily. You'd have to implement that in
 asterisk's dialplan, not on the Gigaset phone.

 My understanding is that the C610IP has a few more features than the 510.
 I might've also read somewhere that the 510 is obsolete. Can't find that
 link right now, but search mgraves.org (use the Gigaset tag to get some
 initial results).


 On Tue, Dec 11, 2012 at 2:37 PM, Roy Abshire r...@coopvr.com wrote:

  That is true about the A580.

 I don't like the interface much to check messages.

 Besides that every time I go to dial a number...it always uses the first
 digit pressed to go into phone mode..so I have to press the first digit
 twice...

 I would test other phones but it's for home and I can't fork over $$ to
 try them all out

 I have tested some Nokia cell phones, the N97, N900, and E71 and the E71
 and N900 worked well.  I didn't like the N97.


 Co-op Vacation Rentalswww.coopvr.com
 15218 Summit Ave
 Suite #300-354
 Fontana, CA 92336
 Phone/Fax (855) 760-COOP (2667)

 On 12/11/2012 12:52 PM, Pete Mundy wrote:

 One thing I dislike about the A580H is that the handset always says 'You 
 have new messages' if I've missed a call. It wouldn't bug me if it said 
 'missed call' but it tells me I have new messages and even lights up a red 
 LED under a button with a picture of an envelope on it.

 I'm about to test an A510IP and an A610IP to compare against the A580. 
 Fingers crossed neither of them has that issue, because the Gigaset phone is 
 a pretty good phone other than that, and the difficulty doing a (blind) 
 transfer, as referred to by the OP.

 Pete


 On 12/12/2012, at 8:57 AM, Roy Abshire r...@coopvr.com r...@coopvr.com 
 wrote:


  I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. 
  Never gave me problems. The call Quality is excellent!
 I only have 1 handset connected to the Base but I want more. I bought a 
 Linksys WIP330 as a 2nd phone to try out and that works just as good without 
 a base unit.

 The A580 Base supports up to 6 handsets.

 I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 
 Handset but you can point each SIP to separate handsets.

 The call goes to the first phone that picks up.  When on a call, picking up 
 another phone makes a separate call and does not conference.  I don't use 
 conference yet but I know you have to put the call on hold or something.

 The thing I don't like about the A580 and might be the same on all of them 
 is that you can only specify 1 Sip Account for making outgoing calls.  In 
 other words, all 6 phones would use the same caller id out, but I wanted to 
 be able to choose that because I have a business number and number for each 
 person in our household.  In order to use a different Caller ID (SIP 
 Account) for making outgoing calls I added a extension to my Dial Plan and 
 before making outgoing calls I press *1-6 before the number.

 I'm going to try adding more handsets that are compatible.  I want the SL78H 
 but they are so expensive for just home everyday use.

 Make sure you check the compatibility page here before buying handsets.
 http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html


 Co-op Vacation Rentalswww.coopvr.com
 15218 Summit Ave
 Suite #300-354
 Fontana, CA 92336
 Phone/Fax (855) 760-COOP (2667)

 On 12/11/2012 11:32 AM, sean darcy wrote:

  Siemens A510IP



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Re: [asterisk-users] SIP password probe

[Mitul Limbani]

You might want to share the know how over here if its not a chan_sip patch.

Mitul
On Nov 28, 2012 12:28 AM, Ron Wheeler rwhee...@artifact-software.com
wrote:

  On 27/11/2012 12:58 PM, Christopher Harrington wrote:

 It's an open source project. Pay a programmer or make the modification
 yourself and submit a patch.

 You don't really want me coding!
 I have solved the problem for me.

 Just add it to the queue of enhancements for the next time someone is
 working on SIP.

 Ron



 On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler 
 rwhee...@artifact-software.com wrote:

 I looking through my logs, I found that people where probing my SIP
 accounts looking for passwords.
 Asterisk was helping them out by processing hundreds of requests per
 minute.
 I did a bit of Googling and this seems to be a frequent knock against
 Asterisk's security.

 It would seem pretty simple to add a configuration setting to sip.conf to
 delay the response to a bad account or password.

 There is a half measure to confuse the probe by sending the same error
 return for either error.
 It appears that many people have complained that this should be the
 default setting only changed if your are debugging a problem.

