Re: [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41
Yeah, that's what I was saying J good it fixed it. BR Gohar From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Friday, July 06, 2012 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41 Thanks Gohar, I found the issue was copy file to outbound folder not moving. that's why after making future time asterisk start reading file. On Fri, Jul 6, 2012 at 11:16 AM, SamyGo govoi...@gmail.com wrote: Hi, Did you get anything working on it !! See the permission for the user running asterisk process and see if that user can touch files like that. Regards, Sammy On Thu, Jul 5, 2012 at 10:47 PM, virendra bhati virbh...@gmail.com wrote: Hi All, It's small issue but making a big problem for my application. I have CentOS release 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41 because Flite work in this version. problem is that when I make changes on .call file to make it future call file with touch command then it not changed. [root@server tmp]# touch -t 201207052137 1341509545.39.call [root@server tmp]# ll -rw-r--r-- 1 root root 52 Jul 5 2012 1341509545.39.call .call file's time is missed with year only that's asterisk make call after move to outgoing folder. please give your suggestion. If I am wrong then correct me ... -- Thanks and regards Virendra Bhati +91-9718300881 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk RTCP
Hello list, Kevin I agree with you on independent monitored entity for A leg while the outbound leg has separate QoS measures. But after this thread I went to my monitoring tool and saw that for some calls on the same asterisk setup I had no RTP or RTCP while there were calls with both RTP and RTCP captured as well. Since I've a SIP proxy on top of asterisk servers layers, could it be possible that RTP and RTCPs bypass asterisk (media redirect) and that's why I see RTCPs and RTPs logged into monitoring tool while those call who couldn't redirect/bypass media from asterisk don't show any RTCPs!? Sammy can you provide further details of your setup please! Regards, Gohar -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, February 17, 2012 5:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk RTCP On 02/17/2012 12:09 AM, Sammy Govind wrote: Hello, Thanks for taking out tome for my query. Yes I do have an actual problem. I've a monitoring tool to record the VoIP QoS (Asterisk servers port mirrored to it). My end points(soft-phones) are sending RTCP connection strings to asterisk, and Asterisk then forwards their call to their destination choosing any suitable carrier. If I don't get RTCP flowing through asterisk the monitoring tool simply fails to display and call stats. Please advice what should I be doing to cater this. As I said before, you will never get RTCP *flowing through* Asterisk. When your softphone calls Asterisk, that will be a separate call leg from the one from Asterisk to your provider. Your monitoring tool should treat those as separate call legs and produce an analysis for them independently. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf
Hi, Create a context in AEL, or LUA and change the context=ael-context or context=lua-context in sip.conf [default] section or for each sip user decalred who needs to start call in context defined in AEL/LUA? Regards, Gohar From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 23, 2011 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind Subject: [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf Hi List, I want to change the asterisk flow. right now call startd from extensions.conf. Is there any way by which we can changed it to extensions.ael or extensions.lua ? - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN connectivity
Hey, How've you configured your Outbound trunk ? DAHDI/1/04712527270 : What do you've in your dahdi configuration file ! I doubt this /1 is the culprit or else your DAHDI channel is not really working at all. Regards, Gohar A. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PSTN connectivity Hi All, I got a busy message like all lines are currently busy and please try again later in call to ZAP trunk. Please help me to resolve this issue == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [904712527270@from-internal:1] Macro(SIP/157-, user-callerid,SKIPTTL,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/157-, AMPUSER=157) in new stack -- Executing [s@macro-user-callerid:2] GotoIf(SIP/157-, 0?report) in new stack -- Executing [s@macro-user-callerid:3] ExecIf(SIP/157-, 1?Set(REALCALLERIDNUM=157)) in new stack -- Executing [s@macro-user-callerid:4] Set(SIP/157-, AMPUSER=157) in new stack -- Executing [s@macro-user-callerid:5] Set(SIP/157-, AMPUSERCIDNAME=Rojar