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

2014-03-13 Thread DHAVAL INDRODIYA
, experts on this list can guide to achieve objective. You can enable SIP trace on asterisk by executing following command in Asterisk console *sip set debug on* *Thanks Regards,* Amit Patkar On 3/12/2014 11:17 PM, DHAVAL INDRODIYA wrote: Thanks Amit, I want following scenario

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

2014-03-12 Thread DHAVAL INDRODIYA
Hello Group Members, I have one question regarding SIP-I/SIP-T support in any of Asterisk versions. We have client which send SIP-I/SIP-T request can asterisk handle it and serve as a normal SIP call. As per mine analysis SIP-I/SIP-T are variant of SIP protocol with adding of ISUP/SS7 packets

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

2014-03-12 Thread DHAVAL INDRODIYA
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

[asterisk-users] dahdi configuration issue

2013-09-04 Thread DHAVAL INDRODIYA
Hello List, I have configure 2 sangoma card each with 8 PRI lines with dahdi 2.6 the problem is i can see all channels configured in dahdi_cfg 480 channels configured but when I see /dev/dahdi i can only see 240 channels. what could be problem I am using it wanrouter and when I put PRI in new

Re: [asterisk-users] stop log/debug messages into /var/log/messages

2012-12-27 Thread DHAVAL INDRODIYA
please refer logger.conf under /etc/asterisk and stop messages log for full. On Thu, Dec 27, 2012 at 2:43 PM, [Digital^Dude] ® millennium@gmail.comwrote: I disabled all logger channels but still it logs to /var/log/messages. Any hints? --

Re: [asterisk-users] multitenanat third party app

2012-11-01 Thread DHAVAL INDRODIYA
Hi Carlos, you can get better idea after reading this. http://lists.digium.com/pipermail/asterisk-users/2007-August/193347.html Dhaval Indrodiya On Thu, Nov 1, 2012 at 5:36 AM, Carlos Alvarez car...@televolve.com wrote: Indeed this is getting ridiculous. This person also called me

Re: [asterisk-users] billsecs for call bridging

2012-10-16 Thread DHAVAL INDRODIYA
Before Dial application please call ResetCDR(v) application with option v. On Tue, Oct 16, 2012 at 12:40 PM, Ashish Agarwal ashisha...@gmail.comwrote: Hello, I have a dialplan using AGI where a user calls a number and an IVR is played. When the user presses 1, the system is suppose to call

[asterisk-users] can we install 10 PCI card on asterisk

2012-08-27 Thread DHAVAL INDRODIYA
Hi All, i would like to know if anyone has done or having idea regarding PRI terminations in asterisk. i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available now i am thinking of arranging 8port sangoma card in this pci slots

Re: [asterisk-users] can we install 10 PCI card on asterisk

2012-08-27 Thread DHAVAL INDRODIYA
[mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Monday, August 27, 2012 8:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] can we install 10 PCI card on asterisk Hi All, i would like to know if anyone has done

Re: [asterisk-users] recording calls

2012-08-22 Thread DHAVAL INDRODIYA
you need to provide dial plan for macro-one-touch-record. i think there is something which records outgoing only On Wed, Aug 22, 2012 at 6:39 AM, Josh Hopkins j...@prorivertech.com wrote: I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just

Re: [asterisk-users] FXO - GSM Gateway Problem

2012-04-18 Thread DHAVAL INDRODIYA
Hi, It can be codec negotiation error or else plese try to print hangupcause sent from telco On Wed, Apr 18, 2012 at 4:27 PM, Tech t...@digital-select.com wrote: Hi, ** ** I have a problem where calling out of asterisk when the call is answered dahdi hangs up immediately. For

[asterisk-users] Is Asterisk Support RFC-5168

2012-03-27 Thread DHAVAL INDRODIYA
Hi All, i am working on video setup within asterisk my simple question is asterisk support RFC-5168. if yes then in which version ? thanks Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] IVR: Dealing with database and returned variables

2012-03-07 Thread DHAVAL INDRODIYA
hi you can look following for better implementation http://phpagi.sourceforge.net/ in this you will find all your answer for get and set variable. cheers Dhaval On Thu, Mar 8, 2012 at 3:11 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; If I need to build IVR using Asterisk (so I will

Re: [asterisk-users] Dahdi Answer a Call On ringing State.

