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

[DHAVAL INDRODIYA]

Amit,

I know how to play with SIP in asterisk and other tools . I want to know
weather asterisk natively support or is there any extra patch or any
workaround for SIP-T/SIP-I.

Regarding packets and other things I am still not integrating it . I am
searching some open-source tool which can send generate this type of
packets and structure .

Once I will integrate to our provider I will definitely check and share
with experts here.








On Thu, Mar 13, 2014 at 11:13 AM, Amit a...@avhan.com wrote:

  Hi Dhaval,

 If you capture and share SIP traces for inbound and outbound calls
 separately, 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.

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

[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 to original SIP request.


If we want to support it then how do we implement it and support it with
asterisk . is there any open-source package or tool available to
communicate and works as SIP-T to SIP and SIP to SIP-T gateway. I got a
reference from kamailio which have SIPT module in latest version is anyone
had worked or having an idea regarding this module and its operations .

Hope any one worked and having some idea

Any help appreciated


Thanks

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

[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 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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[asterisk-users] dahdi configuration issue

[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
card i only got calls on new line that means one of the card is inactive at
same time all the lines and alarms are okay only suspected thing is
/dev/dahdi.

is there nany setting in linux or kernel level which need to be set for
solve this issue.

any help appreciated.

Thanking You

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

[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?

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Re: [asterisk-users] multitenanat third party app

[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 (!!) for
 some free consulting after I had posted the answer a few days ago.

 NOTE:  We aren't going to engineer your system for you!  We as a group
 will provide help and some basic code to get you started.  If you don't
 know how to start working with the fully working stuff I provided already,
 you're not ready to deploy a system this complex.


 On Wed, Oct 31, 2012 at 2:59 PM, Mitul Limbani mi...@enterux.in wrote:

 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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 --
 Carlos Alvarez
 TelEvolve
 602-889-3003



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Re: [asterisk-users] billsecs for call bridging

[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 another
 number and bridge the call. I am able to do this successfully, but I want
 to know the billsecs of the first caller and also the second call from the
 time it was answered by the second user. I am able to successfully get the
 first call time but not the second one.

 Can someone guide me on this.

 --
 Regards,

 Ashish Agarwal

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

[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 so i
can arrenge 10 card in that.

is it possible to run system like that ? is it good idea , can asterisk
handle 2400 calls if machine size and RAM is good.

let me know ideas and suggestions.

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

[DHAVAL INDRODIYA]

Hey All,

Thanks for everyone input on this, this was just mine thoughts to put 80
PRI line in that.but after reading inputs from everyone i think there are
some options to achieve it.

it means i need to put a gateway which convert my SIP calls to PRI line and
another options is to put
multiple asterisk boxes and each box have maximum 16 pri lines . now which
is best choice to work on further. also i need to consider hardware sizing
too as if gateway is expensive i would go with pri cards.
also if i choose gateway then  also i need to put multiple asterisk boxes.

let me know your thoughts.

thanks
Dhaval

On Mon, Aug 27, 2012 at 10:54 PM, Eric Wieling ewiel...@nyigc.com wrote:

 Your best bet is a carrier class device from someone like Adtran and
 convert the PRIs to SIP before passing the calls to Asterisk.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [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 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 so i
 can arrenge 10 card in that.

 is it possible to run system like that ? is it good idea , can asterisk
 handle 2400 calls if machine size and RAM is good.

 let me know ideas and suggestions.

 thanks
 Dhaval


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Re: [asterisk-users] recording calls

[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 fine but not inbound calls or calls from an
 internally between extensions.   I am using the latest asterisk 1.8.x
 certified version.

 ** **

 On an outbound call I see:

 ** **

 == Using SIP RTP CoS mark 5

 -- Called SIP/ BVTrunk /719000

 -- SIP/BVTrunk-0163 is making progress passing it to
 SIP/1010-0162

 -- SIP/BVTrunk-0163 answered SIP/1010-0162

 --  Feature Found: apprecord exten: apprecord

 -- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0162,
 0?Set(THISEXTEN=719)) in new stack

 -- Executing [s@macro-one-touch-record:2] ExecIf(SIP/1010-0162,
 1?Set(THISEXTEN=1010)) in new stack

 -- Executing [s@macro-one-touch-record:3] ExecIf(SIP/1010-0162,
 0?MacroExit()) in new stack

 -- Executing [s@macro-one-touch-record:4] GotoIf(SIP/1010-0162,
 0?stoprec) in new stack

 -- Executing [s@macro-one-touch-record:5] GotoIf(SIP/1010-0162,
 0?stopped) in new stack

 -- Executing [s@macro-one-touch-record:6] GotoIf(SIP/1010-0162,
 0?recording) in new stack

 -- Executing [s@macro-one-touch-record:7] Set(SIP/1010-0162,
 MASTER_CHANNEL(ONETOUCH_REC)=RECORDING) in new stack

 -- Executing [s@macro-one-touch-record:8] Set(SIP/1010-0162,
 MASTER_CHANNEL(REC_STATUS)=RECORDING) in new stack

 -- Executing [s@macro-one-touch-record:9] Set(SIP/1010-0162,
 AUDIOHOOK_INHERIT(MixMonitor)=yes) in new stack

 -- Executing [s@macro-one-touch-record:10]
 MixMonitor(SIP/1010-0162,
 2012/08/21/out-719000-1010-20120821-183119-1345595479.530.wav,a,) in
 new stack

   == Begin MixMonitor Recording SIP/1010-0162

 -- Executing [s@macro-one-touch-record:11] Set(SIP/1010-0162,
 MON_FMT=wav) in new stack

 -- Executing [s@macro-one-touch-record:12] Set(SIP/1010-0162,
 MASTER_CHANNEL(CDR(recordingfile))=out-719000-1010-20120821-183119-1345595479.530.wav)
 in new stack

 -- Executing [s@macro-one-touch-record:13] Set(SIP/1010-0162,
 MASTER_CHANNEL(ONETOUCH_RECFILE)=out-719000-1010-20120821-183119-1345595479.530.wav)
 in new stack

 -- Executing [s@macro-one-touch-record:14]
 Playback(SIP/1010-0162, beep) in new stack

 -- SIP/1010-0162 Playing 'beep.ulaw' (language 'en')

 -- Executing [s@macro-one-touch-record:15]
 MacroExit(SIP/1010-0162, ) in new stack

 -- Executing [h@macro-dialout-trunk:1] Macro(SIP/1010-0162,
 hangupcall,) in new stack

 -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1010-0162,
 1?theend) in new stack

 -- Goto (macro-hangupcall,s,3)

 -- Executing [s@macro-hangupcall:3] ExecIf(SIP/1010-0162,
 1?Set(CDR(recordingfile)=out-719000-1010-20120821-183119-1345595479.530.wav))
 in new stack

 -- Executing [s@macro-hangupcall:4] Hangup(SIP/1010-0162, )
 in new stack

   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
 'SIP/1010-0162' in macro 'hangupcall'

   == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on
 'SIP/1010-0162'

   == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on
 'SIP/1010-0162' in macro 'dialout-trunk'

   == Spawn extension (from-internal, 719000, 6) exited non-zero on
 'SIP/1010-0162'

   == MixMonitor close filestream

   == End MixMonitor Recording SIP/1010-0162

   == Extension Changed 1010[ext-local] new state Idle for Notify User 1004
 

 ** **

 On inbound calls I see:

 ** **

 == Using SIP RTP CoS mark 5

 -- Called SIP/1010

 -- Connected line update to SIP/ BVTrunk -0160 prevented.

   == Extension Changed 1010[ext-local] new state Ringing for Notify User
 1004

 -- SIP/1010-0161 is ringing

 -- Connected line update to SIP/ BVTrunk -0160 prevented.

 -- SIP/1010-0161 answered SIP/ BVTrunk -0160

   == Extension Changed 1010[ext-local] new state InUse for Notify User 1004
 

 -- Executing [s@macro-auto-blkvm:1] Set(SIP/1010-0161,
 __MACRO_RESULT=) in new stack

 -- Executing [s@macro-auto-blkvm:2] Macro(SIP/1010-0161,
 blkvm-clr,) in new stack

 -- Executing [s@macro-blkvm-clr:1] Set(SIP/1010-0161,
 SHARED(BLKVM,SIP/BVTrunk-0160)=) in new stack

 -- Executing [s@macro-blkvm-clr:2] Set(SIP/1010-0161,
 GOSUB_RETVAL=) in new stack

 -- Executing [s@macro-blkvm-clr:3] MacroExit(SIP/1010-0161, )
 in new stack

 -- Executing [s@macro-auto-blkvm:3] ExecIf(SIP/1010-0161,
 

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

[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 example: Sip Client A calls external number. Route: SIP - FXO - GSM
 Gateway -External Landline.

 However when that external landline answers the call dahdi hangs up
 immediately .

 ** **

 Going the other way is fine (External Landline - GSM Gateway - FXO -
 SIP).

 ** **

 I've tried multiple different internet searches and can't seem to find any
 information on this problem.

 ** **

 Below are my config files.

 ** **

 *Sip.conf*

 [office-phone](!)  

 type=friend 

 context=sipofficephone   

 host=dynamic

 nat=yes 

 #secret= 

 dtmfmode=auto   

 disallow=all

 ;allow=ulaw  

 allow=alaw  

 allow=GSM

 ** **

 [lewisphone](office-phone);lewis mobile

 secret=

 ** **

 *Chan_dahdi.conf*

 [channels]

 signalling=fxs_ks 

 context=pstnincomming

 group=0

 channel = 1

 ** **

 ** **

 *Extensions.conf*

 [sipofficephone]

 exten = _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})

 same = n,Dial(DAHDI/1/${EXTEN})

 same = n,Hangup()

 ** **

 [pstnincomming]Diamon

 exten = s,1,Answer()

 same = n,Dial(SIP/lewisphone)

 same = n,Hangup()

 ** **

 ** **

 *Asterisk CLI Output (Verbose 3)*

 My comments bold.

 ** **

   == Using SIP RTP CoS mark 5

 -- Executing [@sipofficephone:1]
 Verbose(SIP/lewisphone-000a, 2,Call from VoIP network to ) in
 new stack

   == Call from VoIP network to 

 -- Executing [@sipofficephone:2] Dial(SIP/lewisphone-000a,
 DAHDI/1/) in new stack

 -- Called DAHDI/1/

 -- DAHDI/1-1 answered SIP/lewisphone-000a *GSM Gateway Answering
 Call then Sending it out.*

 -- Hanging up on 'DAHDI/1-1' *Dest answering call to which DAHDI
 hangs up*

 -- Hungup 'DAHDI/1-1'

   == Spawn extension (sipofficephone, , 2) exited non-zero on
 'SIP/lewisphone-000a'

 ** **

 ** **

 ** **

 Best Regards

 *

 *

 Lewis 

 [image: digitalselect-e]

 www.Digital-Select.com http://www.digital-select.com/

 *

 *

 ** **

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[asterisk-users] Is Asterisk Support RFC-5168

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

[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 read and write to
 database), until now from my reading, I can understand that the best way is
 to use AGI to call external script like php which will manipulate every
 thing, correct?

 Well, the returned values from this script that I can use it to route the
 call to the proper queue or Phone, how I can handle these returned values?
 Do I have to store it in the database? Well, how I will read it from
 database and use it in the extensions.conf?

 From the other side, is there any tool to have IVR script (let us say,
 studio programing) that can be used in Asterisk? Any advise in this way?

 Regards
 Bilal

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

[DHAVAL INDRODIYA]

Yes, it is telco ringing and asterisk answered that line .

thanks
Dhaval

On Fri, Feb 17, 2012 at 8:35 PM, Eric Wieling ewiel...@nyigc.com wrote:

 What does the CLI show?  The ringing you year is likely the telco ringing,
 not the asterisk ringing.

 -Original Message-
 From: asterisk-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,

 but in this case dialing is not completed ring is still going on, so it
 should not answered

 thanks
 Dhaval


 On Thu, Feb 16, 2012 at 7:52 PM, Eric Wieling ewiel...@nyigc.com wrote:


FXO ports are considered Answered as soon as dialing completes.
  This is the way analog FXO ports work.  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-Commercial Discussion
Subject: Re: [asterisk-users] Dahdi Answer a Call On ringing State.

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
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
usecallerid=yes
callerid=asreceived
cidstart=polarity_in
cidsignalling=dtmf
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callprogress=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
relaxdtmf=yes
pulsedial=yes

;Uncomment these lines if you have problems with the disconection
 of your analog lines busydetect=yes
busycount=3
immediate=no
answeronpolarityswitch=yes
polarityonanswerdelay=1000

group=0
channel = 1

group=1
channel = 2

group=0
channel = 3

group=0
channel = 4



Let me know your thoughts on this

thanks
Dhaval





On Thu, Feb 16, 2012 at 1:47 PM, Richard Mudgett 
 rmudg...@digium.com wrote:


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.


   This was recently fixed by
   https://issues.asterisk.org/jira/browse/ASTERISK-18841


so i cannot develop any custom application which can use a
 screening
macro because when a cellphone is in ringing state
call is answered by dahdi channel so it will start
 executing dial
plan which causes an issue.


   Richard

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

[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
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
usecallerid=yes
callerid=asreceived
cidstart=polarity_in
cidsignalling=dtmf
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callprogress=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
relaxdtmf=yes
pulsedial=yes

;Uncomment these lines if you have problems with the disconection of your
analog lines
busydetect=yes
busycount=3
immediate=no
answeronpolarityswitch=yes
polarityonanswerdelay=1000

group=0
channel = 1

group=1
channel = 2

group=0
channel = 3

group=0
channel = 4



Let me know your thoughts on this

thanks
Dhaval




On Thu, Feb 16, 2012 at 1:47 PM, Richard Mudgett rmudg...@digium.comwrote:

  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.