S) in new stack -- Executing [s@macro-user-callerid:6] GotoIf(SIP/157-, 0?report) in new stack -- Executing [s@macro-user-callerid:7] Set(SIP/157-, AMPUSERCID=157) in new stack -- Executing [s@macro-user-callerid:8] Set(SIP/157-, CALLERID(all)=Rojar S 157) in new stack -- Executing [s@macro-user-callerid:9] ExecIf(SIP/157-, 0?Set(CHANNEL(language)=)) in new stack -- Executing [s@macro-user-callerid:10] GotoIf(SIP/157-, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set(SIP/157-, CALLERID(number)=157) in new stack -- Executing [s@macro-user-callerid:20] Set(SIP/157-, CALLERID(name)=Rojar S) in new stack -- Executing [s@macro-user-callerid:21] NoOp(SIP/157-, Using CallerID Rojar S 157) in new stack -- Executing [904712527270@from-internal:2] Set(SIP/157-, _NODEST=) in new stack -- Executing [904712527270@from-internal:3] Macro(SIP/157-, record-enable,157,OUT,) in new stack -- Executing [s@macro-record-enable:1] GotoIf(SIP/157-, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf(SIP/157-, 0?MacroExit()) in new stack -- Executing [s@macro-record-enable:5] GotoIf(SIP/157-, 0?Group:OUT) in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf(SIP/157-, 0?IN) in new stack -- Executing [s@macro-record-enable:16] ExecIf(SIP/157-, 1?MacroExit()) in new stack -- Executing [904712527270@from-internal:4] Macro(SIP/157-, dialout-trunk,1,04712527270,,) in new stack -- Executing [s@macro-dialout-trunk:1] Set(SIP/157-, DIAL_TRUNK=1) in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf(SIP/157-, 0?sub-pincheck,s,1) in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf(SIP/157-, 0?disabletrunk,1) in new stack -- Executing [s@macro-dialout-trunk:4] Set(SIP/157-, DIAL_NUMBER=04712527270) in new stack -- Executing [s@macro-dialout-trunk:5] Set(SIP/157-, DIAL_TRUNK_OPTIONS=tr) in new stack -- Executing [s@macro-dialout-trunk:6] Set(SIP/157-, OUTBOUND_GROUP=OUT_1) in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf(SIP/157-, 0?nomax) in new stack -- Executing [s@macro-dialout-trunk:8] GotoIf(SIP/157-, 0?chanfull) in new stack -- Executing [s@macro-dialout-trunk:9] GotoIf(SIP/157-, 0?skipoutcid) in new stack -- Executing [s@macro-dialout-trunk:10] Set(SIP/157-, DIAL_TRUNK_OPTIONS=) in new stack -- Executing [s@macro-dialout-trunk:11] Macro(SIP/157-, outbound-callerid,1) in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf(SIP/157-, 0?Set(CALLERPRES()=)) in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf(SIP/157-, 0?Set(REALCALLERIDNUM=157)) in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf(SIP/157-, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set(SIP/157-, USEROUTCID=) in new stack -- Executing [s@macro-outbound-callerid:7] Set(SIP/157-, EMERGENCYCID=) in new stack -- Executing [s@macro-outbound-callerid:8] Set(SIP/157-, TRUNKOUTCID=) in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf(SIP/157-, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf(SIP/157-, 0?Set(CALLERID(all)=)) in new
Re: [asterisk-users] record calls of specific agnets
Hi, I think use of any Macro in queue can serve you well. Macro will be called whenever the call is established to the agent. In that Macro check your desired Agent and if condition matched trigger MixMonitor else do nothing. Regards, Gohar A. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle McKarns Sent: Friday, September 30, 2011 5:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] record calls of specific agnets On 09/29/2011 12:37 PM, A J Stiles wrote: On Thursday 29 September 2011, Lyle McKarns wrote: Hello Asterisk List! I have been asked to record calls from specific agents, and I am having difficulty finding if this is possible, and if so, how exactly to do it. Have a recorded context and an unrecorded context in your dialplan, identical save for the lines that start the recording and cleanup processes being absent from the latter. Then set contexts per extension in sip.conf. But this seems to be a way to record a whole queue (any call in that queue), not just calls to specific agents, which is what I need. Is there something I am missing in this explanation? Thanks, Lyle J. McKarns --- Network Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax: 1 207 725 8552 Nexus Management, Inc.│ Registered Office: 4 Industrial Parkway, Suite 101, Brunswick, Maine. 04011│Company No. 19891257D, Registered in Maine│ A member of the Nexus Management Plc group of companies -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add PinCode on my dialplan
Dear Malvin,
I see Sam worked hard to post you the whole info about the application where
it clearly states the use of option a - Please change the configuration
line accordingly now and see if it works for you.