2012-02-17 Thread DHAVAL INDRODIYA
-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Friday, February 17, 2012 1:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi Answer a Call On ringing State. Hi Eric

Re: [asterisk-users] Dahdi Answer a Call On ringing State.

2012-02-16 Thread DHAVAL INDRODIYA
Hi Richard, i update a new version of asterisk to 1.8.9.1 and checked the issue are still same and my call getting answer while it is in ringing. here is brief details for finding root cause. Dahdi -Version: 2.4.1.2 Echo Canceller: OSLEC[channels] File : chan_dahdi.conf context=from-pstn

Re: [asterisk-users] Dahdi Answer a Call On ringing State.

2012-02-16 Thread DHAVAL INDRODIYA
. Use PRI or SIP if you need correct Answer supervision. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, February 16, 2012 6:28 AM To: Asterisk Users Mailing List - Non

[asterisk-users] Dahdi Answer a Call On ringing State.

2012-02-15 Thread DHAVAL INDRODIYA
Hi, I have setup Dahdi with Sangoma FXO A200 card and asterisk 1.8 , everything seems fine and working perfectly incoing/outgoing. but one major issue is, when i made an out call from dahdi trunks and when a number is in ringing state it gives me an answer state. so i cannot develop any custom

Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug

2012-02-14 Thread DHAVAL INDRODIYA
i tried it and it wont work with rtcachefriend=yes On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson jmr.richard...@gmail.comwrote: I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time architecture.when i am doing

Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.

2012-02-09 Thread DHAVAL INDRODIYA
nobody facing any issue with this or nobody using real time architecture On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Hi Group. I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time

[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.

2012-02-08 Thread DHAVAL INDRODIYA
Hi Group. I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time architecture.when i am doing registration it registered fine on asterisk as peer is available in Database. But now i am doing 'sip reload' or 'reload' due to

Re: [asterisk-users] execute command just after Dial()

2011-12-27 Thread DHAVAL INDRODIYA
You can also try special extension hangup and manage your scenario On Sat, Dec 24, 2011 at 12:44 PM, Sammy Govind govoi...@gmail.com wrote: Hi, Please see the Dial application documents from CLI, i.e core show application dial. There is an option which will let you continue in the DIal-plan

Re: [asterisk-users] execute command just after Dial()

2011-12-27 Thread DHAVAL INDRODIYA
so you can try with options of dial application g: Proceed with dialplan execution at the next priority in the current extension if the destination channel hangs up. G([[context^]exten^]priority): If the call is answered, transfer the calling party to the specified priority and the

Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-29 Thread DHAVAL INDRODIYA
Replace your phone number in place of ${EXTEN} and send it to your outgoing provider. with same dial argument. On Thu, Sep 29, 2011 at 3:09 PM, salaheddine elharit salah.elharit...@gmail.com wrote: ok thanks it's work fine now i have one question please it's work fine when i call

Re: [asterisk-users] How to know how many user is connected

2011-08-25 Thread DHAVAL INDRODIYA
Hi You can use simple cli command Manager show connected On Thursday, August 25, 2011, James zhu zhulizh...@live.com wrote: hi: please refer this: http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP and check the manager.conf, make sure the accounts in managers.conf matchs the

Re: [asterisk-users] Any Method for capturing ISUP packets in DAHDI/ASTERISK

2011-08-17 Thread DHAVAL INDRODIYA
configured on my machine. Regards Dhaval On Thu, Aug 11, 2011 at 11:21 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote: On Thu, Aug 11, 2011 at 12:43:43PM -0500, Russ Meyerriecks wrote: On Thu, Aug 11, 2011 at 11:57:46AM +0530, DHAVAL INDRODIYA wrote: Hi All, I want packets

Re: [asterisk-users] Any Method for capturing ISUP packets in DAHDI/ASTERISK

2011-08-16 Thread DHAVAL INDRODIYA
fault I have TE112 Card configured on my machine. Regards Dhaval On Thu, Aug 11, 2011 at 11:21 PM, Russ Meyerriecks rmeyerrie...@digium.comwrote: On Thu, Aug 11, 2011 at 12:43:43PM -0500, Russ Meyerriecks wrote: On Thu, Aug 11, 2011 at 11:57:46AM +0530, DHAVAL INDRODIYA wrote: Hi All