 There is no reason for a working system to ever have bad passwords so
 this is clearly an attack in almost every case.

 A simple delay would solve the problem for most people who use reasonable
 passwords.

 I had to install fail2ban which is a PITA but thanks to someone's clear
 recipe, I was able to get it working.

 I hope that this can be worked into a release soon.

 Ron

 --
 Ron Wheeler
 President
 Artifact Software Inc
 email: rwhee...@artifact-software.com
 skype: ronaldmwheeler
 phone: 866-970-2435, ext 102


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  --
 -Chris Harrington
 ACSDi Office: 763.559.5800
  Mobile Phone: 612.326.4248




 --
 Ron Wheeler
 President
 Artifact Software Inc
 email: rwhee...@artifact-software.com
 skype: ronaldmwheeler
 phone: 866-970-2435, ext 102


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Re: [asterisk-users] leading ghost 0

[Mitul Limbani]

Any changes inside chan_dahdi requires you to unload module chan_dahdi and
load module chan_dahdi, in case you dont wish to.restart asterisk.

pridialplan = national or unknown should help you solve the problem,
however you need to unload n load dahdi module.

Mitul
On Nov 21, 2012 10:26 PM, gincantalupo gincantal...@fgasoftware.com
wrote:

 **
 Alex,

 I had already tried itreloading chan_dahdi.so module is enough...I saw
 Asterisk was behaving differently after reload. To tell the truth, setting
 pridialplan=unknown causes Asterisk to stop reading following channels
 configuration...it says pridialplan is already unknown so it stops
 evaluating chan_dahdi.conf file useless to say that all n+1 channels do
 not work. Maybe it is a bug but with that parameter set in that way I
 cannot dial.

 I'm sure Asterisk is dialling the right number:

 [2012-11-21 09:05:29] VERBOSE[8314] logger.c:  [70 0b a1 33 34 39 3x 3x
 3x 3x 3x 3x 34]
 [2012-11-21 09:05:29] VERBOSE[8314] logger.c:  Called Number (len=13) [
 Ext: 1  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan
 (E.164/E.163) (1)  '3497078884' ]
 [2012-11-21 09:05:29] VERBOSE[8314] logger.c: q931.c:3134 q931_setup: call
 32781 on channel 6 enters state 1 (Call Initiated)
 [2012-11-21 09:05:29] VERBOSE[8314] logger.c: -- Called 6/349xx4

 I'm starting to think it is a telco problem... in case I'd change some
 parameter like pridialplan or similar, shouldn't I just see a leading 0 in
 the frame like this:
 [70 0b a1 *30* 33 34 39 3x 3x 3x 3x 3x 3x 34] added by Asterisk/DAHDI??

 I've used this page as reference about frame fields:
 http://www.acacia-net.com/wwwcla/protocol/q931_ie.htm

 Thank you.

 Giorgio Incantalupo


 On 11/20/2012 05:23 PM, Alex Kauffmann wrote:

 On 11/20/2012 8:03 AM, gincantalupo wrote:

 Hi Leandro,

 I'm sure nobody has added something... tried prilocaldialplan and
 pridialplan but nothing changed.
 Question: if pridialplan or prilocaldialplan would work, should I see
 the 0 inside PRI frame with intense debug or it is hidden?

 Yes...the technician did it...there is only one cable.

 Maybe it is the socket circuitry that has something wrong but I do not
 know ho to check.

 Asap I'll be on site I'll do more testing.

 Thank you

 Giorgio

 On 11/20/2012 01:13 PM, Leandro Dardini wrote:

 That is a real mistery! I like a lots these cases when all seems not
 working despite all being correctly configured, but you know first or
 later you'll find the answer.

 From your website, it seems you are selling/renting PBX based on
 asterisk, so you can be sure nobody has messed with the asterisk or
 dahdi source code adding a zero... I am sure you have already tried
 with a brand new server.

 Have you checked the pridialplan and prilocaldialplan setting?

 If I was in your shoes, I'll get another server, with a PRI configured
 as master and hook it at your PBX to really check if the zero is sent.

 Does the technician try to make phone calls from the same network
 cable you are using?

 Leandro


 2012/11/20 gincantalupo gincantal...@fgasoftware.com
 mailto:gincantal...@fgasoftware.com gincantal...@fgasoftware.com

 Hi Leandro,

 thanks for your answer.