Best Regards,
Gohar
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin Rito
Sent: Thursday, September 22, 2011 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Add PinCode on my dialplan
Hi,
I tried Authenticate where pass codes are stored on the file pass.conf and
it works.
exten = _,1,Authenticate(/etc/asterisk/pass.conf)
Since I have my CDR, I want to have a report wherein I can check which pass
code did the call. How can I achieve it?
Using authenticate through file does not replace ACCOUNT_CODE field with the
pass code entered, it only show ast_h323 under the Account_Code field.
Regards,
Malvin
On 9/21/2011 1:01 PM, Sam Govind wrote:
See core show application autheTAB
If passwords are already the same as those of voicemail.conf go for
application VMAuthenticate() - DIA generates a dial-tone which I don't think
is suitable for dialling out from users(insiders)
-= Info about application 'Authenticate' =-
[Synopsis]
Authenticate a user
[Description]
This application asks the caller to enter a given password in order to
continue
dialplan execution.
If the password begins with the '/' character, it is interpreted as a file
which contains a list of valid passwords, listed 1 password per line in the
file.
When using a database key, the value associated with the key can be
anything.
Users have three attempts to authenticate before the channel is hung
up.
[Syntax]
Authenticate(password[,options[,maxdigits[,prompt]]])
[Arguments]
password
Password the user should know
options
a: Set the channels' account code to the password that is entered
d: Interpret the given path as database key, not a literal file
m: Interpret the given path as a file which contains a list of account
codes and password hashes delimited with ':', listed one per line in the
file. When one of the passwords is matched, the channel will have its
account code set to the corresponding account code in the file.
r: Remove the database key upon successful entry (valid with 'd'
only)
maxdigits
maximum acceptable number of digits. Stops reading after maxdigits
have been entered (without requiring the user to press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
prompt
Override the agent-pass prompt file.
[See Also]
VMAuthenticate(), DISA()
On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito mr...@mail.altcladding.com.ph
wrote:
Thanks. ?If I want to use unique PIN for every user that dials out how can I
implement it using Authenticate app?
Regards,
Malvin
On 9/21/2011 12:42 PM, Sam Govind wrote:
DISA and DB based Auth could be an overkill.
Kyle showed the very simplistic dial plan if Dial-out pin is common for the
whole system.
See application Authenticate(password[,options[,maxdigits[,prompt]]] and if
Voicemail PIN are required to be used use application
MAuthenticate([mailbox][@context][,options]
Regards,
- Sammy
On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton k...@mocker.org wrote:
Something like this should work:
exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)
exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
exten = _011.,n,Hangup
exten = _011.,n,Playback(invalid)
exten = _011.,n,Hangup
Could be cleaned up (the GotoIf isn't very descriptive about where it's
going), but it's a starting point.
On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
Hi List,
I currently have a asterisk server running used for dialing-out for IDD but
I want to Put a pincode wherein only users with the right pin code will be
allowed to dial IDD. Any sample dialplan you can suggest pls?