[asterisk-users] Any Method for capturing ISUP packets in DAHDI/ASTERISK

2011-08-11 Thread DHAVAL INDRODIYA
Hi All, I want packets [request/response] capture for ISUP packets , i have E1 line terminated to my digium card i just want a packets flow between my machine and teleco side, is any tool or utility [command] availabele for observation this packets and data. any help appericiated Thanks Dhaval

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

2011-07-29 Thread DHAVAL INDRODIYA
Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that you will receive in that , also read this for better implementation. http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause regards Dhaval On Fri, Jul 29, 2011 at 11:58 AM, Nikhil d.nik...@cem-solutions.net

Re: [asterisk-users] How to use these feature of Asterisk

2011-07-29 Thread DHAVAL INDRODIYA
Use This Information. You can customize the prompt a bit, if the default prompt is too dull for you. First add these lines to */etc/asterisk/extensions.conf* in the [globals] section: ${ENV(UNIX)} ${ENV(ASTERISK_PROMPT)} Then in */etc/profile* on the Asterisk server, set the ASTERISK_PROMPT

Re: [asterisk-users] Dialplan required for recording

2011-07-28 Thread DHAVAL INDRODIYA
Hi Vinod, You Need to look in MIxmonitor application on asterisk. http://www.voip-info.org/wiki/view/MixMonitor http://www.the-asterisk-book.com/unstable/applikationen-mixmonitor.html Where you can find easy dialplan On Fri, Jul 29, 2011 at 4:35 AM, Vinod Dharashive

[asterisk-users] Meaning Callerid Datatypes.

2011-07-25 Thread DHAVAL INDRODIYA
Hi All, can anybody have document og meaning with example of following CALLERID function data when we receive an incoming call Through PRI line and line E1 all name name-valid name-charset name-pres num num-valid num-plan num-pres subaddr

Re: [asterisk-users] Connect Avaya to Asterisk PBX

2011-07-13 Thread DHAVAL INDRODIYA
you can edit dial-plan by adding following lines to your code [internal] exten = s,1,Dial(SIP/1000) exten = s,2,HangUp() exten = 1000,1,Dial(SIP/1000) exten = 1000,2,HangUp() exten = _,1,Dial(H323/${EXTEN}@ Avaya) exten = _XXX,1,Dial(H323/${EXTEN}@Avaya) exten =

Re: [asterisk-users] DB Driven IVR

2011-07-10 Thread DHAVAL INDRODIYA
Hi, You can use combination of dial-plan and AGI for making DB driven IVR. On Sat, Jul 9, 2011 at 5:50 PM, G M gm.cu...@gmail.com wrote: Anyone has Experience ? On Fri, Jul 8, 2011 at 2:18 PM, G M gm.cu...@gmail.com wrote: I am using Vicidial and I am looking for someone who can help

Re: [asterisk-users] Problem in Detecting Dtmf on FXO line.

2011-07-10 Thread DHAVAL INDRODIYA
Hi, I tried with cid_rxgain,rxgain to put upto 5.0 and 10.0 values but not getting success. regards Dhaval On Fri, Jul 8, 2011 at 8:34 PM, Ruben Rögels ruben.roeg...@jumping-frog.org wrote: Am 08.07.2011 08:58, schrieb DHAVAL INDRODIYA: Hi All, I am having Problem in detecting DTMF

[asterisk-users] Problem in Detecting Dtmf on FXO line.

2011-07-08 Thread DHAVAL INDRODIYA
Hi All, I am having Problem in detecting DTMF on analog lines. basically are system is in india and telco provider is BSNL [Bharat sanchar Nigam LImited]. We have Purchased Analog card From chinaroby.com which is X1600P 16 port FXO card. they also provide us wctdm.c file. card is detected

Re: [asterisk-users] Problem in Detecting Dtmf on FXO line.

2011-07-08 Thread DHAVAL INDRODIYA
...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* Friday, July 08, 2011 11:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Problem in Detecting Dtmf on FXO line. ** ** Hi All, I am having Problem in detecting DTMF on analog lines. basically

Re: [asterisk-users] call file challenge...