 This was recently fixed by
 https://issues.asterisk.org/jira/browse/ASTERISK-18841

  so i cannot develop any custom application which can use a screening
  macro because when a cellphone is in ringing state
  call is answered by dahdi channel so it will start executing dial
  plan which causes an issue.

 Richard

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

[DHAVAL INDRODIYA]

Hi Eric,

but in this case dialing is not completed ring is still going on, so it
should not answered

thanks
Dhaval

On Thu, Feb 16, 2012 at 7:52 PM, Eric Wieling ewiel...@nyigc.com wrote:

 FXO ports are considered Answered as soon as dialing completes.  This is
 the way analog FXO ports work.  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-Commercial Discussion
 Subject: Re: [asterisk-users] Dahdi Answer a Call On ringing State.

 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
 signalling=fxs_ks
 rxwink=300  ; Atlas seems to use long (250ms) winks
 usecallerid=yes
 hidecallerid=no
 usecallerid=yes
 callerid=asreceived
 cidstart=polarity_in
 cidsignalling=dtmf
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 callprogress=yes
 echocancel=yes
 echocancelwhenbridged=no
 faxdetect=incoming
 echotraining=800
 rxgain=0.0
 txgain=0.0
 callgroup=1
 pickupgroup=1
 relaxdtmf=yes
 pulsedial=yes

 ;Uncomment these lines if you have problems with the disconection of your
 analog lines busydetect=yes
 busycount=3
 immediate=no
 answeronpolarityswitch=yes
 polarityonanswerdelay=1000

 group=0
 channel = 1

 group=1
 channel = 2

 group=0
 channel = 3

 group=0
 channel = 4



 Let me know your thoughts on this

 thanks
 Dhaval





 On Thu, Feb 16, 2012 at 1:47 PM, Richard Mudgett rmudg...@digium.com
 wrote:


 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.


This was recently fixed by
https://issues.asterisk.org/jira/browse/ASTERISK-18841


 so i cannot develop any custom application which can use a
 screening
 macro because when a cellphone is in ringing state
 call is answered by dahdi channel so it will start executing dial
 plan which causes an issue.


Richard

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

[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 application which can use a screening macro
because when a cellphone is in ringing state
call is answered by dahdi channel so it will start executing dial plan
which causes an issue.

let me know if there is any parameter from which i can set in
chan_dahdi.conf and check if it worked or not.

Note: I am from INDIA and line is from BSNL

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

[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 registration it registered fine on asterisk
  as peer is available in Database.
 
  But now i am doing 'sip reload' or 'reload' due to some reason my peer
  registration is going out and i cannot able to call that peer even though
  in SIP client it shows me 'registered'.
 
  Can any body elaborate on this issue which settings i need to put in
  sip.conf.
 
  I also tried to follow this patch
  https://issues.asterisk.org/view.php?id=14196 But it allready applied in
  code base so why it wont work?
 
  Here is my sip.conf settings.
 
  [general]
  context=from-internal; Default context for incoming cal
  rtcachefriends=no
  rtupdate=yes
  rtautoclear=yes
  rtsavesysname=yes
  callcounter = yes
  callevents=yes
  bindport=5060; UDP Port to bind to (SIP standard port is
 5060)
  srvlookup=yes; Enable DNS SRV lookups on outbound calls
  pedantic=yes; Enable slow, pedantic checking for Pingtel
  tos=184; Set IP QoS to either a keyword or numeric val
  tos_sip=cs3; Sets TOS for SIP packets.
  tos_audio=ef   ; Sets TOS for RTP audio packets.
  tos=lowdelay; lowdelay,throughput,reliability,mincost,none
  maxexpiry=3600; Max length of incoming registration we allow
  defaultexpiry=120; Default length of incoming/outoing
 registration
  preferred_codec_only=yes
  disallow=all; First disallow all codecs
  allow=ulaw; Allow codecs in order of preference
  allow=alaw
  insecure=invite
  language=en   ; Default language setting for all
  users/peers
  rtpholdtimeout=300; Terminate call if 300 seconds of no RTP
  activity
  useragent=dhaval  ; Allows you to change the user agent
 string
  dtmfmode = rfc2833; Set default dtmfmode for sending DTMF.
 Default:
  rfc2833
  qualify=yes
  nat=yes
  ;canreinvite=yes
  directmedia=yes
  directrtpsetup=yes
 
  And here is DB fields snapshots.
 
id: 1
  name: 201
ipaddr: 172.18.100.243
  port: 53624
regseconds: 1328716180
   defaultuser: 201
   fullcontact: NULL
 regserver: dhaval
 useragent: CSipSimple r1133 / b
lastms: 554
  host: dynamic
  type: friend
   context: from-internal
permit: NULL
  deny: NULL
secret: 201
 md5secret: NULL
  remotesecret: NULL
 transport: NULL
  dtmfmode: NULL
   directmedia: yes
   nat: NULL
 allow: ulaw
  disallow: g729
  insecure: invite
  callerid: NULL
  rfc2833compensate: NULL
   mailbox: NULL
session-timers: NULL
   session-expires: NULL
 session-minse: NULL
  session-refresher: NULL
 
  Kindly help me to resolve this.
 
  Thanks
  Dhaval
 

 The first thing I would try is 'rtcachefriends=yes', that should do it.

 JR
 --
 JR Richardson
 Engineering for the Masses

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

[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
 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 some reason my peer
 registration is going out and i cannot able to call that peer even though
 in SIP client it shows me 'registered'.

 Can any body elaborate on this issue which settings i need to put in
 sip.conf.

 I also tried to follow this patch
 https://issues.asterisk.org/view.php?id=14196 But it allready applied in
 code base so why it wont work?

 Here is my sip.conf settings.


 [general]
 context=from-internal; Default context for incoming cal
 rtcachefriends=no
 rtupdate=yes
 rtautoclear=yes
 rtsavesysname=yes
 callcounter = yes
 callevents=yes
 bindport=5060; UDP Port to bind to (SIP standard port is 5060)
 srvlookup=yes; Enable DNS SRV lookups on outbound calls
 pedantic=yes; Enable slow, pedantic checking for Pingtel
 tos=184; Set IP QoS to either a keyword or numeric val
 tos_sip=cs3; Sets TOS for SIP packets.
 tos_audio=ef   ; Sets TOS for RTP audio packets.
 tos=lowdelay; lowdelay,throughput,reliability,mincost,none
 maxexpiry=3600; Max length of incoming registration we allow
 defaultexpiry=120; Default length of incoming/outoing registration
 preferred_codec_only=yes
 disallow=all; First disallow all codecs
 allow=ulaw; Allow codecs in order of preference
 allow=alaw
 insecure=invite
 language=en   ; Default language setting for all
 users/peers
 rtpholdtimeout=300; Terminate call if 300 seconds of no RTP
 activity
 useragent=dhaval  ; Allows you to change the user agent string
 dtmfmode = rfc2833; Set default dtmfmode for sending DTMF.
 Default: rfc2833
 qualify=yes
 nat=yes
 ;canreinvite=yes
 directmedia=yes
 directrtpsetup=yes

 And here is DB fields snapshots.

id: 1
  name: 201
ipaddr: 172.18.100.243
  port: 53624
regseconds: 1328716180
   defaultuser: 201
   fullcontact: NULL
 regserver: dhaval
 useragent: CSipSimple r1133 / b
lastms: 554
  host: dynamic
  type: friend
   context: from-internal
permit: NULL
  deny: NULL
secret: 201
 md5secret: NULL
  remotesecret: NULL
 transport: NULL
  dtmfmode: NULL
   directmedia: yes
   nat: NULL
 allow: ulaw
  disallow: g729
  insecure: invite
  callerid: NULL
 rfc2833compensate: NULL
   mailbox: NULL
session-timers: NULL
   session-expires: NULL
 session-minse: NULL
 session-refresher: NULL


 Kindly help me to resolve this.

 Thanks
 Dhaval


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

[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 some reason my peer
registration is going out and i cannot able to call that peer even though
in SIP client it shows me 'registered'.

Can any body elaborate on this issue which settings i need to put in
sip.conf.

I also tried to follow this patch
https://issues.asterisk.org/view.php?id=14196 But it allready applied in
code base so why it wont work?

Here is my sip.conf settings.


[general]
context=from-internal; Default context for incoming cal
rtcachefriends=no
rtupdate=yes
rtautoclear=yes
rtsavesysname=yes
callcounter = yes
callevents=yes
bindport=5060; UDP Port to bind to (SIP standard port is 5060)
srvlookup=yes; Enable DNS SRV lookups on outbound calls
pedantic=yes; Enable slow, pedantic checking for Pingtel
tos=184; Set IP QoS to either a keyword or numeric val
tos_sip=cs3; Sets TOS for SIP packets.
tos_audio=ef   ; Sets TOS for RTP audio packets.
tos=lowdelay; lowdelay,throughput,reliability,mincost,none
maxexpiry=3600; Max length of incoming registration we allow
defaultexpiry=120; Default length of incoming/outoing registration
preferred_codec_only=yes
disallow=all; First disallow all codecs
allow=ulaw; Allow codecs in order of preference
allow=alaw
insecure=invite
language=en   ; Default language setting for all users/peers
rtpholdtimeout=300; Terminate call if 300 seconds of no RTP activity
useragent=dhaval  ; Allows you to change the user agent string
dtmfmode = rfc2833; Set default dtmfmode for sending DTMF. Default:
rfc2833
qualify=yes
nat=yes
;canreinvite=yes
directmedia=yes
directrtpsetup=yes

And here is DB fields snapshots.

   id: 1
 name: 201
   ipaddr: 172.18.100.243
 port: 53624
   regseconds: 1328716180
  defaultuser: 201
  fullcontact: NULL
regserver: dhaval
useragent: CSipSimple r1133 / b
   lastms: 554
 host: dynamic
 type: friend
  context: from-internal
   permit: NULL
 deny: NULL
   secret: 201
md5secret: NULL
 remotesecret: NULL
transport: NULL
 dtmfmode: NULL
  directmedia: yes
  nat: NULL
allow: ulaw
 disallow: g729
 insecure: invite
 callerid: NULL
rfc2833compensate: NULL
  mailbox: NULL
   session-timers: NULL
  session-expires: NULL
session-minse: NULL
session-refresher: NULL


Kindly help me to resolve this.

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

[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 after the Dial command on hangup.

 Regards,
 Sammy.

 On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:

  Hello,

 I'm using AGI scripting with asterisk and need to execute certain
 commands just after Dial(). But once dial command is executed, further
 commands/instructions are ignored.


 $agi-exec(Dial,SIP/100);
 $dialstatus = $agi - get_variable(DIALSTATUS);

 if($dialstatus[data]==ANSWER)

 {
do something...
 }

 thanks,
 Kamlesh

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

[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 called party to
the specified  priority plus one.
NOTE: You cannot use any additional action post answer options in
conjunction with this option.


On Tue, Dec 27, 2011 at 6:15 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:

  hangup extension works once the call is terminated but I want to know the
 status of call immediately after connected, cancelled, or rejected and so
 on.

 thanks,
 Kamlesh

  --
 Date: Tue, 27 Dec 2011 16:59:35 +0530
 From: dhaval.it01...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] execute command just after Dial()


 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 after the Dial command on hangup.

 Regards,
 Sammy.

  On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar 
 kamlesh_...@hotmail.comwrote:

   Hello,

 I'm using AGI scripting with asterisk and need to execute certain commands
 just after Dial(). But once dial command is executed, further
 commands/instructions are ignored.


 $agi-exec(Dial,SIP/100);
 $dialstatus = $agi - get_variable(DIALSTATUS);

 if($dialstatus[data]==ANSWER)

 {
do something...
 }

 thanks,
 Kamlesh

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

[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  extension 222 but i want to call any number
 from my sip account 222 and the call hang up after 1 Min

 for exemple i call my mobile phone 067XXX using my sip 222 (x-lite) and
 the call hangup after 1 min

 any help please

 thanks and regards



 2011/9/28 Tarek Sawah tareksa...@hotmail.com

  one adjustment i would suggest is using (|) instead of (,)


 exten = 222,n,Dial(SIP/${EXTEN}||KkTtL(6))




 Tarek Sawah

 Information Technology  Adviser

 Integrated Digital Systems

 CCNP, MCSE, RHCE, TELECOM

 USA: +1 386 492 9993



  --
 Date: Wed, 28 Sep 2011 18:32:28 +

 From: salah.elharit...@gmail.com
 To: asterisk-users@lists.digium.com
   Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute

  sorry but the issue still the same there is no hangup after 1Min

 regards

 2011/9/28 Danny Nicholas da...@debsinc.com

  As I read this, the following should be correct:

 exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6))

 

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
 elharit
 *Sent:* Wednesday, September 28, 2011 1:23 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Limit outbond calls duration to 1 minute*
 ***

 ** **

 but there is no exemple for when i must put X in order to limit the call*
 ***

  

 can you please give me an exemple

  

 regards

 2011/9/28 Tarek Sawah tareksa...@hotmail.com

 have a look at the following:
 *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are
 left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are
 optional.


 source
 http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

 Tarek Sawah

 Information Technology  Adviser

 Integrated Digital Systems

 CCNP, MCSE, RHCE, TELECOM

 USA: +1 386 492 9993


 
  --

 Date: Wed, 28 Sep 2011 17:59:27 +
 From: salah.elharit...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Limit outbond calls duration to 1 minute 

 ** **

 hello list 

  
 i have configured a sip account in order to do an outbound calls and i
 want to force a hang up after 1 min for 222 sip

  

  

 in extensions.conf i have 

  

 exten = 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = 222,n,AbsoluteTimeout(60)

 exten = 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten = 222,n,Dial(SIP/${EXTEN},,KkTt)
 exten = 222,n,Hangup();
 could you please see this code and tell me waht is wrong
 thanks and regards

  

  

 ** **

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

[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 managers displayed.