 I already have tried those parameters but without any positive result.

 The telco technician has tried the line with its machine and it
 worked...remote telco technicians say they get a leading zero...
 I'm thinking there is something strange in the middle that adds
 the zero but do not know what it is.
 Strange is the fact that you can call some numbers with or without
 the prefix zero...
 Moreover we had no problem with the previous telco (fastweb).

 So we can only call PTSN numbersnot mobile phones.

 Giorgio


 On 11/20/2012 11:12 AM, Leandro Dardini wrote:

 2012/11/20 gincantalupo gincantal...@fgasoftware.com
 mailto:gincantal...@fgasoftware.com gincantal...@fgasoftware.com

 Hi all,

 I have problems dialling out because my new telco (the
 previous gave no problems) tells me my PBX adds a leading 0
 and that's why I cannot dial out (but I can receive calls).

 I make a small extensions.conf as a test:

 exten = 666,1,Dial(DAHDI/g1/339xx)
 but cannot dial out

 Curious thing is that
 exten = 666,1,Dial(DAHDI/g1/0233xx)
 and
 exten = 666,1,Dial(DAHDI/g1/233xx)
 call the same number!!!

 Line in use is a PRI.

 My Asterisk version is 1.4.26.2
 dahdi version: 2.2.0.2
 wanpipe-3.4.6

 I checked with intense pri debug and see no 0 inside frames

 How can I really be SURE Asterisk is not adding some leading
 zero?

 Thank you.

 Giorgio.


 I have never heard of a way to automatically add digits when
 using PRI, however can you check your chan_dahdi.conf about the
 following lines:

 internationalprefix 

Re: [asterisk-users] Need help for Aculab Prosody X PCI card installation and configuration with Asterisk

[Mitul Limbani]

AFAIK its a propreitary card from Aculab and wont work on Asterisk unless
you buy software or support or both from them.

My advice is to dump it n get a digium card in same or lesser cost which
you need to pay aculab.

Mitul Limbani
On Nov 12, 2012 1:23 PM, RAJNI VANZA rajniva...@gmail.com wrote:

 Hi All,

 I need to install and configuration of Aculab prosody X PCI card with
 Asterisk-1.8.9.1 on Centos-5.7 system.

 I will try for that but not success. so, please suggest me way to achieve
 it.

 Thanks in Advance.

 --
 Best Regards,
 Rajni Vanza

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Re: [asterisk-users] Installation Problem with Asterisk 1.6

[Mitul Limbani]

Dont enable everything that you see in installation without doing homework.

Just delete the extracted directory and reextract from tarball n follow the
INSTALL.txt peacefully !!!

Mitul
On Nov 4, 2012 12:30 PM, akhilesh chand omakhileshch...@gmail.com wrote:

 Hello, I am working with CentOS 5.3, asterisk 1.6.2.24 ,Whenever i 
 executemake command, i got the following error when installing asterisk:

 . make[2]: *** No rule to make target `anaFilter.o', needed by
 `libilbc.a'. Stop. make[1]: *** [ilbc/libilbc.a] Error 2 make: *** [codecs]
 Error 2

 i will really appreciate your help, thank you.




 Regards

 Akhilesh

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Re: [asterisk-users] multitenanat third party app

[Mitul Limbani]

Stop asking same questions !!!
On Oct 31, 2012 11:54 PM, Darin Iv adari...@gmail.com wrote:

 Is it possible to bul multitenant system using some third party opensouce
 application My design is like this.

 Company A:
 Context Company_A
 IVR Company A
 Extensions: 101,102,103,104 etc.

 Company B:
 Context Company_B
 IVR Company B
 Extensions: 101,102,103,104 etc.

 Company C:
 Context Company_C
 IVR Company
 Extensions: 101,102,103,104 etc.


 Company D:
 Context Company_D
 IVR Company D
 Extensions: 101,102,103,104 etc.

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Re: [asterisk-users] multi tenant

[Mitul Limbani]

Not possible to have same sip usernames.

However you can create
custA_user1 == 101
custB_user1 == 101

In the dialplan context.

Mitul
 On Oct 30, 2012 12:47 PM, Darin Iv adari...@gmail.com wrote:

 Hi all,

 I need to configure DIDs for different companies and they should reach on
 different extension with different context. Cant we have same extension
 in different context?