2011-06-15 Thread DHAVAL INDRODIYA
Hi, I think you need to update *waittime* parameter in .call file please put atleast 10 seconds. for more understanding please try to read *http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out* Regards Dhaval On Wed, Jun 15, 2011 at 12:15 PM, Positively Optimistic

Re: [asterisk-users] asterisk queue 'ringall' stratagy

2011-06-13 Thread DHAVAL INDRODIYA
rajib, You can use DIALGROUP function as well On Mon, Jun 13, 2011 at 7:36 PM, Mike l...@net-wall.com wrote: Quite simply: don’t use a queue. Simply ring all phones at the same time using Dial(SIP/phone1SIP/phone2….) A queue will only send the first call until it is answered, then move

Re: [asterisk-users] no audio with SIP:INFO in meetme

2011-05-11 Thread DHAVAL INDRODIYA
Hi Rajib, There is nothing like that Asterisk is blocking an audio if you use without F it gives you and audio or not. cheers Dhaval On Wed, May 11, 2011 at 5:34 PM, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: Hello List, Asterisk is blocking audio if ‘F’ flag is enabled in

Re: [asterisk-users] asterisk HA for queue calls

2011-05-04 Thread DHAVAL INDRODIYA
Hi Rajib, I think It is not possible with asterisk , as primary server goes down it will stop asterisk services so once asterisk service down i think all connected calls to queue will hangup automatically, and you cannot retrive those calls as they all are disconnected . I think you need to

Re: [asterisk-users] Meetme Time Limit?

2011-04-21 Thread DHAVAL INDRODIYA
Hi, You can use Meetme(1234,dL(1800)) where 1800 = 6 hours after 6 hours channel is hanf up regards Dhaval On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman brya...@zktech.comwrote: Is there a way to place a hangup time on a dynamic Meetme conference. I am using Page() with a

Re: [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND

2011-04-20 Thread DHAVAL INDRODIYA
hey try with app_rpt in asterisk regards dhaval On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield t...@softins.co.ukwrote: In article 2658e54b540d284981ea57e6a549ea70abd1fdf...@inblrk77m1msx.in002.siemens.net , Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: The requirement is

Re: [asterisk-users] No voice in MeetMe for SIP with

2011-04-20 Thread DHAVAL INDRODIYA
Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA dhaval.it01...@gmail.com Subject: Re: [asterisk-users] No voice in MeetMe for SIP

Re: [asterisk-users] ConfBridge and AGI

2011-04-19 Thread DHAVAL INDRODIYA
Hi Rajib, this is your second post on Meetme with SIP channel and AGI script, Can you provide your requirement to run an AGI for Meetme , what you want to run an AGI with meetme. in confbridge there is nothing option for running AGI in background mode. let us know what you want to do exactly, on

Re: [asterisk-users] MeetMe headache

2011-04-06 Thread DHAVAL INDRODIYA
hey just change following [status-one-en] exten = 100,1,Meetme (12345,qdM) exten = 100,1,Hangup() Channel: Local/100@status-one-en CallerID: Rick 55 MaxRetries: 0 RetryTime: 15 WaitTime: 45 Application: Playback Data: my_status_message On Mon, Apr 4, 2011 at 10:38 PM, D. Rick

Re: [asterisk-users] agi voicemail callback

2011-04-06 Thread DHAVAL INDRODIYA
try this!!! http://www.voip-info.org/wiki/view/Asterisk+tips+callback On Wed, Apr 6, 2011 at 5:30 PM, vip killa vipki...@gmail.com wrote: What about a executing an AGI script with: [general] externnotify = /some_agi_script.agi Would that work? On Wed, Apr 6, 2011 at 3:20 AM, Thorsten

Re: [asterisk-users] Read VoiceMail direct

2011-04-06 Thread DHAVAL INDRODIYA
${CALLERID(num):-4} On Tue, Apr 5, 2011 at 2:53 AM, satish patel satish...@hotmail.com wrote: Perfect! Thanks what about :-4 ? I want to remove some digits -satish -- From: asannu...@gmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011

Re: [asterisk-users] ** to disconnect and make a new call

2011-04-01 Thread DHAVAL INDRODIYA
Hi, Please have a look on feature.conf and implement feature in [applicationmap] dialfeature = ##,peer,DIAL,{DATA_OF_DIAL_NUMBER_WITH_TECHNOLOGY} regards Dhaval On Thu, Mar 31, 2011 at 7:09 PM, Abid Saleem abid_aster...@hotmail.comwrote: Hi, Does anyone know how to implement the feature

Re: [asterisk-users] ** to disconnect and make a new call

2011-04-01 Thread DHAVAL INDRODIYA
it may be an argument of DIAL application for example if an user came into IVRS and then at some step presses ## if you decide that call is going to transfer to extension 1002 argument should be SIP/1002,30,r regards dhaval On Fri, Apr 1, 2011 at 3:03 PM, Abid Saleem