 Best regards,
 James.zhu
 Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
 website: www.voipviews.com


 
 From: gohar.ah...@vopium.com
 To: asterisk-users@lists.digium.com
 Date: Thu, 25 Aug 2011 11:26:53 +0500
 Subject: Re: [asterisk-users] How to know how many user is connected

 I’m not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :

 See this page 
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will help
you.





 From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
 Sent: Wednesday, August 24, 2011 6:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] How to know how many user is connected



 Hi List,

 I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI

 ?php

   $socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
   if (!$socket)
   {
  $done=0;
   } else {
   fputs($socket, Action: Login\r\n);
   fputs($socket, UserName: root\r\n);
   fputs($socket, Secret: energy\r\n\r\n);
   fputs($socket, Action: Command\r\n);
   fputs($socket, Command: manager show connected\r\n);
   $done=1;
   }

 ?

 Now how to get information into this PHP file

 -
 Thanks and regards

  Virendra Bhati
 +91-9172341457
 Software Engineer



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

[DHAVAL INDRODIYA]

/etc/dahdi/system.conf

span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
echocanceller=mg2,1-15,17-31





/etc/asterisk/chan_dahdi.conf
[trunkgroups]

[channels]
language=en
context=from-pstn
switchtype=euroisdn
pridialplan=local
prilocaldialplan=local
callerid=asreceived
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=8000
resetinterval=never
rxgain=5.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
relaxdtmf=yes
faxdetect=both
cidstart=polarity_IN

group=0
channel = 1-15
channel = 17-31


2011/8/18 James zhu zhulizh...@live.com

  hi:
 please show the config files.

 Best regards,
 James.zhu
 Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
 gateway(fxs/fxo/pri-SIP).
 website: www.voipviews.com


 --
 Date: Wed, 17 Aug 2011 10:51:48 +0530
 From: dhaval.it01...@gmail.com
 To: asterisk-users@lists.digium.com; rmeyerrie...@digium.com
 Subject: Re: [asterisk-users] Any Method for capturing ISUP packets in
 DAHDI/ASTERISK


 Hi Russ,

 I have tried given patch and successfully compiled dahdi_pcap but when i
 try to run below command it gives me error.

 *./dahdi_pcap lapd 16 test.pcap *

 error setting channel err=-1!
 error setting channel err=-1!
 error setting channel err=-1!
 error setting channel err=-1!
 Segmentation fault

 I have TE112 Card 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 [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.
 
  This issue and patch added pcap support for a guy who wanted to monitor
  ss7 traffic. You'll need to use dahdi 2.5. You'll also need to compile
  the dahdi_pcap program on your own, or write a script to exercise the
  DAHDI_RXMIRROR and DAHDI_TXMIRROR ioctl's. This is an unsupported
  interface.

 Forgot to link to the feature request:
 https://issues.asterisk.org/view.php?id=16831

 --
 Russ Meyerriecks
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 direct: +1 256-428-6025
 Check us out at: www.digium.com  www.asterisk.org

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

[DHAVAL INDRODIYA]

Hi Russ,

I have tried given patch and successfully compiled dahdi_pcap but when i try
to run below command it gives me error.

*./dahdi_pcap lapd 16 test.pcap *

error setting channel err=-1!
error setting channel err=-1!
error setting channel err=-1!
error setting channel err=-1!
Segmentation 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,
  
   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.
 
  This issue and patch added pcap support for a guy who wanted to monitor
  ss7 traffic. You'll need to use dahdi 2.5. You'll also need to compile
  the dahdi_pcap program on your own, or write a script to exercise the
  DAHDI_RXMIRROR and DAHDI_TXMIRROR ioctl's. This is an unsupported
  interface.

 Forgot to link to the feature request:
 https://issues.asterisk.org/view.php?id=16831

 --
 Russ Meyerriecks
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 direct: +1 256-428-6025
 Check us out at: www.digium.com  www.asterisk.org

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

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

[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 wrote:

 **
 find the inline comment...


 On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:

 The dialplan is very simple. When the call comes in, we hand the call over
 to adhearsion.
 This is how the dialplan looks:

 ;group 0 will be used for incoming calls
 EXOIN = DAHDI/g0

 ;group 11 for outgoing
 EXOOUT = DAHDI/G11

 ;This will be used by adhearsion
 EXOCID=

 [general]
 autofallthrough = yes ;really?
 clearglobalvars = no

 [frompstn]
 ;Send everything to adhearsion
 exten = _X.,1,Ringing
 exten = _X.,n,AGI(agi://127.0.0.1)

 exten = _X.,n,Hangup() ; Please try this.


 ; End dialplan

 The rest of the logic happens in adhearsion.

 --
 Thanks,
 Ishwar.


 On Thu, Jul 28, 2011 at 6:33 PM, Nikhil d.nik...@cem-solutions.netwrote:

  Can you share the dialplan ,where SIP call is dialing...
 Thanks
 Nikhil


 On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:

  Hello everybody,

 We have an asterisk 1.8.4.1 setup, connected to a PRI line.

 We're currently facing an issue where asterisk does not recognise the
 event when the called party declines/cuts the call. This happens
 specifically over calls on a PRI line. For calls over SIP, call decline
 event is captured properly.

 I wasn't able to find a solution on the asterisk-users mailing list
 archive. Any suggestions/help would be much appreiciated :) I can share the
 relevant parts of the configuration files, if needed.

 Here's an excerpt from asterisk logs for a SIP call.
 -- SIP/x- requested special control 16, passing it to
 SIP/x-0001
 -- Started music on hold, class 'default', on SIP/x-0001
 -- SIP/x- requested special control 20, passing it to
 SIP/x-0001
 -- Got SIP response 603 Decline back from 127.0.0.1:5063
 -- SIP/x-0001 is busy
 -- Stopped music on hold on SIP/x-0001

 As you can see, on a SIP call, a call reject event is identified.

 For a call over the PRI, on the other hand, this event is not recognised.
 Here's an excerpt from asterisk log for a call over PRI.
 Call from  to .
 -- Requested transfer capability: 0x10 - 3K1AUDIO
 -- Called G11/x
 -- Started music on hold, class 'default', on DAHDI/i1/y
 -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
 -- DAHDI/i1/x-18f8 is ringing
 # At this point in time, x rejects the call. The event that's logged
 in asterisk is the following:
 -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y
 # And the call times out after the default 30s.
 -- Nobody picked up in 3 ms

 Is there a reason why asterisk doesn't recognise the call decline, and
 does it need any configuration changes to enable this?

 Thanks for your help.

 --
 Cheers,
 Ishwar.


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

[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
values:

 ASTERISK_PROMPT='%t, %l2, %h* '
export PATH USER LOGNAME MAIL HOSTNAME HISTSIZE INPUTRC ASTERISK_PROMPT

Your *export* variables will probably be different; just tack
ASTERISK_PROMPT on at the end. Reboot, run *asterisk -r* from your X
terminal, and voilá! The prompt is customized and your colors do not change:


*17:51:30, 0.54, asterisk1.alrac.net**





On Fri, Jul 29, 2011 at 4:26 PM, virendra bhati virbh...@gmail.com wrote:

 Hi List,

 I want to use these features but nothing was found after googling . please
 give me some examples

 Asterisk CLI prompt
 Changing the CLI Prompt

 The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable
 that
 you set from the Unix shell before starting the Asterisk CLI (not the
 server).

 You may include the following variables, that will be replaced by
 the current value by Asterisk:

 %d Date (year-month-date)
 %s Asterisk system name (from asterisk.conf)
 %h Full hostname
 %H Short hostname
 %t Time
 %% Percent sign
 %# '#' if Asterisk is run in console mode, '' if running as remote console

 %Cn[;n] Change terminal foreground (and optional background) color to
 specified
 *A full list of colors may be found in include/asterisk/term.h
 *

 On Linux systems, you may also use
 %l1 Load average over past minute
 %l2 Load average over past 5 minutes
 %l3 Load average over past 15 minutes
 %l4 Process fraction (processes running / total processes)
 %l5 The most recently allocated pid


 --



 -
 Thanks and regards

  Virendra Bhati
 +91-9172341457
 Software Engineer


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Re: [asterisk-users] Dialplan required for recording

[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 vdharash...@gmail.comwrote:

 Hi team,

 Can any one help me to implement dialplan in which conversation between
 a-party and b-party (call patch) needs to be recorded.

 Thanks
 Vinod Dharashive
 Sent from BlackBerry® on Airtel
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[asterisk-users] Meaning Callerid Datatypes.

[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
subaddr-valid
subaddr-type
subaddr-odd
tag
ANI-all
ANI-name
ANI-name-valid
ANI-name-charset
ANI-name-pres
ANI-num
ANI-num-valid
ANI-num-plan
ANI-num-pres
ANI-tag
RDNIS
DNID
dnid-num-plan
dnid-subaddr
dnid-subaddr-valid
dnid-subaddr-type
dnid-subaddr-odd
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Re: [asterisk-users] Connect Avaya to Asterisk PBX

[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  = _XX,1,Dial(H323/${EXTEN}@Avaya)


On Wed, Jul 13, 2011 at 1:35 PM, Malvin Rito
mr...@mail.altcladding.com.phwrote:

 **
 How do I write it on my code?


 On 7/13/2011 4:04 PM, Warren Selby wrote:

 Looks like you need an 's' exten in your [internal] context.

 Thanks,
 --Warren Selby, dCAP

 On Jul 13, 2011, at 2:02 AM, Malvin Rito mr...@mail.altcladding.com.ph
 wrote:

   Hi List,

 I have another issue on allowing outgoing calls to PSTN on Asterisk via
 Avaya Phones, I hope that anyone could help me fix this issue:

 *When I dial through Avaya phone i just here a good bye message reply
 from asterisk server. And here is the log:*

  == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back
 to exten 's'
   == Starting OOH323/(null)-b7db8aa0 at internal,s,1 still failed so
 falling back to context 'default'
 -- Executing [s@default:1] Playback(OOH323/(null)-b7db8aa0,
 vm-goodbye) in new stack
 -- OOH323/(null)-b7db8aa0 Playing 'vm-goodbye.ulaw' (language 'en')
 -- Executing [s@default:2] Macro(OOH323/(null)-b7db8aa0,
 hangupcall) in new stack
 -- Executing [s@macro-hangupcall:1] GotoIf(OOH323/(null)-b7db8aa0,
 1?skiprg) in new stack
 -- Goto (macro-hangupcall,s,4)
 -- Executing [s@macro-hangupcall:4] GotoIf(OOH323/(null)-b7db8aa0,
 1?skipblkvm) in new stack
 -- Goto (macro-hangupcall,s,7)
 -- Executing [s@macro-hangupcall:7] GotoIf(OOH323/(null)-b7db8aa0,
 1?theend) in new stack
 -- Goto (macro-hangupcall,s,9)
 -- Executing [s@macro-hangupcall:9] Hangup(OOH323/(null)-b7db8aa0,
 ) in new stack
   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
 'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
   == Spawn extension (default, s, 2) exited non-zero on
 'OOH323/(null)-b7db8aa0'
 -- Executing [h@default:1] Macro(OOH323/(null)-b7db8aa0,
 hangupcall,) in new stack
 -- Executing [s@macro-hangupcall:1] GotoIf(OOH323/(null)-b7db8aa0,
 1?skiprg) in new stack
 -- Goto (macro-hangupcall,s,4)
 -- Executing [s@macro-hangupcall:4] GotoIf(OOH323/(null)-b7db8aa0,
 1?skipblkvm) in new stack
 -- Goto (macro-hangupcall,s,7)
 -- Executing [s@macro-hangupcall:7] GotoIf(OOH323/(null)-b7db8aa0,
 1?theend) in new stack
 -- Goto (macro-hangupcall,s,9)
 -- Executing [s@macro-hangupcall:9] Hangup(OOH323/(null)-b7db8aa0,
 ) in new stack
   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
 'OOH323/(null)-b7db8aa0' in macro 'hangupcall'
   == Spawn extension (default, h, 1) exited non-zero on
 'OOH323/(null)-b7db8aa0'

 *Here is also the content of my extensions_custom.conf:*
 [general]
 static=yes
 autofallthrough=yes

 [internal]
 exten = 1000,1,Dial(SIP/1000)
 exten = 1000,2,HangUp()

 exten = _,1,Dial(H323/${EXTEN}@Avaya)
 exten = _XXX,1,Dial(H323/${EXTEN}@Avaya)
 exten  = _XX,1,Dial(H323/${EXTEN}@Avaya)

 *Here is also the content of my ooh323.conf:*
 [general]
 faststart=yes
 h245tunneling=yes
 gatekeeper=DISABLE
 bindaddr=10.1.129.231
 port=1720
 callerID=ALT Asterisk PBX
 progress_setup=8
 progress_alert=8
 disallow=all
 allow=all
 dtmfmode=inband
 faststart=yes
 context=internal

 [Avaya]
 type=friend
 context=internal
 host=10.1.129.247
 port=1720
 canreinvite=no
 disallow=all
 allow=alaw
 dtmfmode=inband

 *Here is also the content of sip_custom.conf:*
 [general]
 context=internal
 videosupport=yes
 allow=h261
 allow=h263
 allow=h263p
 bindaddr=10.1.129.231
 srvlookup=yes
 conreinvitte=no

 [1000]
 type=friend
 secret=malvin123
 host=dynamic
 dtmfmode=inband
 disallow=all
 allow=all
 nat=yes


 Thanks  regards,
 Malvin

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Re: [asterisk-users] DB Driven IVR

[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 with DB
 Driven IVR.



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

[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 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 http://chinaroby.com
  which is X1600P 16 port FXO  card. they also provide us wctdm.c file.
 
  card is detected successfully, incoming and outgoing calls scenario is
  also fine.
 
  we are unable to receive dtmf properly it means there is some digit are
  missing when we receive dtmf the ratio of sucess is about to 70% and 30%
  of calls are getting wrong dtmf .
 
  Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24
 
  I load module using
  modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1
  fixedtimepolarity=16
 
  here id  chan_dahdi.conf.

