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

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

Re: [asterisk-users] Using g729 now that patents have expired

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

Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

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>

Re: [asterisk-users] Asterisk Development Company in India

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

Re: [asterisk-users] 1000 analogue lines with asterisk

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>

Re: [asterisk-users] 1000 analogue lines with asterisk

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 > > -- > _ > --

Re: [asterisk-users] No sound with internal calls depending on which phones

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

Re: [asterisk-users] Fwd: Sublime Text License Key

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: > > --

Re: [asterisk-users] Tdm4010p 4 port card

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

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

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

Re: [asterisk-users] German sounds on Asterisk

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)

Re: [asterisk-users] asterisk google contacts

://lists.digium.com/mailman/listinfo/asterisk-users -- 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

Re: [asterisk-users] Am I cracked?

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

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

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

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

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

Re: [asterisk-users] OpenVZ with asterisk 13

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

Re: [asterisk-users] OpenVZ with asterisk 13

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

Re: [asterisk-users] OpenVZ with asterisk 13

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

Re: [asterisk-users] OpenVZ with asterisk 13

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

Re: [asterisk-users] OpenVZ with asterisk 13

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

Re: [asterisk-users] WebRTC demo phones

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

Re: [asterisk-users] Music on hold

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

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

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

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

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

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

Re: [asterisk-users] One way audio internal

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

Re: [asterisk-users] ITSP Gateway Solution?

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

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

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

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

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,

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

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

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

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

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

://lists.digium.com/mailman/listinfo/asterisk-users -- 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

Re: [asterisk-users] if statement recording - after hours

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)

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

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

Re: [asterisk-users] packet2packet bridging

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

Re: [asterisk-users] packet2packet bridging

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

Re: [asterisk-users] recording in mp3

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

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

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

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

, 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

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

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

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

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

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

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

Re: [asterisk-users] Webrtc and adventures with Asterisk 11

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

Re: [asterisk-users] SIP fraud IP blacklist

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

Re: [asterisk-users] Regarding SIP-T/SIP-I support in Asterisk.

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)

Re: [asterisk-users] High Availability with Asterisk

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

Re: [asterisk-users] Enterprise VoIP Trunk

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

Re: [asterisk-users] Dialer software for Asterisk...

://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mitul Limbani

Re: [asterisk-users] Connect to remote GW

. 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

Re: [asterisk-users] Asterisk on Windows

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

Re: [asterisk-users] Asterisk on Windows

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

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

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-

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

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

Re: [asterisk-users] Bulk forwarding to another Asterisk

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

Re: [asterisk-users] Asterisk 1.8.22

? 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

Re: [asterisk-users] TE420, is it possible do disable span (red blinking)?

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

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

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

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

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

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

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

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

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

Re: [asterisk-users] MDL-ERROR

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

Re: [asterisk-users] using E1 PRI lines

/asterisk-users -- 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

Re: [asterisk-users] IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6

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

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

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

Re: [asterisk-users] SIGTRAN Integration

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http

Re: [asterisk-users] SIGTRAN Integration

://lists.digium.com/mailman/listinfo/asterisk-users -- 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

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

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

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

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

Re: [asterisk-users] Stress testing Asterisk

Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani

Re: [asterisk-users] Performance Asterisk large installation on Vmware/Xen

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

Re: [asterisk-users] dial and bridge

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

Re: [asterisk-users] dial and bridge

, 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

Re: [asterisk-users] dial and bridge

, 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

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

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

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

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

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

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

Re: [asterisk-users] asterisk sizing for play and dtmf detection

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

Re: [asterisk-users] digium card and virualbox

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

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

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

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

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

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

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 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] DIDForSale spam

+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.

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

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

Re: [asterisk-users] SIP password probe

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

Re: [asterisk-users] leading ghost 0

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,

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

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

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

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,

Re: [asterisk-users] multitenanat third party app

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:

Re: [asterisk-users] multi tenant

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

Re: [asterisk-users] Cant we have same extension in different context?

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

Re: [asterisk-users] high capacity analog - sip gateway

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

Re: [asterisk-users] dahdi dummy

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

Re: [asterisk-users] Counting calls in progress from AMI

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:*

Re: [asterisk-users] asterisk as IVR using 3g usb modem

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

Re: [asterisk-users] Issue with PRI connection

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

Re: [asterisk-users] Issue with PRI connection

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

Re: [asterisk-users] Issue with PRI connection

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

Re: [asterisk-users] Issue with PRI connection

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

Re: [asterisk-users] Need to Help for setup Video IVRS on Asterisk

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

Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

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

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