Re: [asterisk-users] Checking status of a cell phone

2011-03-31 Thread DHAVAL INDRODIYA
hi, if you only want that by any how you reachable , so you just make a simple DIALPLAN and do your work with DIal-plan, just DIAL those 3 numbers simuntenously, by seperating '', in that your cell number and other phone rings simultenously , and you can pick any of them other are automatically

Re: [asterisk-users] Dialplan help: hang up incoming call and call the number back

2011-03-29 Thread DHAVAL INDRODIYA
design your dial-plan for routing a specific number to different context , you can try func_odbc for query to DB if you have a large number of setup. ideally its called click to call but you are made it as, miss call and you will get a call. regards dhaval On Mon, Mar 28, 2011 at 5:21 PM, Roger

Re: [asterisk-users] Problems Extension with a Call In on Asterisk 1.6

2011-03-25 Thread DHAVAL INDRODIYA
Hi Olivier, here is solutions for your situation , ideally you need to talk with Provider and they can set SIP URI for given DID numbre , but that can be solved by dial-plan like this. exten = _003318364,1,Set(foo=${SIP_HEADER(To)}) exten = _003318364,n,Set(cut1=${CUT(foo,:,2)}) exten =

Re: [asterisk-users] Problems Extension with a Call In on Asterisk 1.6

2011-03-23 Thread DHAVAL INDRODIYA
Hi Oliver , This is a simple scenario with asterisk you can edit sip.conf and in peer entry, try to add, context=(desired_context for peer) and then into context write a dial-plan for given number and route a call or whatever you want to do. On Wed, Mar 23, 2011 at 11:31 AM, Olivier CALVANO

[asterisk-users] Is this true for Asterisk as SBC?

2011-03-10 Thread DHAVAL INDRODIYA
*Hi All, I have starting to reading About SBC and found one artical reagding SBC and they gives a solutions like this. i want to know is this true in realtime sceanario while we think of an big implementation and is it possible with cloud computing. i have found from

[asterisk-users] DIAL through Specific number in PRI

2011-02-23 Thread DHAVAL INDRODIYA
Hi ALL, I have PRI line everything is fine , but my customer having a requirement that they want to DIAL a number from PRI which gives callerid as Specific number. i.e PRI start from 3055 to 30550100 i have purchased a 100 number from telco and our pilot number is 3055, now if some

Re: [asterisk-users] Newbie´s question about Asterisk...

2011-02-18 Thread DHAVAL INDRODIYA
i prefer to go with Elastix very easy to setup and maintain and reach UI rather than freePBX cheers Dhaval On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote: Dean’s link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure.

Re: [asterisk-users] Meet me recording

2011-02-18 Thread DHAVAL INDRODIYA
Hi Satish, You can Pass 'r' flag to meetme Application and file will be recorded nothin to load mixmonitor and other Application on Channel, i think 'r' is better than all options Cheers Dhaval On Sat, Feb 19, 2011 at 1:37 AM, satish patel satish...@hotmail.com wrote: Thanks, look like

[asterisk-users] voice quality measurement using dahdi_monitor

2011-02-04 Thread DHAVAL INDRODIYA
hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and

Re: [asterisk-users] PRI voice optimization

2011-02-04 Thread DHAVAL INDRODIYA
there is a parameter for echo cancel in the card configuration, try disabling that because already you have enabled echo cancel in dahdi file. Hope it help.:) On Fri, Feb 4, 2011 at 11:11 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi All, This posting regarding PRI voice

[asterisk-users] PRI voice optimization

2011-02-03 Thread DHAVAL INDRODIYA
Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls

Re: [asterisk-users] Can a duration limit be specified in spool call file?

2011-01-29 Thread DHAVAL INDRODIYA
hi, what about this *WaitTime: number* Seconds to wait for an answer. Default is 45 http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out try out this regards Dhaval On Sat, Jan 29, 2011 at 6:19 AM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Friday 28 January 2011 18:27:15 Bruce B

[asterisk-users] Asterisk and Kamailio integration on cloud EC2 amazon no voice.