 This is what we we want
 Company A:
 Context Company_A
 IVR Company A
 Extensions: 101,102,103,104 etc.

 Company B:
 Context Company_B
 IVR Company B
 Extensions: 101,102,103,104 etc.

 Company C:
 Context Company_C
 IVR Company
 Extensions: 101,102,103,104 etc.


 Company D:
 Context Company_D
 IVR Company D
 Extensions: 101,102,103,104 etc.

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Re: [asterisk-users] Cant we have same extension in different context?

[Mitul Limbani]

FYI

SIP usernames =! Extensions

You have to use unique sip usernames to be identified inside dialplan for
mapping to extensions

[contextA]
Exten = 101,1,Dail(sip/custA_user1)

[contextB]
Exten = 101,1,Dial(sip/custB_user2)

Hope this makes it clear.

Mitul
On Oct 30, 2012 1:16 PM, Darin Iv adari...@gmail.com wrote:

 Cant we have same extension in different context?

 This is what we we want in same pbx server?
 Company A
 Context Company_A
 IVR Company A
 Extensions: 101,102,103,104 etc.
 Company B
 Context Company_B
 IVR Company B
 Extensions: 101,102,103,104 etc.

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Re: [asterisk-users] high capacity analog - sip gateway

[Mitul Limbani]

yealink T18 and T20 are decent phones available for $60

Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Fri, Oct 26, 2012 at 2:44 AM, Christopher Harrington ch...@acsdi.comwrote:

 On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.comwrote:

 I always advocate throwing out old analog phones as they will be a pain,
 but understand if you absolutely cannot.  Just keep in mind you can get a
 decent VoIP phone for $60 that is very likely to be nicer than what they
 have now and do much more.


 Out of curiosity, would you mind sharing that with us?


 --
 -Chris Harrington
 ACSDi Office: 763.559.5800
 Mobile Phone: 612.326.4248



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Re: [asterisk-users] dahdi dummy

[Mitul Limbani]

Need dummy to provide timing on machines that do not have a tdm board. Also
meetme dependency was on dummy or one of the tdm card.

I believe meetme has been rewritten since then.

Mitul
On Oct 23, 2012 9:58 PM, Warren Selby wcse...@selbytech.com wrote:

 If I remember correctly, dahdi dummy was removed and the functionally
 added by default when you load dahdi with no TDM cards installed. I could
 be wrong though.

 What do you need dummy for?

 Thanks,
 --Warren Selby, dCAP

 On Oct 23, 2012, at 10:28 AM, Jerry Geis ge...@pagestation.com wrote:

  I need to use the dahdi dummy driver.
  Its not being compiled at this time.
 
  When I go into tools subdirectory under dahdi-linux-complete-2.4.1
  and do make menuselect all I get is
  CC= CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
 CONFIGURE_SILENT=--silent nmenuselect
  make[1]: Entering directory
 `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect'
  make[1]: Nothing to be done for `nmenuselect'.
  make[1]: Leaving directory
 `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect'
  CC= CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
 CONFIGURE_SILENT=--silent gmenuselect
  make[1]: Entering directory
 `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect'
  make[1]: Nothing to be done for `gmenuselect'.
  make[1]: Leaving directory
 `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect'
  make[1]: Entering directory
 `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools'
  Terminal must be at least 80 x 27.
  menuselect changes NOT saved!
  make[1]: Leaving directory
 `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools'
 
  How can I get the dahdi_dummy.c driver compiled?
 
 
 
  Jerry
 
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Re: [asterisk-users] Counting calls in progress from AMI

[Mitul Limbani]

I guess you are looking for event handler, which can be polled
programatically n not via manual command entry?

Mitul
On Oct 18, 2012 8:53 PM, Danny Nicholas da...@debsinc.com wrote:

 The AMI Command function issues CLI commands, but carry on.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michelle Dupuis
 *Sent:* Thursday, October 18, 2012 10:20 AM
 *To:* Asterisk Users List
 *Subject:* Re: [asterisk-users] Counting calls in progress from AMI

 ** **

 I need to do this from the AMI (not the CLI)...I don't *think* a
 comparable command exists from the AMI.

  

 As well, I don't want to poll the system for calls so I'm hoping to trap a
 call bridged,unbridged type event.