 Hello,

 did you try plaing with rxgain and txgain?
 When I set up a TDM400, I had some issues with DTMF because the signals
 where overmodulated.

 Regards,
 Ruben

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

[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 successfully, incoming and outgoing calls scenario is also
fine.

we are unable to receive dtmf properly it means there is some digit are
missing when we receive dtmf the ratio of sucess is about to 70% and 30% of
calls are getting wrong dtmf .

Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24

I load module using
modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1
fixedtimepolarity=16

here id  chan_dahdi.conf.

[trunkgroups]

[channels]
context=from-zaptel
signalling=fxs_ks
busydetect=yes
busycount=4
;rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
callerid=asreceived
cidstart=polarity_in
cidsignalling=dtmf
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callprogess=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
;cid_rxgain=5.0
relaxdtmf=yes
callgroup=1
pickupgroup=1
toneduration=500
;answeronpolarityswitch=yes
hanguponpolarityswitch=yes
;polarityonanswerdelay=1000

group=0
channel = 1
;channel = 2
;channel = 3
;channel = 4
;channel = 5
;channel = 6
;channel = 7
;channel = 8
;channel = 9
;channel = 10
;channel = 11
;channel = 12
;channel = 13
;channel = 14
;channel = 15
;channel = 16


Also set tonezone = in in system.conf, tried many solutions and changed so
many parameters of chan_dahdi.cong but still i am not getting successful
result.


Please share your comments if anyone have idea for india specific region .

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

[DHAVAL INDRODIYA]

Yes dear i have tried diable also with yes and no. but no successful result
found/

On Fri, Jul 8, 2011 at 12:41 PM, Faisal Hanif fai...@vopium.com wrote:

 Did u tried by disabling relaxdtmf?

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@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 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 successfully, incoming and outgoing calls scenario is also
 fine.

 we are unable to receive dtmf properly it means there is some digit are
 missing when we receive dtmf the ratio of sucess is about to 70% and 30% of
 calls are getting wrong dtmf .

 Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24

 I load module using
 modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1
 fixedtimepolarity=16

 here id  chan_dahdi.conf.

 [trunkgroups]

 [channels]
 context=from-zaptel
 signalling=fxs_ks
 busydetect=yes
 busycount=4
 ;rxwink=300  ; Atlas seems to use long (250ms) winks
 usecallerid=yes
 callerid=asreceived
 cidstart=polarity_in
 cidsignalling=dtmf
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 callprogess=yes
 echocancel=yes
 echocancelwhenbridged=no
 echotraining=800
 rxgain=0.0
 txgain=0.0
 ;cid_rxgain=5.0
 relaxdtmf=yes
 callgroup=1
 pickupgroup=1
 toneduration=500
 ;answeronpolarityswitch=yes
 hanguponpolarityswitch=yes
 ;polarityonanswerdelay=1000

 group=0
 channel = 1
 ;channel = 2
 ;channel = 3
 ;channel = 4
 ;channel = 5
 ;channel = 6
 ;channel = 7
 ;channel = 8
 ;channel = 9
 ;channel = 10
 ;channel = 11
 ;channel = 12
 ;channel = 13
 ;channel = 14
 ;channel = 15
 ;channel = 16


 Also set tonezone = in in system.conf, tried many solutions and changed so
 many parameters of chan_dahdi.cong but still i am not getting successful
 result.


 Please share your comments if anyone have idea for india specific region .

 Regards
 Dhaval

 

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

[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 
positivelyoptimis...@gmail.com wrote:

 Greetings!!

 We're getting some strange results using call files..  no matter the
 technology, DAHDI, SIP, etc., we get a Call failed to go through, reason
 (3) Remote end Ringing message when attempting to originate a call from a
 call file.  Numbers changed to protect the innocent



 using call file
 //CALL FILE//

 Channel: DAHDI/g1/918005551212
 Callerid: 8002211212
 WaitTime: 2
 MaxRetries: 6
 RetryTime: 8

 Context: xs-globx-ds3
 Extension: 12564286000
 Priority: 1

 //CALL FILE//

 //CLI SNIPPET//

 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 1)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:14] NOTICE[27176]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 2)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:24] NOTICE[27177]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 3)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:34] NOTICE[27179]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 4)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:44] NOTICE[27182]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 5)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:35:54] NOTICE[27183]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 6)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:36:04] NOTICE[27185]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing
 -- Attempting call on DAHDI/g1/918005551212 for 12564286000@
 xs-globx-ds3:1 (Retry 7)
 -- Requested transfer capability: 0x00 - SPEECH
 -- PROGRESS with cause code 31 received
 -- Hungup 'DAHDI/1-1'
 [2011-06-15 01:36:14] NOTICE[27188]: pbx_spool.c:339 attempt_thread: Call
 failed to go through, reason (3) Remote end Ringing

 //CLI SNIPPET//

 Software Version(s)

 Asterisk 1.6.2.16.1
 DAHDI Version: 2.4.0
 libpri version: 1.4.11.5




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

[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 on to
 the second one (I may be simplifying a bit)



 Mike







 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Deka, Rajib IN MAA
 SL
 *Sent:* Monday, June 13, 2011 6:44 AM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] asterisk queue 'ringall' stratagy



 Hi List,



 I have faced a problem in asterisk queue implementation.



 I configured a queue with ‘ringall’ strategy and ‘ringinuse=yes’ in
 queues.conf. If three calls come to this queue in parallel, the logged in
 queue agent used to get only one call (may be the first one), not all the
 calls waiting in the queue at a time. Once the agent answers the call the
 next call is displayed.

 I want to display all the waiting calls on the agent’s desktop. Is it
 possible to do, if yes how? Please help me with this.



 Regards,

 Rajib



 *Rajib Deka*

 SIEMENS Ltd.

 Robert V Chandran Tower, First Floor, West Wing,

 #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.

 www.siemens.com



 Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com




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 please notify us immediately by reply e-mail and delete this e-mail and its
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Re: [asterisk-users] no audio with SIP:INFO in meetme

[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 meetme with DTMF mode
 enabled as INFO for SIP channel.

 If it is a bug in asterisk or something need to be enabled in sip.conf for
 the same.



 Dialplan looks like

 Exten = 100,1,MeetMe(100,dmF)



 Sip.conf

 dtmfmode=info



 Regards,

 Rajib





 *Rajib Deka*

 SIEMENS Ltd.

 Robert V Chandran Tower, First Floor, West Wing,

 #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.

 www.siemens.com



 Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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

[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 consider more on load balancing per asterisk server in
that case the problem of Availability is solved to some level, If You using
SIP protocol then you can think of OPENSER and from that you can use
loadbalancer which routed calls in a way an depend on machine strength.

I hope this idea will useful to solve your requirement.

Regards
Dhaval

On Wed, May 4, 2011 at 1:13 PM, Deka, Rajib IN MAA SL 
rajib.d...@siemens.com wrote:

  Hello List,



 We are running two asterisk machines in virtual IP as primary and secondary
 server.

 Initially virtual IP will be active in primary server; during the failure
 of primary secondary will get the virtual IP.



 Is there any way to retrieve pending queue calls from primary to secondary,
 in case primary fails?

 Does asterisk provide any interface to do it or we have to write some
 application on asterisk to do the same.



 Regards,

 Rajib



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Re: [asterisk-users] Meetme Time Limit?

[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 Meetme conf and I have had a few instances where someone
 from a wifi voip phone looses ip while doing a page and the page never hangs
 up. I have to kill it. I need to somehow limit the page so after a worse
 case 2Min timeout it hangs up.

 Thanks
 Bryant

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

[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 little complicated as it is H/W specific.
  Basically we are integrating a radio gateway (SIP) with asterisk. The
 gateway will be
  connected to a meetme room, so that any operator (with IP phone
 registered as SIP user to
  asterisk) can login to the room and listen to radio communications and
 talk.
 
  Using a PTT button someone can talk on a radio channel. Once someone
 presses the PTT button
  a SIP MESSAGE is sent to the gateway with a string as payload to enable
 half duplex
  communication. So, we were planning to run an AGI script with meetme
 (AGI_BACKGROUND) to
  receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends and
 to generate a
  VarSet AMI event.
 
  Operator (wants to talk) - SIP:MESSAGE -MeetMe(asterisk)- SIP:MESSAGE
 - radio gateway
  And vise versa.
 
  Any suggestions on the above scenario.

 I don't think it can be done without making modifications to Asterisk.

 The first thing I would do, if you haven't done so already, would be to
 try it without MeetMe:

 Operator (wants to talk) - SIP:MESSAGE -Dial(asterisk) SIP:MESSAGE -
 radio gateway

 If that works, then it would suggest that the SIP MESSAGE is
 successfully getting translated into an ast_frame, which is then getting
 translated back into a SIP MESSAGE. If that is not happening, you might
 need to add some code to chan_sip.c to do those steps.

 Once Asterisk is converting the message to and from an ast_frame, the
 next step would be to add some code to app_meetme.c in the conf_run()
 function, to pass those frames through, in the same way as DTMF frames
 get passed through when the F option is enabled.

 Presumably the messages represent PTT PRESS and PTT RELEASE. You will
 need to decide what to do if you have two operators connected and they
 both press the PTT.

 You might also need to automatically unmute or mute the operator
 channel when their PTT is pressed or released. That could also be done
 within the MeetMe code.

 There may be other approaches too...

 Hope this helps!
 Tony
 --
 Tony Mountifield
 Work: t...@softins.co.uk - http://www.softins.co.uk
 Play: t...@mountifield.org - http://tony.mountifield.org

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

[DHAVAL INDRODIYA]

is your problem solved or not

On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL 
rajib.d...@siemens.com wrote:

 Thanks a lot Tony and Dhaval for your much appreciable suggestions.

 Regards,
 Rajib

 Rajib Deka
 SIEMENS Ltd.
 Robert V Chandran Tower, First Floor, West Wing,
 #149, Velechery 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 with
AGI_BACKGROUND
 To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
 Message-ID: BANLkTikgRHjCVJhBC097S8n9YM66VWp=q...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 hey try with app_rpt in asterisk

 regards
 dhaval

 On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield t...@softins.co.uk
 wrote:

  In article 
 
 2658e54b540d284981ea57e6a549ea70abd1fdf...@inblrk77m1msx.in002.siemens.net
  ,
  Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote:
  
   The requirement is little complicated as it is H/W specific.
   Basically we are integrating a radio gateway (SIP) with asterisk. The
  gateway will be
   connected to a meetme room, so that any operator (with IP phone
  registered as SIP user to
   asterisk) can login to the room and listen to radio communications and
  talk.
  
   Using a PTT button someone can talk on a radio channel. Once someone
  presses the PTT button
   a SIP MESSAGE is sent to the gateway with a string as payload to enable
  half duplex
   communication. So, we were planning to run an AGI script with meetme
  (AGI_BACKGROUND) to
   receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends
 and
  to generate a
   VarSet AMI event.
  
   Operator (wants to talk) - SIP:MESSAGE -MeetMe(asterisk)-
 SIP:MESSAGE
  - radio gateway
   And vise versa.
  
   Any suggestions on the above scenario.
 
  I don't think it can be done without making modifications to Asterisk.
 
  The first thing I would do, if you haven't done so already, would be to
  try it without MeetMe:
 
  Operator (wants to talk) - SIP:MESSAGE -Dial(asterisk) SIP:MESSAGE -
  radio gateway
 
  If that works, then it would suggest that the SIP MESSAGE is
  successfully getting translated into an ast_frame, which is then getting
  translated back into a SIP MESSAGE. If that is not happening, you might
  need to add some code to chan_sip.c to do those steps.
 
  Once Asterisk is converting the message to and from an ast_frame, the
  next step would be to add some code to app_meetme.c in the conf_run()
  function, to pass those frames through, in the same way as DTMF frames
  get passed through when the F option is enabled.
 
  Presumably the messages represent PTT PRESS and PTT RELEASE. You will
  need to decide what to do if you have two operators connected and they
  both press the PTT.
 
  You might also need to automatically unmute or mute the operator
  channel when their PTT is pressed or released. That could also be done
  within the MeetMe code.
 
  There may be other approaches too...
 
  Hope this helps!
  Tony
  --
  Tony Mountifield
  Work: t...@softins.co.uk - http://www.softins.co.uk
  Play: t...@mountifield.org - http://tony.mountifield.org
 
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Re: [asterisk-users] ConfBridge and AGI

[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 that basis people of group can
help you.

regards
Dhaval

On Tue, Apr 19, 2011 at 2:13 PM, Deka, Rajib IN MAA SL 
rajib.d...@siemens.com wrote:

  Hello List,



 Is it possible to run an AGI script in backgroung for all the associated
 SIP channels in ConfBridge Application? If yes how?

 This can be done using ‘b’ parameter in MeetMe for non SIP channels.



 Regards,

 Rajib



 *Rajib Deka*

 SIEMENS Ltd.

 Robert V Chandran Tower, First Floor, West Wing,

 #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.

 www.siemens.com



 Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] MeetMe headache

[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 Anderson 
rander...@customteleconnect.com wrote:

 Ok, I've been running applications on 1.4 for quite some time using
 meetme to hold a person, while the person on the other end of the call
 accepts, etc. I was playing status messages to the calling party using a
 context like this:

 [status-one-en]
 exten = 100,1,Playback(my_status_message)
 exten = 100,1,Hangup()

 and then creating a call file like this:

 Channel: Local/100@status-one-en
 CallerID: Rick 55
 MaxRetries: 0
 RetryTime: 15
 WaitTime: 45
 Application: MeetMe
 Data: 12345,qdM

 and it would hook into the meetme, play the message, then hangup and
 drop out.