2011-01-24 Thread DHAVAL INDRODIYA
Hi All, i am stuck in NAT issue on ec2 cloud computing from last 2-3 days , may be some of you are doing setup and integration on cloud. below is my setup details which may help you to suggest me solution. Asterisk version : 1.6.2.6 1) Kamailio server having public_ip as well local ip .i am

Re: [asterisk-users] How to check a number online or offline

2011-01-11 Thread DHAVAL INDRODIYA
at 10:48 PM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: HI Phuong, JIM is right way but if you want to use extension state then there is a simple way of achiving through AMI, you need to fire this action on AMI and response have your answer , Please read about Action ExtensionState

Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread DHAVAL INDRODIYA
Hello , You can use Dialplan function DEVICE_STATE, which will gives you perfect status of DEVICE. regards Dhaval On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes steve-li...@geekinter.netwrote: On 10 Jan 2011, at 10:37, Phuong Hoang wrote: I found the link you have just sent to me but it do`nt

Re: [asterisk-users] How to check a number online or offline

2011-01-10 Thread DHAVAL INDRODIYA
other way to do this for me?thanks and looks forward to listening your reply. Regards! Phuong On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hello , You can use Dialplan function DEVICE_STATE, which will gives you perfect status of DEVICE. regards

Re: [asterisk-users] Call queues on load-balanced asterisks

2011-01-08 Thread DHAVAL INDRODIYA
Hello Pan, You can user DB for this just make real time configuration of Queue and make all asterisk server connected to Same DB if more load then use replication for different server on DB, also So that Quque name should be same for all server and asterisk can call same agent. you didnot

Re: [asterisk-users] Question About Conferencing Capabilities

2011-01-04 Thread DHAVAL INDRODIYA
Hi Siobhan, Asterisk is all capacity to work-on but you need to find out some way of handling conference system through WEB part , also one more thing on last point for switching between conference i am not much sure about it but i think it is possible if i will look into code implementation.

Re: [asterisk-users] Asterisk and Dahdi ON Amazon EC2

2010-12-15 Thread DHAVAL INDRODIYA
On Tue, Dec 14, 2010 at 5:23 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Dec 14, 2010 at 12:03:52PM +0100, Olivier wrote: 2010/12/14 DHAVAL INDRODIYA dhaval.it01...@gmail.com One More thing,once i installed dahdi-2.3.0 complete it installed successfully , but when i tried

Re: [asterisk-users] Asterisk and Dahdi ON Amazon EC2

2010-12-14 Thread DHAVAL INDRODIYA
...@xorcom.comwrote: On Tue, Dec 14, 2010 at 12:28:00PM +0530, DHAVAL INDRODIYA wrote: Hello Friends, I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86 version. DAHDI and asterisk: from packages or from source? What version of asterisk? and here is snap of uname

Re: [asterisk-users] Asterisk and Dahdi ON Amazon EC2

2010-12-14 Thread DHAVAL INDRODIYA
PM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Thanks For your reply, A in previous version we were used a dahdi-linux-2.1.0.4 and we were changed dahdi_dummy.c file as we are using on xen kernel we changed following things #if defined(__i386__) || defined(__x86_64__

[asterisk-users] Asterisk and Dahdi ON Amazon EC2

2010-12-13 Thread DHAVAL INDRODIYA
Hello Friends, I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86 version. and here is snap of uname- a command *Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200 x86_64 x86_64 x86_64 GNU/Linux* when I try to run DAHDI distribution

[asterisk-users] Asterisk 1.6.2.6 and ENUM LOOKUP? E.164

2010-11-11 Thread DHAVAL INDRODIYA
Hello, All i have one issue regarding caller id, once i received a call from my SIP provider it always set caller id with append 1 into original callerID if a call from USA then there is no problem , but if i receive a call from other country like INDIA i have also found callerID part as 191

[asterisk-users] Trixbox/Asterisk integration With SugarCRM

2010-11-07 Thread DHAVAL INDRODIYA
Hello All, i have one simple Question regarding integration of asterisk into sugar crm whether using trixbox or normal asterisk, can anyone have any link , forum or tutorial where i can find some information and some starting point . any help appreciated regards Dhaval --

[asterisk-users] ring delay and DTMF related problem in asterisk 1.6.2.6

2010-11-04 Thread DHAVAL INDRODIYA
Hi All, I am trying to call my own service through Asterisk and the DTMF is not recognized . I also noticed the following issue, the phone rings for about 8-9 times before the line is picked up but when it is picked up it seems that our system has picked up the call much earlier, I could just

[asterisk-users] Asterisk Strange Problem while call received from customer On PRI.