  
 --

 *From:* asterisk-users-boun...@lists.digium.com [
 asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [
 da...@debsinc.com]
 *Sent:* Thursday, October 18, 2012 10:59 AM
 *To:* Asterisk Users List
 *Subject:* Re: [asterisk-users] Counting calls in progress from AMI

 The simplest way to accurately do this would be to issue command “core
 show channels verbose”

  

 *From:* asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michelle
 Dupuis
 *Sent:* Thursday, October 18, 2012 9:58 AM
 *To:* Asterisk Users List
 *Subject:* [asterisk-users] Counting calls in progress from AMI

  

 I want to track the number of calls up at any given time, through the
 AMI.  I found the Link and Unlink commands as the most likely candidates -
 is that the right way?

  

 Also, a comment on the wiki suggests that Link may be called several times
 for a single bridge if transcoding is required.  That blows up accuracy of
 my count of course...

  

 Ideas?

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Re: [asterisk-users] asterisk as IVR using 3g usb modem

[Mitul Limbani]

Short answer is, its not possible

Long answer, why it is not !!

U would have to write a dahdi module for this 3G modem to help asterisk
understand it as standard gsm channel.

Hope that help,

Mitul
On Oct 18, 2012 9:16 PM, Mahendra Dobariya mahendra_mahen...@hotmail.com
wrote:

 hi,
 I want to use asterisk as IVR system ,
 but to make and receive GSM call, i want to use 3g usb modem.(voice
 enabled)
 http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php

 and i want to install this system on two different machine
 1 on mac os x -
 2 raspberry pi- (debian wheezy)--http://www.raspberrypi.org/

 thanx in advance..



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Re: [asterisk-users] Issue with PRI connection

[Mitul Limbani]

put signalling=euroisdn in chan_dahdi.conf and restart asterisk.

Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Mon, Sep 24, 2012 at 11:33 AM, Ashish Agarwal ashisha...@gmail.comwrote:

 Hi,

 I have tried disabling and enabling crc4 before but that did not help.

 I have not defined any signalling value under chan_dahdi.conf

 Also, with respect to cabling we tried switching tx and rx but in that
 case we see alarm on the dahdhi status.

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Re: [asterisk-users] Issue with PRI connection

[Mitul Limbani]

Are you sure if PRI is the signalling or its EM based E1 links.

Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Sun, Sep 23, 2012 at 9:01 PM, Ashish Agarwal ashisha...@gmail.comwrote:

 Hello,

 I have install libpri 1.4.12 + Asterisk 10.7.0

 vi*CLI dahdi show status
 Description  Alarms  IRQbpviol CRCFra
 Codi Options  LBO
 T4XXP (PCI) Card 0 Span 1OK  0  0  0  CCS
 HDB3  0 db (CSU)/0-133 feet (DSX-1)
 T4XXP (PCI) Card 0 Span 2OK  0  0  0  CCS
 HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
 T4XXP (PCI) Card 0 Span 3OK  0  0  0  CCS
 HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
 T4XXP (PCI) Card 0 Span 4OK  0  0  0  CCS
 HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

 vi*CLI pri show spans
 PRI span 1/0: Down, Active
 PRI span 2/0: Down, Active
 PRI span 3/0: Down, Active
 PRI span 4/0: Down, Active

 vi*CLI pri show span 2
 Primary D-channel: 47
 Status: Down, Active
 Switchtype: EuroISDN
 Type: CPE
 Remote type: Unknown node type
 Overlap Dial: 0
 Logical Channel Mapping: 0
 Timer and counter settings:
   N200: 3
   N202: 3
   K: 7
   T200: 1000
   T201: 1000
   T202: 1
   T203: 1
   T303: 4000
   T305: 3
   T308: 4000
   T309: 6000
   T312: 6000
   T313: 4000
   T-HOLD: 4000
   T-RETRIEVE: 4000
   T-RESPONSE: 4000
   T-STATUS: 4000
   T-ACTIVATE: 1
   T-DEACTIVATE: 4000
   T-INTERROGATE: 4000
   T-RETENTION: 3
   T-CCBS1: 4000
   T-CCBS2: 270
   T-CCBS3: 2
   T-CCBS4: 5000
   T-CCBS5: 360
   T-CCBS6: 360
   T-CCNR2: 1080
   T-CCNR5: 1170
   T-CCNR6: 1170
 Q931 RX: 0
 Q931 TX: 0
 Q921 RX: 0
 Q921 TX: 304
 Q921 Outstanding: 0 (TEI=0)
 Total active-calls:0 global:0
 CC records:
 Overlap Recv: No

 For some reason pri show spans does not show up. Can someone assist me to
 fix this issue.