 I've been building an application with 1.6, and this isn't working at
 all. In verbose mode, I see the message played, and the call hang up,
 but the music never even stops on the meetme. After about 20 seconds I
 get:

 Call failed to go through, reason (3) Remote end Ringing

 Is there some other way to do this in 1.6 that I'm unaware of? I've
 tried creating a context and extension for the meetme portion (rather
 than using the Application/Data in the call file, and switched the order
 around (which does cause the music to stop, but the announcement still
 doesn't get played, and I get the same call failed message). I've been
 googling on this for days now, and really just need to get it working.

 TIA

 Rick


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Re: [asterisk-users] agi voicemail callback

[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 Göllner t...@ovm-group.com wrote:

 Am 05.04.2011 18:50, schrieb vip killa:

  I'm wondering if there is a simply way to perform a voicemail callback
 feature using AGI.
 For instance, a caller leaves a voicemail, the voicemail will then call
 the owner of the voicemailbox determined by a database look up.


 One possibility: look via cron job, if there is a new message and if so,
 you can drop a call file in /var/spool/asterisk/outgoing.


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Re: [asterisk-users] Read VoiceMail direct

[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 23:16:30 +0200
 Subject: Re: [asterisk-users] Read VoiceMail direct


 Hi,

 maybe:

 exten = 8500,3,VoiceMailMain(${CALLERID(num)}@default)

 Regards

 - Andrea

 - Original Message -

 *From:* satish patel satish...@hotmail.com
 *To:* asterisk-users asterisk-users@lists.digium.com
 *Sent:* Monday, April 04, 2011 11:08 PM
 *Subject:* [asterisk-users] Read VoiceMail direct

 Hey Guy!

 I want direct access of VoiceMail without asking mailbox number (Direct ask
 PIN). I am using following dialplan but its still asking me Mailbox number.
 Look like asterisk 1.8 doesn't support CALLERIDNUM variable.

 Do you have any idea ?


 exten = 8500,1,answer
 exten = 8500,2,wait(1)
 exten = 8500,3,voicemailmain(${CALLERIDNUM:-4}@default)
 exten = 8500,4,hangup
 exten = i,1,playback(invalid)
 exten = i,2,hangup


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

[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 in asterisk calling card when
 a user has dialed the access number and during the IVR or any time during
 the call, he can press ## or ** to end the current call and dial a new
 destination number?

 Please help and give me a step by step help. Thanks.

 Rgrds
 -
 Abid

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

[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 abid_aster...@hotmail.comwrote:

  what will be the actual value
 inside {DATA_OF_DIAL_NUMBER_WITH_TECHNOLOGY}. Can you please give an
 example.

 --
 Date: Fri, 1 Apr 2011 11:48:11 +0530
 From: dhaval.it01...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] ** to disconnect and make a new call


 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 in asterisk calling card when
 a user has dialed the access number and during the IVR or any time during
 the call, he can press ## or ** to end the current call and dial a new
 destination number?

 Please help and give me a step by step help. Thanks.

 Rgrds
 -
 Abid

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

[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
disconnected .


regards
dhaval

On Tue, Mar 29, 2011 at 3:27 PM, Gilles codecompl...@free.fr wrote:

 On Tue, 29 Mar 2011 07:48:08 +0200, magnu...@inputinterior.se wrote:
 I was a little unclear, it is not the cell phone that does the call-back,
 it
 is the cell-phone-network.

 Makes more sense :-) Thank you.


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

[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 Burton West ro...@firedrake.orgwrote:

 On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote:

 Is there a better way of handling the post-hangup
 processing?

 Callfiles?

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

[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 = _003318364,n,Set(CLI=${CUT(cut1,,1)})
exten = _003318364,n,Set(toexten=${CUT(CLI,@,1)})
exten = _003318364,n,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _003318364,n,ExecIf($[${toexten} =
81169]?Dial(SIP/204,180,rt):Noop(${toexten}))
exten = _003318364,n,ExecIf($[${EXTEN} =
003318364]?Dial(SIP/203,180,rt):Noop(${toexten}))


On Thu, Mar 24, 2011 at 11:13 AM, Olivier CALVANO o.calv...@gmail.comwrote:

 Hi

 Anyone know a solution at my problems ?

 Thanks
 Olivier







 2011/3/23 Olivier CALVANO o.calv...@gmail.com:
  Hi
 
  I request your help because i don't have actually a solution at my
 problems.
 
 
  I have a Asterisk Server in 1.6
  Connected at a SIP Provider
  This provider supply me 2 numbers:
  003318364 (official number)
  081169 (Nddi Number)
 
  When i receive a call on the 081169, he don't use
  the extension. He use the 003318364 extension.
 
  SIP Debug:
 
  --- SIP read from UDP://91.121.xxx.xxx:5060 ---
  INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
  Allow: UPDATE,REFER,INFO
  Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
  Contact: sip:91.121.xxx.xxx:5060
  Content-Type: application/sdp
  CSeq: 1602837515 INVITE
  From: 033426aa
  sip:033426aaa...@sip.myoperator.net
 ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
  Max-Forwards: 30
  P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone
  To: sip:081169x...@91.121.xxx.xxx;user=phone
  User-Agent: Cirpack/v4.42s (gw_sip)
  Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
  Content-Length: 481
 
  v=0
  o=cp10 130085910854 130085910854 IN IP4 10.7.1.121
  s=SIP Call
  c=IN IP4 91.121.bbb.bbb
  t=0 0
  m=audio 36146 RTP/AVP 18 4 0 8 125 111 101
  b=AS:21
  a=rtpmap:18 G729/8000/1
  a=fmtp:18 annexb=no
  a=rtpmap:4 G723/8000/1
  a=fmtp:4 annexa=no
  a=rtpmap:0 PCMU/8000/1
  a=rtpmap:8 PCMA/8000/1
  a=rtpmap:125 CLEARMODE/8000/1
  a=rtpmap:111 iLBC/8000/1
  a=fmtp:111 mode=30
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15
  a=ptime:30
  a=sendrecv
  a=sqn:0
  a=cdsc: 1 image udptl t38
 
  -
  --- (13 headers 22 lines) ---
  Sending to 91.121.xxx.xxx : 5060 (no NAT)
  Using INVITE request as basis request -
  04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
  Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060
  Found RTP audio format 18
  Found RTP audio format 4
  Found RTP audio format 0
  Found RTP audio format 8
  Found RTP audio format 125
  Found RTP audio format 111
  Found RTP audio format 101
  Peer audio RTP is at port 91.121.bbb.bbb:36146
  Found audio description format G729 for ID 18
  Found audio description format G723 for ID 4
  Found audio description format PCMU for ID 0
  Found audio description format PCMA for ID 8
  Found unknown media description format CLEARMODE for ID 125
  Found audio description format iLBC for ID 111
  Found audio description format telephone-event for ID 101
  Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d
  (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing),
  combined - 0x109 (g723|alaw|g729)
  Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
  (telephone-event), combined - 0x1 (telephone-event)
  Peer audio RTP is at port 91.121.bbb.bbb:36146
  Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx)
 
  --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 ---
  SIP/2.0 404 Not Found
  Via: SIP/2.0/UDP
  91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx
  From: 033426aa
  sip:033426aaa...@sip.myoperator.net
 ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
  To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a
  Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
  CSeq: 1602837515 INVITE
  Server: Asterisk PBX 1.6.1.8
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces, timer
  Content-Length: 0
 
 
  
  [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527
  handle_request_invite: Call from '0033459aa' to extension
  '003318364' rejected because extension not found.
  Scheduling destruction of SIP dialog
  '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method:
  INVITE)
  --- SIP read from UDP://91.121.xxx.xxx:5060 ---
  ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
  Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
  Contact: sip:91.121.xxx.xxx:5060
  CSeq: 1602837515 ACK
  From: 033426aa
  sip:033426aaa...@sip.myoperator.net
 ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
  Max-Forwards: 30
  To: 

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

[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 o.calv...@gmail.comwrote:

 Hi

 I request your help because i don't have actually a solution at my
 problems.


 I have a Asterisk Server in 1.6
 Connected at a SIP Provider
 This provider supply me 2 numbers:
 003318364 (official number)
 081169 (Nddi Number)

 When i receive a call on the 081169, he don't use
 the extension. He use the 003318364 extension.

 SIP Debug:

 --- SIP read from UDP://91.121.xxx.xxx:5060 ---
 INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
 Allow: UPDATE,REFER,INFO
 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
 Contact: sip:91.121.xxx.xxx:5060
 Content-Type: application/sdp
 CSeq: 1602837515 INVITE
 From: 033426aa
 sip:033426aaa...@sip.myoperator.net
 ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
 Max-Forwards: 30
 P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone
 To: sip:081169x...@91.121.xxx.xxx;user=phone
 User-Agent: Cirpack/v4.42s (gw_sip)
 Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
 Content-Length: 481

 v=0
 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121
 s=SIP Call
 c=IN IP4 91.121.bbb.bbb
 t=0 0
 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101
 b=AS:21
 a=rtpmap:18 G729/8000/1
 a=fmtp:18 annexb=no
 a=rtpmap:4 G723/8000/1
 a=fmtp:4 annexa=no
 a=rtpmap:0 PCMU/8000/1
 a=rtpmap:8 PCMA/8000/1
 a=rtpmap:125 CLEARMODE/8000/1
 a=rtpmap:111 iLBC/8000/1
 a=fmtp:111 mode=30
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
 a=ptime:30
 a=sendrecv
 a=sqn:0
 a=cdsc: 1 image udptl t38

 -
 --- (13 headers 22 lines) ---
 Sending to 91.121.xxx.xxx : 5060 (no NAT)
 Using INVITE request as basis request -
 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
 Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060
 Found RTP audio format 18
 Found RTP audio format 4
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 125
 Found RTP audio format 111
 Found RTP audio format 101
 Peer audio RTP is at port 91.121.bbb.bbb:36146
 Found audio description format G729 for ID 18
 Found audio description format G723 for ID 4
 Found audio description format PCMU for ID 0
 Found audio description format PCMA for ID 8
 Found unknown media description format CLEARMODE for ID 125
 Found audio description format iLBC for ID 111
 Found audio description format telephone-event for ID 101
 Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d
 (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing),
 combined - 0x109 (g723|alaw|g729)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
 (telephone-event), combined - 0x1 (telephone-event)
 Peer audio RTP is at port 91.121.bbb.bbb:36146
 Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx)

 --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 ---
 SIP/2.0 404 Not Found
 Via: SIP/2.0/UDP
 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx
 From: 033426aa
 sip:033426aaa...@sip.myoperator.net
 ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a
 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
 CSeq: 1602837515 INVITE
 Server: Asterisk PBX 1.6.1.8
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Length: 0


 
 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527
 handle_request_invite: Call from '0033459aa' to extension
 '003318364' rejected because extension not found.
 Scheduling destruction of SIP dialog
 '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method:
 INVITE)
 --- SIP read from UDP://91.121.xxx.xxx:5060 ---
 ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
 Contact: sip:91.121.xxx.xxx:5060
 CSeq: 1602837515 ACK
 From: 033426aa
 sip:033426aaa...@sip.myoperator.net
 ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
 Max-Forwards: 30
 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a
 User-Agent: Cirpack/v4.42s (gw_sip)
 Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
 Content-Length: 0







 I see in the debug:
 To: sip:081169x...@91.121.xxx.xxx;user=phone

 but he search the 003318364 extension
 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527
 handle_request_invite: Call from '0033459aa' to extension
 '003318364' rejected because extension not found.




 Anyone know the solution for he use the extension based on the To: ?

 thanks
 Olivier

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

[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
http://www.smartvox.co.uk/products_gateways_explained.htm

Asterisk as a Session Border Controller*
Equip the Asterisk server with two ethernet ports, connect one to the
Internet and the other to your internal network; set up the firewall,
configure the dial plans and you've got everything you need for a fully
functional Session Border Controller.

   - IP phones can register with the SBC either from the internal network or
   from the Internet.
   - Use your SBC as an Inbound and/or Outbound proxy to have complete
   control over incoming and outbound calls
   - Use it to control access to your IPBX and to overcome the usual
   problems associated with interfacing VoIP between your private network and
   the Internet
   - Solve one-way audio and other notoriously difficult and annoying NAT
   traversal problems while, at the same time, improving your systems security

regards
dhaval
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[asterisk-users] DIAL through Specific number in PRI

[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 caller want to dial any
number but caller should shown is 30550008 like this.

is there any solution from asterisk side.

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

[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.  Alternatives to TB are FreePBX  PBX in
 a Flash.  All are Asterisk based and very easy to set up.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins
 *Sent:* Thursday, February 17, 2011 7:29 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...



 If you already have experience with linux asterisk will be easy for you.



 Other people will reply with official links but here is how I use Asterisk
 in my small home office www.cognation.net/asterisk





 Cheers,

 Dean




 --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier
 Cintrón Olguín
 *Sent:* Thursday, February 17, 2011 7:26 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Newbie´s question about Asterisk...



 Hi, My name is Francisco from México.

 Here, in my work we have a very very old panasonic PBX(12 years old). We
 are growing and we need to increase our external lines(from 3 to 4) and our
 internal lines(from 6 to 10). Besides we need voice mail and voice menu too.


 We asked for a quote to our panasonic dealer. The whole thing cost about
 4,500 dollars.

 My boss just saw a thing called Asterisk this morning looking for options
 in Google. He asked my to investigate what this thing called Asterisk is and
 if we could save some money using it instead of the panasonic solution. So,
 here I am.

 I have some experience as linux sysadmin(we have 1 oracle linux server and
 1 linux print server) nevertheless I don´t have any idea where and how to
 start this evaluation?


 Please
 Would you give us a clue where to see If Asterisk could work for us?

 Thanks for your kind help.

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

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Meet me recording

[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 monitor application resolved my issue.

 --
 From: da...@debsinc.com
 To: asterisk-users@lists.digium.com
 Date: Fri, 18 Feb 2011 09:16:36 -0600
 Subject: Re: [asterisk-users] Meet me recording


   --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *satish patel
 *Sent:* Friday, February 18, 2011 9:12 AM
 *To:* asterisk-users
 *Subject:* [asterisk-users] Meet me recording



 Hey Users,

 I am using record application to record MeetMe conf. but look like its
 creating individual files for every channel. What applucation is best to
 record MeetMe conf ?