2010-10-27 Thread DHAVAL INDRODIYA
HI group, this is very strange problem with me when i received a call from Germany i am able to receive call on my PRI line everything is fine User connected with IVRS and user trying to enter a extension number like *1660976 *call goes to users company extension starting with *16.* is this

Re: [asterisk-users] Asterisk Realtime Billing Question???

2010-10-21 Thread DHAVAL INDRODIYA
sherwood.mcgo...@gmail.com wrote: On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hello All, after so long time i posted a new question regarding billing, hope anyone have some solution. I have situation in that i want to do billing of more than 1 call

Re: [asterisk-users] Asterisk Realtime Billing Question???

2010-10-21 Thread DHAVAL INDRODIYA
, regards Dhaval On Thu, Oct 21, 2010 at 12:49 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Thu, Oct 21, 2010 at 1:30 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi Sherwood , well , i think you did not understand my question , i want real time billing like as i

[asterisk-users] CDR record for call originated from CLI originate

2010-10-05 Thread DHAVAL INDRODIYA
hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval --

Re: [asterisk-users] CDR record for call originated from CLI originate

2010-10-05 Thread DHAVAL INDRODIYA
Regards, Arjan Kroon *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *DHAVAL INDRODIYA *Verzonden:* 05-10-2010 09:09 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] CDR record for call

Re: [asterisk-users] Speech To Text on linux with asterisk

2010-09-16 Thread DHAVAL INDRODIYA
Thanks for update if a file is converted to text then where can i find a text file like after running pocketsphinx_continuous command where text saved. regards dhaval On Thu, Sep 16, 2010 at 12:29 PM, Nickolay V. Shmyrev nshmy...@nexiwave.com wrote: В Чтв, 16/09/2010 в 10:35 +0530, DHAVAL

Re: [asterisk-users] Speech To Text on linux with asterisk

2010-09-14 Thread DHAVAL INDRODIYA
Thanks for update. is there any command for using sphinix to convert speech to text On Tue, Sep 14, 2010 at 1:18 PM, Nickolay V. Shmyrev nshmy...@nexiwave.comwrote: В Втр, 14/09/2010 в 01:55 -0400, Zeeshan Zakaria пишет: It is simply not possible, though it might be in the distant future.

Re: [asterisk-users] Speech To Text on linux with asterisk

2010-09-14 Thread DHAVAL INDRODIYA
-- www.ilovetovoip.com On 2010-09-14 4:52 AM, Nickolay V. Shmyrev nshmy...@nexiwave.com wrote: В Втр, 14/09/2010 в 14:00 +0530, DHAVAL INDRODIYA пишет: Thanks for update. is there any command for using sphinix to convert speech to text Yes, first of all make sure you compiled latest

Re: [asterisk-users] Speech To Text on linux with asterisk

2010-09-14 Thread DHAVAL INDRODIYA
/Communicator_semi_40.cd_semi_6000 -lm lm_giga_20k_nvp_3gram.lm.DMP *FATAL_ERROR: continuous.c, line 149: Failed to calibrate voice activity detection* regards Dhaval On Tue, Sep 14, 2010 at 8:40 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Sep 14, 2010 at 1:41 AM, DHAVAL INDRODIYA

[asterisk-users] Speech To Text on linux with asterisk

2010-09-13 Thread DHAVAL INDRODIYA
Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Speech To Text on linux with asterisk

2010-09-13 Thread DHAVAL INDRODIYA
. is this possible with lumenvox or any other engine. regards Dhaval On Tue, Sep 14, 2010 at 10:57 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Sep 14, 2010 at 1:01 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Is it possible to record say 30 seconds of audio and then have

[asterisk-users] How to Add IP address to SIP Domain

2010-06-29 Thread DHAVAL INDRODIYA
Dear All, I have Asterisk and Kamailio Configuration. everything works fine, now the situation is like i have following Dial pattern in Dialplan. exten = s,n, Dial(SIP/1...@glbvoice.com,20,m) now in my /etc/hosts i have following entry 192.168.1.30 glbvoice.com then call get forwarded to

[asterisk-users] [Asterisk-User] Asterisk Video support

2010-05-23 Thread DHAVAL INDRODIYA
Hi All, I am new to asterisk-video, is it possible to install video apps in 1.6.2.6 and play live video calls on weburl? please help me i dont have much idea for asterisk-video even dont know about installation. any help appreciated regards Dhaval --

Re: [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.