 --
 Regards,

 Ashish

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Re: [asterisk-users] Issue with PRI connection

[Mitul Limbani]

Signalling frm remote side is down.

Also just add crc4 in span 1,0,0 in dahdi/system.conf just like other spans.

what is signalling=  defined in your asterisk/chan_dahdi.conf ?

Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Mon, Sep 24, 2012 at 9:01 AM, Raj Mathur (राज माथुर) 
r...@linux-delhi.org wrote:

 On Monday 24 Sep 2012, Ashish Agarwal wrote:
  I have used 1 2 4 5 combination. Is that right?

 I wouldn't know, since I'm not the wizard :)  But basically we had to do
 each provider's connections from scratch -- Airtel, BSNL, MTNL,
 Reliance, Tata.  And as far as I recall, each provider had a different
 cable signalling scheme.

 Please don't top-post, it may get your posts ignored on the list.  Use
 bottom- or inline-posting, and trim your replies.

 Regards,

 -- Raj
 --
 Raj Mathur  || r...@kandalaya.org   || GPG:
 http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
 It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] Issue with PRI connection

[Mitul Limbani]

On Mon, Sep 24, 2012 at 10:11 AM, Raj Mathur (राज माथुर) 
r...@linux-delhi.org wrote:

 On Monday 24 Sep 2012, Mitul Limbani wrote:
  Signalling frm remote side is down.
 
  Also just add crc4 in span 1,0,0 in dahdi/system.conf just like other
  spans.
 
  what is signalling=  defined in your asterisk/chan_dahdi.conf ?

 Not necessarily.  I guess you remember the problems we had in getting
 lines to work in Mumbai until we found the correct wire connections for
 each of our providers (Reliance and BSNL AFAIR).  Eventually Raj Kumar
 (the wizard) did manage to figure it out and after that the lines worked
 just fine.

 I m aware of that, however just wanted to double check with his configs
before any of these conclusions :)
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Re: [asterisk-users] Need to Help for setup Video IVRS on Asterisk

[Mitul Limbani]

this is quite complicated to be setup. however you can try using :

asterisk 1.4.11 with libpri patch for h324m and app_h324m.

Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Thu, Sep 20, 2012 at 2:19 PM, RAJNI VANZA rajniva...@gmail.com wrote:

 Hi All;

 I was wondering if anyone has any experience for Video IVRS on Asterisk.

 3G User dial some number and he is able to see Video IVRS on his mobile. I
 need to setup Video IVRS on Asterisk PBX. So, please suggest me best
 solutions or way for achieve this requirement. If its possible on Asterisk
 then which version is use for that.

 Thanks in Advance.

 --
 Best Regards,

 Rajni Vanza
 Software Engineer
 ---
 Working On Linux,C/C++,VoIP Technology


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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Mitul Limbani]

Operator sends callerId after 1st small ring (actually this is not audible
since its very small duration ring) post which all the data flows.

However, sometimes due to line distrubance this first small ring is missed.

Regards,
Mitul Limbani,
Chief Architech  Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Fri, Sep 14, 2012 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote:

 On Fri, Sep 14, 2012 at 9:02 AM, Raj Mathur (राज माथुर) 
 r...@linux-delhi.org wrote:

 So if there's a good chance that the latest Asterisk and Dahdi packages
 will give better results in testing or might actually solve the problem,
 I'll be glad to compile from source.  If not, then perhaps it's not
 worth polluting a production box with locally-compiled packages.


 Try adding a Wait(2) between your NoOp and your Verbose lines.  I don't
 know about your telco, but sometimes the CID is not sent with the first
 ring, and you have to add a Wait(2) to grab it.

 You may even want to call your upper level support at your telco and ask
 them how and when they send your callerid information...


 --
 Thanks,
 --Warren Selby, dCAP
 http://www.SelbyTech.com http://www.selbytech.com


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