 ~ # ls -l /var/spool/asterisk/monitor/
 total 489220
 -rw-r--r-- 1 asterisk asterisk   44 Feb 16 08:42
 8881-conf-20110216-084224.wav
 -rw-r--r-- 1 asterisk asterisk  1858284 Feb 16 13:05
 8881-conf-20110216-130321.wav
 -rw-r--r-- 1 asterisk asterisk  1604204 Feb 16 13:05
 8881-conf-20110216-130337.wav
 -rw-r--r-- 1 asterisk asterisk   241964 Feb 17 08:20
 8881-conf-20110217-081957.wav
 -rw-r--r-- 1 asterisk asterisk 78678124 Feb 17 11:12
 8881-conf-20110217-095056.wav
 -rw-r--r-- 1 asterisk asterisk   612204 Feb 17 09:53
 8881-conf-20110217-095310.wav
 -rw-r--r-- 1 asterisk asterisk 81183084 Feb 17 11:13
 8881-conf-20110217-095414.wav
 -rw-r--r-- 1 asterisk asterisk 69488044 Feb 17 11:12
 8881-conf-20110217-100012.wav
 -rw-r--r-- 1 asterisk asterisk 68917164 Feb 17 11:12
 8881-conf-20110217-100052.wav
 -rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11
 8881-conf-20110217-100117.wav
 -rw-r--r-- 1 asterisk asterisk 66648684 Feb 17 11:12
 8881-conf-20110217-100327.wav
 -rw-r--r-- 1 asterisk asterisk 45056044 Feb 17 11:06
 8881-conf-20110217-102007.wav


 Thanks,
 S



 From what I read, mixmonitor.

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[asterisk-users] voice quality measurement using dahdi_monitor

[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 making .wav file and visulal
mode of RX and TX of PRI line.

what i want is measurement of voice quality so that i can talk with provider
that i am getting % of voice quality.i am sure there is
some better way to solve or debug .raw file and taking a decision.


let me help please to solve and finding problem of voice quality.


regards
Dhaval
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Re: [asterisk-users] PRI voice optimization

[DHAVAL INDRODIYA]

Hi Gopal,

i am using *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V* card
with tata PRI lines.

regards
dhaval

On Fri, Feb 4, 2011 at 3:23 PM, Gopalakrishnan A.N sai...@gmail.com wrote:

 It seems to be you are using Sangoma T1/E1 card with echo cancellation. If
 I am not wrong 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 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 having issue for callquality and other voice related
 issues. now my question is that is there
 any voice related parameter that we need to set for INDIA specific region
 and is ther any voice hardware tester for PRI
 that we can use and tell us our PRI [telco] provider that problem is not
 from our side. let give some idea . below are my configuration as well.



 # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
 # Zaptel Configuration File
 #
 # This file is parsed by the Zaptel Configurator, ztcfg
 #

 # It must be in the module loading order


 # Global data

 loadzone= in
 defaultzone = in


 span = 1,0,0,ccs,hdb3
 bchan = 1-15
 dchan = 16
 bchan = 17-31

 span = 2,0,0,ccs,hdb3
 bchan = 32-46
 dchan = 47
 bchan = 48-62

 span = 3,0,0,ccs,hdb3
 bchan = 63-77
 dchan = 78
 bchan = 79-93

 span = 4,0,0,ccs,hdb3
 bchan = 94-108
 dchan = 109
 bchan = 110-124



 [channels]
language=en
context=from-pstn
switchtype=euroisdn
pridialplan=local
prilocaldialplan=local
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
relaxdtmf=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
resetinterval=never
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
group = 0
channel = 1-15
channel = 17-31
channel = 32-46
channel = 48-62
channel = 63-77
channel = 79-93
channel = 94-108
channel = 110-124



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 Thank you  with regards,
 Gopalakrishnan A.N.
 VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.com



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[asterisk-users] PRI voice optimization

[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 having issue for callquality and other voice related
issues. now my question is that is there
any voice related parameter that we need to set for INDIA specific region
and is ther any voice hardware tester for PRI
that we can use and tell us our PRI [telco] provider that problem is not
from our side. let give some idea . below are my configuration as well.



# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Global data

loadzone= in
defaultzone = in


span = 1,0,0,ccs,hdb3
bchan = 1-15
dchan = 16
bchan = 17-31

span = 2,0,0,ccs,hdb3
bchan = 32-46
dchan = 47
bchan = 48-62

span = 3,0,0,ccs,hdb3
bchan = 63-77
dchan = 78
bchan = 79-93

span = 4,0,0,ccs,hdb3
bchan = 94-108
dchan = 109
bchan = 110-124



[channels]
   language=en
   context=from-pstn
   switchtype=euroisdn
   pridialplan=local
   prilocaldialplan=local
   signalling=pri_cpe
   usecallerid=yes
   hidecallerid=no
   callwaiting=yes
   usecallingpres=yes
   callwaitingcallerid=yes
   threewaycalling=yes
   transfer=yes
   cancallforward=yes
   callreturn=yes
   relaxdtmf=yes
   echocancel=yes
   echocancelwhenbridged=yes
   echotraining=yes
   resetinterval=never
   rxgain=0.0
   txgain=0.0
   callgroup=1
   pickupgroup=1
   immediate=no
   group = 0
   channel = 1-15
   channel = 17-31
   channel = 32-46
   channel = 48-62
   channel = 63-77
   channel = 79-93
   channel = 94-108
   channel = 110-124
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Re: [asterisk-users] Can a duration limit be specified in spool call file?

[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 wrote:
  Hi Everyone,
 
  I don't see any parameter for limiting duration of a call in the .call
  file for Asterisk spool outgoing directory.
 
  I'd rather not use a MeetMe to drop the call in a conference room and to
  then limit the call duration as that complicates things unnecessarily.
 
  I am wondering if there is anything else I can do or if the Channel
  parameter take call duration like the DIAL parameter?

 No, but you can specify a Local channel as the channel in the call file and
 then set a TIMEOUT(absolute) for the call, before you Dial() the actual
 channel you want to use.  Keep in mind that the actual channel could be
 specified by a Set variable in the callfile.

 --
 Tilghman

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

[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 using mediaproxy
[also tried rtpproxy] .

2) Asterisk server having public_ip as well local ip.

setup:

*UAC - KAMAILIO - ASTERISK*

UAC  registered to kamailio registration is successful. once it dial PSTN
number  i forwarded a call to asterisk server and then is created problem
because i am not getting any media from asterisk server.

so basically UAC sends a registered request to kamailio public ip and
kamailio and asterisk works on private ip , it sends data to asterisk
private ip, i am getting sip signaling and it looks okay. i can provide it
too if we required.

here is my asterisk sip.conf kamailio context looks like

[vmserver]
type=friend
context=default
host=***local_ip_of_kamailio***
; for below three i have tried all available options
*directmedia=nonat
directrtpsetup=yes
nat=yes
* t1min=500
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
qualify=yes


let me know how to solve this nating issue also i opened all required ports
for sip. and rtp


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

[DHAVAL INDRODIYA]

Hi Phuong,

i see your code is looking nice and there is no problem in implementation ,
if you have any problem
then first send me manager.conf file then try to connect through manager
using telnet and then fire same action on this in that you can get proper
error codes .

one more thing the channel you set is this channel is available to
redirected???

regards
Dhavak

On Tue, Jan 11, 2011 at 2:05 PM, Phuong Hoang
ducphuongbk200...@gmail.comwrote:

 Hi Dhaval,
 Can you say how to fire action on AMI in this case and recieve response on
 AMI. I also tried to do with HangupAction and RedirectAction action (using
 asterisk-java library) in application java (AMI) to hang up or redirect a
 channel that is online at the extension on asterisk but not successfully.
 This is my code:



 package Test;

 import java.io.IOException;

 import org.asteriskjava.manager.AuthenticationFailedException;
 import org.asteriskjava.manager.ManagerConnection;
 import org.asteriskjava.manager.ManagerConnectionFactory;
 import org.asteriskjava.manager.TimeoutException;
 import org.asteriskjava.manager.action.HangupAction;
 import org.asteriskjava.manager.action.OriginateAction;
 import org.asteriskjava.manager.action.RedirectAction;
 import org.asteriskjava.manager.response.ManagerResponse;

 public class TestOriginate {

 /**
  * @param args
  */
 private ManagerConnection managerConnection;

 public TestOriginate() throws IOException {
 ManagerConnectionFactory factory = new ManagerConnectionFactory(
 192.168.0.178, manager, pa55w0rd);

 this.managerConnection = factory.createManagerConnection();

 }
 public void run() {
 RedirectAction redirectAction;
 ManagerResponse originateResponse;
 String state = ;
 String receiver = 0976468586;
 redirectAction = new RedirectAction();
 redirectAction.setContext(from-smg);
 redirectAction.setExten(9220);
 redirectAction.setPriority(new Integer(1));
 redirectAction.setChannel(SIP/+ receiver);

 try {
 System.out.println(Starting login 192.168.0.178);
 managerConnection.login();

 System.out.println(After login 192.168.0.178);

 } catch (IllegalStateException e) {

 } catch (TimeoutException e) {

 } catch (IOException e) {

 } catch (AuthenticationFailedException e) {

 }
 try {
 originateResponse =
 managerConnection.sendAction(redirectAction,
 3);
 state = originateResponse.getResponse();
 System.out.println(State value is : + state);
 } catch (IllegalArgumentException e) {
 // TODO Auto-generated catch block
 e.printStackTrace();
 } catch (IllegalStateException e) {
 // TODO Auto-generated catch block
 e.printStackTrace();
 } catch (IOException e) {
 // TODO Auto-generated catch block
 e.printStackTrace();
 } catch (TimeoutException e) {
 // TODO Auto-generated catch block
 e.printStackTrace();
 }

 managerConnection.logoff();
 }

 public static void main(String[] args) throws IOException {
 // TODO Auto-generated method stub

 TestOriginate test = new TestOriginate();
 test.run();
 }

 }

 *While i run above code, the result printed on console likes following:*


 Starting login 192.168.0.178
 Jan 11, 2011 3:26:01 PM
 org.asteriskjava.manager.internal.ManagerConnectionImpl connect
 INFO: Connecting to 192.168.0.178:5038
 Jan 11, 2011 3:26:02 PM
 org.asteriskjava.manager.internal.ManagerConnectionImpl
 setProtocolIdentifier
 INFO: Connected via Asterisk Call Manager/1.1
 Jan 11, 2011 3:26:02 PM
 org.asteriskjava.manager.internal.ManagerConnectionImpl
 setProtocolIdentifier
 WARNING: Unsupported protocol version 'Asterisk Call Manager/1.1'. Use at
 your own risk.
 Jan 11, 2011 3:26:02 PM
 org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
 INFO: Successfully logged in
 Jan 11, 2011 3:26:04 PM
 org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin
 INFO: Determined Asterisk version: Asterisk 1.0
 After login 192.168.0.178
 State value is :Error
 Jan 11, 2011 3:26:04 PM
 org.asteriskjava.manager.internal.ManagerConnectionImpl disconnect
 INFO: Closing socket.
 Jan 11, 2011 3:26:04 PM org.asteriskjava.manager.internal.ManagerReaderImpl
 run
 INFO: Terminating reader thread: socket closed

 I hope you can spend your time to read what i have written above and help
 me solve this problem.

 Can you contact with me by my yahoo nick : ducphuongbk200...@yahoo.com

 Thanks and best regards.
 Phuong

 On Mon, Jan 10, 2011 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

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

[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 help me to resolve
 this. Can you say clearlier for me?

 Not really. It's a list of manager commands. There is 'SIPshowpeer' which
 will work for sip stuff. Try the command 'Command' action and you can send
 any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work
 in some cases..

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

[DHAVAL INDRODIYA]

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.

http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState

If you are looking for extension state just pass extension and you will
receive perfect response of that extension then you cans code as you want.

regards
Dhaval

On Tue, Jan 11, 2011 at 9:56 AM, Phuong Hoang
ducphuongbk200...@gmail.comwrote:

 Hi Jim,
 Really, I have`nt understood what you said yet. I am building a system on
 asterisk, and want to check a number online, offline or unreachable. If
 number is online on the extension then i want to redirect other extension.
 Redirecting is done by application java using AMI. can you help me do it?
 Thanks and best regards!
 Phuong


 On Mon, Jan 10, 2011 at 7:09 PM, Jim Dickenson dicken...@cfmc.com wrote:

 If you do an AMI packet like this:

 Action: Originate
 Channel: Local/get_i...@some_context
 Exten: do_noop
 Context: some_context
 Priority: 1
 ActionID: GetInfo
 Async: true

 and then have a couple extensions that do what you want. Here is what I do
 in my case:

 exten = get_info,1,Answer()
 exten = get_info,n,UserEvent(GetInfo,Version:ABE 
 DateTime:${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}  CfMC:83351)
 exten = get_info,n,Hangup()

 exten = do_noop,1,Answer()
 exten = do_noop,n,Wait(1)
 exten = do_noop,n,Hangup()

 You would then do what you need to do in your extensions.



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

 CfMC
 http://www.cfmc.com/



 On Jan 10, 2011, at 6:16 PM, Phuong Hoang wrote:

 Thanks Jim,
 Can you say about your idea clearlier? I want to use AMI in an application
 java to check a number online, offline or unreachable and result is returned
 to the appliction java. If the number is online now, i will use AMI to
 hangup it, else i do nothing.
 Best regards,
 Phuong.