2010-05-19 Thread DHAVAL INDRODIYA
is broken. On Tue, May 18, 2010 at 11:46 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi Motiejus, sorry for inconvenience , because asterisk mailing list could not accept wav file attachment here i am attached a file named test.wav, regards Dhaval 2010/5/18 Motiejus

[asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.

2010-05-18 Thread DHAVAL INDRODIYA
hello All, i have one issue with Asterisk Meetme Application i am recording through Meetme channels through option *'r'* and format for recording a file is '*wav*' lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5. i have very strange problem of meetme_recording , once

[asterisk-users] Calling a RESTful Web service from Dialplan????

2010-05-02 Thread DHAVAL INDRODIYA
Dear All, Last Week i tried and goggling more on how to call RESTful webservice from Dialplan? i found *CURL* function but while i tried to use it ,it 's not supported HTTPS request and we cannot set headers while send a request. also without HTTPS . i get result it will return a string

[asterisk-users] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM

2010-03-17 Thread DHAVAL INDRODIYA
of causing this error, is there any configuration needed. or is there any settings needed for safe_asterisk . because this is running in production environment. regards Dhaval Indrodiya. -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM

2010-03-17 Thread DHAVAL INDRODIYA
- Hash: SHA1 DHAVAL INDRODIYA wrote: where as I am using Asterisk 1.6.0.5 and my machine is using *safe_asterisk* script asterisk running Why are you using such an old version in the 1.6.0 branch? 1.6.0.25 is current, upgrade to there and then worry about the problem if it recurs

[asterisk-users] PBX_DUNDI question

2010-03-13 Thread DHAVAL INDRODIYA
hello All, what could be the problem in dundi lookup *pbx_dundi.c:4109 dundi_result_read: Result number 1 is not valid for DUNDi query results for ID 879!* though it should return some results , it failed in getting those . foloowing is my DIALPLAN exten =

[asterisk-users] Echo cancellation on DAHDI

2010-03-02 Thread DHAVAL INDRODIYA
Dear All, How can we know the On board supports echo cancellation I have *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02)*board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval --

Re: [asterisk-users] Echo cancellation on DAHDI

2010-03-02 Thread DHAVAL INDRODIYA
, 2010 at 10:25 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Tue, 2010-03-02 at 15:44 +0530, DHAVAL INDRODIYA wrote: Dear All, How can we know the On board supports echo cancellation I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) board all working fine

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

2010-01-06 Thread DHAVAL INDRODIYA
hi arun can you paste a dialplan here and version of asterisk regards dhaval On Thu, Jan 7, 2010 at 11:51 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Jan 07, 2010 at 09:54:09AM +0530, Arun Sasidhar wrote: hi, I made changes in zapata.conf but no result. You use

Re: [asterisk-users] AMI originate and PHP

2009-12-28 Thread DHAVAL INDRODIYA
Hi, Bruce , would you remove Async from your php script, and give it a try regards Dhaval On Thu, Dec 24, 2009 at 5:45 AM, Bruce Nik brucev...@gmail.com wrote: Jarrod, Thanks for the input. Can you please include a sample of your work? It will really save me days of headache and tests if I

[asterisk-users] Asterisk Heartbeat Monitor for Fail safe.

2009-12-20 Thread DHAVAL INDRODIYA
Dear All, I want to configure Asterisk/Kamailio Like system monitor with Heartbeat is there any way to monitor Service If NODE1 is stopped or over loaded then NODE 2 will work and vice verse. any help appreciated because i m stuck in heartbeat to configure service. regards Dhaval

Re: [asterisk-users] Asterisk Heartbeat Monitor for Fail safe.

2009-12-20 Thread DHAVAL INDRODIYA
Thanks Alex, regards Dhaval On Mon, Dec 21, 2009 at 11:52 AM, Alex Balashov abalas...@evaristesys.comwrote: On 12/21/2009 12:24 AM, DHAVAL INDRODIYA wrote: I want to configure Asterisk/Kamailio Like system monitor with Heartbeat There is, but Asterisk/Kamailio-like system

[asterisk-users] SIP_CODEC related question

2009-12-09 Thread DHAVAL INDRODIYA
hello ALL, My question is regarding SIP_CODEC. 1). How can I get which codec is used for this channel . Ex: if incoming call to asterisk i want to know which codec is used for this channel. is there any way for printing codec in dial plan 2). How can I set codec for outbound

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