 On Mon, Jan 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.comwrote:

 You can always place a call to an extension that sends a user event
 from AMI. If there are no native AMI commands that can return what you want
 originate a call to a local extension that returns a user event.
  --
 Jim Dickenson
 mailto:dicken...@cfmc.com dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/



 On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote:

 Thanks Dhaval,
 My purpose is that i want to use java application (using Asterisk Manager
 Interface) to check a number online, offline or unreachable. Your suggest
 uses function DEVICE_STATE but this is written in dialplan not application
 java. Do you know 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
 Dhaval


 On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes steve-li...@geekinter.net
  wrote:


 On 10 Jan 2011, at 10:37, Phuong Hoang wrote:
 I found the link you have just sent to me but it do`nt help me to
 resolve this. Can you say clearlier for me?

 Not really. It's a list of manager commands. There is 'SIPshowpeer'
 which will work for sip stuff. Try the command 'Command' action and you 
 can
 send any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might
 work in some cases..

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

[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 mentioned that which purpose youwere use queue other wise i can
give answer in better way.

regards
Dhaval

On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen p...@ibidium.no wrote:

  Hello,

 I have been asked to implement the following design:

 Load-balanced Kamailio servers handling registrations and routing.
 Load-balanced asterisk feature servers handling voicemail and other things
 Kamailio cannot do. Plus several load-balanced gateways, but they are not
 relevant to my question.

 All this is working fine.

 I've now been asked to start implementing calling queues, and my question
 is this:
 How can I implement the same queue on multiple Asterisk servers?

 Let's say that 10 people call the same queue. These calls would then
 currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make
 Asterisk A respect the 5 people queued on the other server and vice versa?

 Will the customer need to change their design to make the feature servers
 master-slave with failover instead of load-balanced?

 Mvh
 Pan

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Re: [asterisk-users] Question About Conferencing Capabilities

[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.

regards
dhaval

On Tue, Jan 4, 2011 at 10:34 AM, Siobhan Hamilton 
siobhan.plugge...@gmail.com wrote:

 My company is building a VOIP application, and initially were just using a
 barebones OpenSIPS implementation to host one-on-one calls; however, we want
 to expand the functionality to conferencing (which, of course, OpenSIPS
 doesn't handle) and was looking into Asterisk (the other option being
 Freeswitch).  I've been poring through the docs, and have even set up a test
 server myself, but there are some very specific things we are looking for
 that I can't figure out if Asterisk can do or not.

 We want to be able to do the following:
 - Create dynamic, on-the-fly conferences that can remain active even when
 initiating user leaves
 - Within a conference, give users the ability to mute and/or deaf
 individual users
 - Give users the ability to enter a whisper mode with another user -
 where they are holding a private conversation that can only be heard by the
 two of them ( It sounds like the Meetme module has a functionality like
 this, but it is a little vague in the documentation)
 - Allow users to be in two conferences at once; the user would most likely
 have one muted at any given time so as to hear the other one, but we want
 them to be able to switch back and forth easily

 Could anyone advise me on whether Asterisk can accomplish these needs, or
 perhaps what it might take to do so?  We are not averse to doing some
 customization if we can find the people who know how to make it happen!

 Thanks,
 Siobhan Hamilton

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

[DHAVAL INDRODIYA]

Guys,

I have rebooted system, and also same issue i have found that DAHDI module
is not found
i am stuck in what to do for loading DAHDI onto EC2


*/etc/init.d/dahdi restart
Unloading DAHDI hardware modules: done
Loading DAHDI hardware modules:
FATAL: Module dahdi not found.*

regards
dhaval



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 to starting it gives me following error.
  
   *No DAHDI found. Unable to open /dev/dahdi/ctl: No such file or
 directory*
  
   what could be possible suggestion.
  
 
  reboot ?

 Or rather:

  /etc/init.d/dahdi restart

 The old dahdi module may still be loaded. You may need to stop asterisk
 first, as it may be using DAHDI (for timing).

 --
Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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

[DHAVAL INDRODIYA]

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__)
#if LINUX_VERSION_CODE = VERSION_CODE(2,6,13)
/* The symbol hrtimer_forward is only exported as of 2.6.22: */
#if defined(CONFIG_HIGH_RES_TIMERS)  LINUX_VERSION_CODE =
VERSION_CODE(2,6,22)
#define USE_HIGHRESTIMER
#else
/* #define USE_RTC */
#endif
#else
#if 0
/* #define USE_RTC */
#endif
#endif
#endif

simply put USE_RTC in comment , as in 2.3.0 version we cannot find this code
into dahdi_dummy.c file,

my question is that , if USE_RTC is not in dahdi_dummy.c file can we able to
continue with dahdi.??
as it still not started on my EC2 machine.

i will keep posted for more information.

regards
Dhaval

On Tue, Dec 14, 2010 at 1:53 PM, Tzafrir Cohen tzafrir.co...@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- 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 dahdi-linux-2.1.0.4
 
  I am getting following error
 
  *echo You do not appear to have the sources for the 2.6.32.19-0.3-ec2
  kernel installed.
  You do not appear to have the sources for the 2.6.32.19-0.3-ec2 kernel
  installed.
  exit 1

 You need a kernel source (Porbably the huge kernel-source-* package is
 not required. IIRC SUSE now has a kernel-devel-* package (not sure how
 it is called. Please install it.

 That said, that version of DAHDI is rather old. Specifcally a version =
 2.3.0 may not be a good timing source in your settings.

 Newer versions of asterisk As of 1.6.1 (and even more so: as of 1.6.2)
 DAHDI is much less required (as a timing source and as a mixer for
 conference rooms).

 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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

[DHAVAL INDRODIYA]

One More thing,once i installed dahdi-2.3.0 complete it installed
successfully , but when
i tried to starting it gives me following error.

*No DAHDI found. Unable to open /dev/dahdi/ctl: No such file or directory*

what could be possible suggestion.

regards
dhaval

On Tue, Dec 14, 2010 at 2:38 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__)
 #if LINUX_VERSION_CODE = VERSION_CODE(2,6,13)
 /* The symbol hrtimer_forward is only exported as of 2.6.22: */
 #if defined(CONFIG_HIGH_RES_TIMERS)  LINUX_VERSION_CODE =
 VERSION_CODE(2,6,22)
 #define USE_HIGHRESTIMER
 #else
 /* #define USE_RTC */
 #endif
 #else
 #if 0
 /* #define USE_RTC */
 #endif
 #endif
 #endif

 simply put USE_RTC in comment , as in 2.3.0 version we cannot find this
 code into dahdi_dummy.c file,

 my question is that , if USE_RTC is not in dahdi_dummy.c file can we able
 to continue with dahdi.??
 as it still not started on my EC2 machine.

 i will keep posted for more information.

 regards
 Dhaval


 On Tue, Dec 14, 2010 at 1:53 PM, Tzafrir Cohen 
 tzafrir.co...@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- 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 dahdi-linux-2.1.0.4
 
  I am getting following error
 
  *echo You do not appear to have the sources for the 2.6.32.19-0.3-ec2
  kernel installed.
  You do not appear to have the sources for the 2.6.32.19-0.3-ec2 kernel
  installed.
  exit 1

 You need a kernel source (Porbably the huge kernel-source-* package is
 not required. IIRC SUSE now has a kernel-devel-* package (not sure how
 it is called. Please install it.

 That said, that version of DAHDI is rather old. Specifcally a version =
 2.3.0 may not be a good timing source in your settings.

 Newer versions of asterisk As of 1.6.1 (and even more so: as of 1.6.2)
 DAHDI is much less required (as a timing source and as a mixer for
 conference rooms).

 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Asterisk and Dahdi ON Amazon EC2

[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 dahdi-linux-2.1.0.4

I am getting following error

*echo You do not appear to have the sources for the 2.6.32.19-0.3-ec2
kernel installed.
You do not appear to have the sources for the 2.6.32.19-0.3-ec2 kernel
installed.
exit 1
make: *** [modules] Error 1

*
Any one have some idea how to resolve it i am also goggling about 2 hours.


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

[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 which is wrong

as from provider says that you should support E.164 ?? is that true that we
do enable E.164 as per reading from some
forums and after goggling i think that this would be a part from provider ,
they should send me call with correct callerID and
should not append 1 as prefix.

the meaning for posting this question is what is E.164 ?? and if i want to
do this then how should i start to enable this
on asterisk 1.6.2.6

give me some tips,tricks regarding issue.

hope for good help !!!

regards
dhaval
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[asterisk-users] Trixbox/Asterisk integration With SugarCRM

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

[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 not hear
anything except the ring.


that means other system picked UP a call and my SIP phone still here RINGS
when i get connected it give me that my IVRS is started and some welcome
prompt are also goes  and once i connected i got prompt for entering
something.


is this due to dialoptions I passed '*rt*' or something version related
issue with asterisk,


also i note-down one thing that once my IVRS received call from my asterisk
machine i am getting SIP 183 'session progress' not 200 OK for INVITE ,


Please help me to solve out this suggest some DIAL OPTIONS or some setting
in SIP if i am missing.

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

[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 very strange  with me on asterisk. how this possible even if i want
to explain to user in technical terms.
i don't know user is using which PBX system.

i think there is one possibility which i think User entered a number but i
do not receive anything and user will try to re-enter number again
in this time user PBX will redirect call to extension with 16


let give your thoughts regarding this.

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

[DHAVAL INDRODIYA]

Hi Sherwood ,

well , i think you did not understand my question , i want real time billing
like as i mentioned that if i want to dial 5 number with different call rate
how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think.

give your thougts.

regards
dhaval

On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan 
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 in real
 time below are scenario and explanation.


 Scenario:
  A customer called my DID number and after that from here i dial few
 number let say 5 number. once number are placed into DIAL
 i will put this customer into conference [MEETME] , once a Members are
 picked up call they will also patched into conference and
 talking is started, every thing working fine with DIAL-PLAN and DB look
 up.

 Now, i want to do billing on customer dialed my DID, and from that
 actually it DIALED 5 numbers, how can i DO real time billing
 into this situation, like numbers can be different It can be ISD,STD,Local
 and also free .

 if customer having initial balance of $100 then how can i check balance
 every time.in a situation once balance is nil then i want to disconnect
 calls . is any one facing this type of situation.

 give me some  idea ,

 regards
 Dhaval


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 Dhaval,
 This sounds very much like a system I'm working on for a client right now.
 I'm not permitted to disclose much about it due to the NDA i signed, but
 I'll risk giving you a point in the right direction.

 First, you should create a table in your database that has a column called
 callid, and other columns that you will have to decide upon. This table will
 be called something like '*call_references*'. Oh, and you'll want to
 define callid as the primary key for records in that table, but DO NOT make
 it an autoincrement, you're going to populate it with a value that is
 described in the next step.

 Second, at the beginning of the original call you mentioned, define a
 variable that will be unique to that call. I personally have done this by
 stripping all non-digits from the caller's callerid (using
 Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
 to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.

 Next (this is where I have to start being a bit vague), you're going to
 perform an INSERT query, creating a new call_references record (using that
 variable I just showed you how to construct as callid's value).

 Now, when you defined that variable, you should have preceded the variable
 name with two underscores ( __ ), which will tell Asterisk that channels
 spawned by the current channel will inherit that variable and it's value.

 Voila, you now have a method for storing realtime data such as billing
 information between MULTIPLE calls.

 I wish I could tell you more, but I can't violate my client's
 Non-Disclosure Agreement.

 Hope this helps you out!

 Sherwood McGowan


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

[DHAVAL INDRODIYA]

thanks mate,

for useful and good information provided by you, i am not asking you that
please write down your all LOGIC and explain everything to me, as per your
explanation i can see it will deduct amount for only 1 call but what
actually i am searching for is if user made 5 concurrent calls and i have to
limit
all calls and each destination number having different rate may be some of
them ISD and some of them local. that will create more problem to me, i
think there is some solutions for this . could you suggest any reference for
the same, it will be more helpful to me.

thanks in advance,
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 mentioned that if i want to dial 5 number with different call
 rate how can i access same
 balance into those 5 people, if all are connected how can i periodically
 update billing , as you suggested it will assign total balance to those 5
 people but actually we can not do like this as total balance of user $100 ,
 as per your suggestion it will give $100 for those 5 people which is
 practically wrong i think.

 give your thougts.

 regards
 dhaval


 On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan 
 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 in
 real time below are scenario and explanation.


 Scenario:
  A customer called my DID number and after that from here i dial few
 number let say 5 number. once number are placed into DIAL
 i will put this customer into conference [MEETME] , once a Members are
 picked up call they will also patched into conference and
 talking is started, every thing working fine with DIAL-PLAN and DB look
 up.

 Now, i want to do billing on customer dialed my DID, and from that
 actually it DIALED 5 numbers, how can i DO real time billing
 into this situation, like numbers can be different It can be
 ISD,STD,Local and also free .

 if customer having initial balance of $100 then how can i check balance
 every time.in a situation once balance is nil then i want to disconnect

 calls . is any one facing this type of situation.

 give me some  idea ,

 regards
 Dhaval


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



 Dhaval,
 This sounds very much like a system I'm working on for a client right
 now. I'm not permitted to disclose much about it due to the NDA i signed,
 but I'll risk giving you a point in the right direction.

 First, you should create a table in your database that has a column
 called callid, and other columns that you will have to decide upon. This
 table will be called something like '*call_references*'. Oh, and you'll
 want to define callid as the primary key for records in that table, but DO
 NOT make it an autoincrement, you're going to populate it with a value that
 is described in the next step.

 Second, at the beginning of the original call you mentioned, define a
 variable that will be unique to that call. I personally have done this by
 stripping all non-digits from the caller's callerid (using
 Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
 to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.

 Next (this is where I have to start being a bit vague), you're going to
 perform an INSERT query, creating a new call_references record (using that
 variable I just showed you how to construct as callid's value).

 Now, when you defined that variable, you should have preceded the
 variable name with two underscores ( __ ), which will tell Asterisk that
 channels spawned by the current channel will inherit that variable and it's
 value.

 Voila, you now have a method for storing realtime data such as billing
 information between MULTIPLE calls.

 I wish I could tell you more, but I can't violate my client's
 Non-Disclosure Agreement.

 Hope this helps you out!

 Sherwood McGowan


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

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

[DHAVAL INDRODIYA]

Hi Arjan,

i am able to solve this problem after adding this patch and adding
unanswered=yes onto cdr.conf

https://issues.asterisk.org/file_download.php?file_id=24431type=bug

regards
Dhaval

On Tue, Oct 5, 2010 at 1:12 PM, Arjan Kroon | Mobillion 
arjan.kr...@mobillion.nl wrote:

  Hi Dhaval,



 I ‘m in the almost same situation.

 I’ve already post a issue with asterisk.

 https://issues.asterisk.org/view.php?id=17826





 Is you only use an originate and not an originate en then redial maybe this
 link helps you further.

 https://issues.asterisk.org/view.php?id=17592nbn=16#bugnotes



 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 originated from CLI
 originate



 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

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

[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 INDRODIYA пишет:
 
  Hi Nickolay,
 
  here i attached my file. please have a look into it.

 Hello DHAVAL

 As I wrote your file has wrong format.

  $ file ask-propertyid.WAV
   ask-propertyid.WAV: RIFF (little-endian) data, WAVE audio,
   GSM 6.10, mono 8000 Hz

 See GSM 6.10 there. You need to convert it to PCM

  sox ask-propertyid.WAV -e signed-integer ask-propertyid-converted.WAV

 Then decode.


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

[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.

 Let me respectively disagree with you. It's perfectly possible even with
 open source tools. You can download pocketsphinx from

 http://cmusphinx.sourceforge.net

 To convert speech to text you need to download Communicator acoustic
 telephone model and LM giga large vocabulary language model.


 http://www.speech.cs.cmu.edu/sphinx/models/communicator_mar2008/communicator_semi_6000_20080321.tar.gz
 http://www.keithv.com/software/giga/

 --
 Nexiwave - Speech Mining Solution For Call Centers
 http://nexiwave.com



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

[DHAVAL INDRODIYA]

is it possible with lumenvox i will purchase liceance

regards
Dhaval

On Tue, Sep 14, 2010 at 4:20 PM, Zeeshan Zakaria zisha...@gmail.com wrote:

 In theory it should work but in real life it doesn't. Converting reliably
 half an hour of speech into text is simply a dream.

 Zeeshan A Zakaria

 --
 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 snapshot. Then run

 # sphinx_lm_sort   lm_giga_20k_nvp_3gram.arpa 
 lm_giga_20k_nvp_3gram.arpa.sorted

 # sphinx_lm_convert -i lm_giga_20k_nvp_3gram.arpa.sorted -o
 lm_giga_20k_nvp_3gram.lm.DMP

 This will create a language model lm_giga_20k_nvp_3gram.lm.DMP

 And finally convert audio

 pocketsphinx_continuous -infile your_audio_file.wav -samprate 8000 \
 -hmm Communicator_semi_40.cd_semi_6000 -lm lm_giga_20k_nvp_3gram.lm.DMP


 --
 Nexiwave - Speech Mining Solution For Call Centers
 http://nexiwave.com

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

[DHAVAL INDRODIYA]

Hello i have tried to convert through sphinx as suggested by Nickolay

i am not getting convert my simple audio file.

i am having following error while i fire following command


pocketsphinx_continuous -infile /usr/etc/ask-propertyid.WAV -samprate 8000 \
-hmm /usr/etcSpeechToText/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
 dhaval.it01...@gmail.com wrote:
  - Call comes in
  - start recording
  - call remains for 30 minutes
  - stop recording
  - convert wav file audio to text.
 
  is this possible with lumenvox or any other engine.
 
 Not realistically, because you need to define grammars into your
 speech engine, it would take a large amount of work to set this up.
 In the past when this has been a customer requirement, I have had to
 hire a transcribing service for my audio file.

 --
 Paul Belanger | dCAP
 Polybeacon | Consultant
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 blog.polybeacon.com

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

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

[DHAVAL INDRODIYA]

Thanks Paul,

i think still i have some problem to understand , i mean to say that i have
30 minutes audio file in
WAV format and i wnat its text here are the scenario .

- Call comes in
- start recording
- call remains for 30 minutes
- stop recording
- convert wav file audio to text.

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 LumenVox
  convert to text ?
 
 ASR, yes.

 http://www.digium.com/en/products/software/lumenvox.php

 --
 Paul Belanger | dCAP
 Polybeacon | Consultant
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 blog.polybeacon.com

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

[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 kamailio and everything is working fine

now question is if i want add one more domain like abc.com so for that i
need to add every entry in /etc/hosts file.

is there anyway to resolve it out, Means if SIP wants to send each call to
192.168.1.30 , but without entry in /etc/hosts.


regards
Dhaval.
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[asterisk-users] [Asterisk-User] Asterisk Video support

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

[DHAVAL INDRODIYA]

hi Motiejus,

Can you give a command for converting it to normal voice , in audacity.

also i tired with more users still problem persists ,

can i try with gsm format , what you say?

regards
Dhaval

2010/5/18 Motiejus Jakštys desired@gmail.com

 Hi,
 The record is not double faster, it's 50% faster (100 seconds original
 record - 66.6 seconds recording). Reducing tempo by 33% without
 losing pitch sort of fixes the situation, although adds alot garbage
 to sound file (you can do this in Audacity).
 Sample rate 8kHz is OK, changing it to 5280 fixes the tempo, but
 reduces pitch to unacceptable.

 Try with more callers in a conference, does it change anything
 (increased/decreased tempo)?

 You could also try ConfBridge:
 http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge
 or other conference backends (Conference, Konference...)
 These could solve the problem if Dahdi 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 Jakštys desired@gmail.com:
  Please check WAV headers, what is the sample rate of the file? It
  should be 8kHz. Does the WAV sound normal when you decrease sample
  rate by hand?
 
  You can just upload one WAV for testing - I'll say what may be wrong with
 it.
 
  On Tue, May 18, 2010 at 9:52 AM, DHAVAL INDRODIYA
  dhaval.it01...@gmail.com wrote:
  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 conference starts recording file having a   recording is 2x faster
 than
  normal recording .
 
  is there any setting to solve it out , my card type is TE410P used E1
 lines
  .
 
  please help me . any help appreciated.
 
  regards
  Dhaval
 
 
 
 
 
 
 
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[asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.

[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 conference starts recording file having a   *recording is 2x faster *than
normal recording .

is there any setting to solve it out , my card type is *TE410P* used E1
lines .

please help me . any help appreciated.

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

[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 means whole
xml,json request  is represented in string format, how can i parse that
request?

my question is that is there any  best utility in asterisk that support
calling a webservie from Dialplan?

i am also comfortable with C, or PERL based AGI.

please guide me as i am new to this Webservice part...

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

[DHAVAL INDRODIYA]

Dear All,

i have following CLI error while try to run this command from Dialplan

*TrySystem(DAHDI/45-1, asterisk -rx dialplan add extension
1234111,1,Goto(incomingdundi,s,1) into dundilookup) in new stack

 WARNING[32626]: app_system.c:81 system_exec_helper: Unable to execute
'asterisk -rx dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into
dundilookup'*

where as I am using Asterisk 1.6.0.5 and my machine is using
*safe_asterisk*script asterisk running

after abnormally terminated asterisk safe_asterisk restart it then i am
getting this error on CLI , i want to know the reason 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.
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Re: [asterisk-users] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM

[DHAVAL INDRODIYA]

As it is happening in our production server we can not upgrade it ,
 moreover there is no change in app_system in 1.6.0.25 compared to 1.6.0.5
any other help appreciated.

regards
Dhaval


On Thu, Mar 18, 2010 at 12:15 AM, Barry L. Kline blkl...@attglobal.netwrote:

 -BEGIN PGP SIGNED MESSAGE-
 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.

 Barry


 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.5 (GNU/Linux)

 iD8DBQFLoSMvCFu3bIiwtTARAvDNAJ4ql+42gKH20vMAJLNsYVxqqOhMjgCfRuF9
 R6QAJbu5ZSHmJVSkO7UErmY=
 =VYx9
 -END PGP SIGNATURE-

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[asterisk-users] PBX_DUNDI question

[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 = s,n,Set(ID=${DUNDIQUERY(${NUMBER},priv,b)})
exten = s,n,NoOp(DUNDI-QUERY-ID [ ${ID} ])
exten = s,n,Set(NUM=${DUNDIRESULT(${ID},getnum)})
exten = s,n,NoOp(There are [ ${NUM} ] dundi results)

regards
Dhaval
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[asterisk-users] Echo cancellation on DAHDI

[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
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Re: [asterisk-users] Echo cancellation on DAHDI

[DHAVAL INDRODIYA]

Hi,

Carlos

I checked dmesg on my server and i found following message

what is meaning for this ? i cant understand

VPM400: Not Present
VPM450: echo cancellation for 128 channels
VPM450: hardware DTMF disabled.
VPM450: Present and operational servicing 4 span(s)

regards
Dhaval
On Tue, Mar 2, 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 but sometimes i got echo when user are calling a PRI.
 
  is there any way to know on board echo cancellation .
 
 
 Check dmesg on your system for messages like:

 VPM400: Support Enabled/Disabled
 VPM450: Support Enabled/Disabled

That should tell you if the hardware echo cancellation is working or
 not.  The TE410P does not have hardware echo cancellation the model was
 TE411P.  If you can open the server you should be able to see if the
 card has a daughter board installed which is the echo module.

  regards
 
  Dhaval
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 --
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 Carlos Chávez Prats
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 +52-55-91169161 ext 2001

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

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

 --
Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] AMI originate and PHP

[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 can start with something that
 is tested to work.

 I really appreciate your response.

 In the meantime, I will go check meetme creation rules.

 Regards,
 Bruce


 On Wed, Dec 23, 2009 at 7:03 PM, Jarrod Lash jar...@fed-com.com wrote:

 Bruce,

 What I have done for apps like this is call the first guy and at the
 end of your dialplan put him in a meetme room.  In your script watch
 for the meetme room to be created in the AMI output.

 Once the room is created originate a call to the other guy and dump
 him into that meetme room when he answers.


 --
 Jarrod Lash, jar...@fed-com.com
 Federated Communications, LLC.
 www.fed-com.com
 Office: +1-412-357-2127
 Mobile: +1-412-999-0049
 Fax: +1-412-545-8368


 On Wed, Dec 23, 2009 at 6:19 PM, Bruce Nik brucev...@gmail.com wrote:
  Hi Guys,
  I am trying to make a web form where a person is allowed to put in
  $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof
 caller
  ID. There are a few problems that I am facing with Asterisk AMI
 Originate
  command. The reason why I want to use the darn AMI Originate is because
 I am
  sending calls to mobile phones and I want to have some accountability
 and to
  know if a call was connected for billing purposes or not. Calls go to
 PSTN
  through SIP provider so all signaling is available.
  First, if i use AMI Originate to dial both parties with the set CallerID
  then, one party may pick up than the other and channel is not bridged at
  ringing. So, this can confuse the callee. So, I thought I should send
 calls
  to a context first and then ask customer enter $spoofNumber and then
 place
  call but then I am facing another problem. Using that, the internal
 context
  is called first and all announcements are made and then the
  SIP/sipProvider/$phoneNumb is dialed. Or at least it's dialed at the
 same
  time but since it takes time to pick ones phone context already goes
 over
  it's announcement for putting in spoof number and dialnumber. Please
 guide
  me how to do this properly. Following is the code and the context:
  $sys_ip = 127.0.0.1;
  $User_str = test;
  $Secret_str = test;
  $phoneNumb = 1416777;
  $dialNumb = 1416888;
  $spoofNumb = 141699;
  $oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or
 die(Connection to
  host failed);
  fputs($oSocket, Action: login\r\n);
  fputs($oSocket, Username: $User_str\r\n);
  fputs($oSocket, Secret: $Secret_str\r\n\r\n);
  fputs($oSocket, Events: off\r\n\r\n);
  fputs($oSocket, Action: originate\r\n);
  fputs($oSocket, Channel: SIP/testTrunk/$phoneNumb\r\n);
  fputs($oSocket, Exten: $dialNumb\r\n);
  fputs($oSocket, Context: testphp\r\n);
  fputs($oSocket, Priority: 1\r\n\r\n);
  fputs($oSocket, Timeout: 1\r\n);
  fputs($oSocket, CallerId: $spoofNumb\r\n);
  fputs($oSocket, Async: true\r\n);
  fputs($oSocket, Action: Logoff\r\n\r\n);
  fclose($oSocket);
 
  /etc/asterisk/extensions.conf
  [testphp]
  exten = _X.,1,Answer()
  exten =
 
 _X.,n,Playback(/var/lib/asterisk/sounds/please_enter_dialnumber_and_spoof_callerid)
  exten = _X.,n,Read(dialnumber,,10)
  exten = _X.,n,Read(spoofnumber,,10)
  exten = _X.,n,Playback(connecting_now)
  exten = _X.,n,Dial(SIP/testTrunk/$dialNumb)
  exten = _X.,n,Hangup()
  Thanks a lot.
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[asterisk-users] Asterisk Heartbeat Monitor for Fail safe.

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

[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 is a meaninglessly vague
 description.  That's like saying, Is there any way I can ride
 car/elephant-like transportation?

  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.

 You would want to use a tool like SIP Swiss Army Knife (sipsak) to check
 for basic SIP responsiveness.  It can be wrapped inside a custom OCF
 resource agent script, if you're using Heartbeat v2.

 --
 Alex Balashov - Principal
 Evariste Systems
 Web : http://www.evaristesys.com/
 Tel : (+1) (678) 954-0670
 Direct  : (+1) (678) 954-0671

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[asterisk-users] SIP_CODEC related question

[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 dialing.
 ex: In 1.0 there is some variable called SIP_CODEC which can be setted
.
   what about newer version like 1.6 or greater.

is anybody know regarding this

regards
Dhaval..
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