Thanks,
Malvin
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Re: [asterisk-users] Console Stereo - One call per ear
Couldn't help LOL on Kyle's remarks. But it could be two users listening to two different streams/calls. Obviously both can't share single mic on their call(if they ever need it). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kyle Sexton Sent: Wednesday, September 21, 2011 8:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Console Stereo - One call per ear I have no solution, but my head hurts thinking about listening to separate calls simultaneously in each ear. On Sep 9, 2011, at 9:22 AM, fhirschberg wrote: Hi list! I'm using the latest Asterisk 1.8.6.0 cross compiled for an i.MX27 board and it works really good. But I need a feature and don't know how to do this. What I need is the ability to have 2 separate calls on each ear on the console channel. Is there a way to get this working? It should be possible to have one call on both ears or, if another call is made, to hear this on one (selectable L/R) ear, while the other call stays on the other ear. Do I need a new console driver? I'm currently using chan_alsa and I already have Alsa devices for left, right and left + right output. It would be great if anybody can help with informations or tips where to start with my problem. Greetings Florian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] broadcast
Hey there
You are not moving the call file to spool/outgoing directory. Maybe that's
why you aren't getting anything. I don't feel good about the call file also.
Its not doing what you want it to do.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Tuesday, September 13, 2011 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] broadcast
Hi List,
I make a script for .call file and then I started playback on local channel
but nothing was hearing at another channles.
exten = 1234,1,Answer()
exten = 1234,n,System(echo -e Channel: Channel:
local/23@contest-call\\nContext: contest-call\\nExtension: 23\\nPriority: 1
/tmp/${UNIQUEID}.call)
exten = 1234,n,Konference(43689956,ADMRSTVL)
[contest-call]
exten = _X!,1,Answer()
exten = _X!,n,Set(p=/var/spool/asterisk/monitor/)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que3)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que4)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que5)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que6)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que7)
exten = _X!,n,Konference(43689956,ADMRSTV)
exten = _X!,n,Wait(10)
exten = _X!,n,Hangup()
in it I am dialing 1234 from softphone then join to conf in mute mode, after
it .call file start playback at it's own channels but I am not able to hear
anything into conf.
As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?
On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:
Good to know,
I think it'll be a feedback score or a poll from members of the conference.
So if you use the R option and collect DTMF from members, and an AMI script
listening to that particular DTMF event collects all. This way your AMI
listener script should be able to tell you at the end of poll what user
inserted with DTMF.
So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script should
theoretically work for you.
On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.com wrote:
Hi Sam,
You are right. I am looking for the same
On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind govoi...@gmail.com wrote:
IMHO, I think Bhaati is trying to get feedback from multiple conference
users. See DTMF options in Konference module.
'R' : enable DTMF relay: DTMF tones generate a manager event
If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
members in the conference
While some file is played and users press any DTMF collect the AMI events
from each user and use them as you require.
Ref: http://main.voiptoday.org/index.php?option=com_content
http://main.voiptoday.org/index.php?option=com_contentview=articleid=566:
asterisk-conferencing-module-appkonference-16-is-now-availablecatid=35:gene
ralItemid=173
view=articleid=566:asterisk-conferencing-module-appkonference-16-is-now-av
ailablecatid=35:generalItemid=173
On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati virbh...@gmail.com wrote:
Hi Ahmed,
Konference is also an conferencing application.
On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed gohar.ah...@vopium.com wrote:
Hhhmmm..I dunt have any experience with module Konference. Maybe anyone else
can help you on that.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, September 12, 2011 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] broadcast
Hi Ahmed,
I did the same thing earlier to test the load of Digium card. But this time
I want to play file and want to get some DTMF from all the members of
conference.
So in this case I need more control into Konference module. But when I use
.call files then control will not go longer with all events.
Is there any alternate way to do it?
I appreciate your suggestion and will doing in parallel at higher priority
On Mon, Sep 12, 2011 at 12:33 PM, Gohar Ahmed gohar.ah...@vopium.com
wrote:
Make a .call file..join one leg to local extension which plays the file and
the other leg to conference. The local extension will be like a conference
member.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, September 12, 2011 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] broadcast
Hi List,
Is there any way by which I can broadcast any audio file to all members into
the conference ?
I don't want to play file individual channels.
--
-
Thanks and regards
Virendra Bhati
+91-9172341457 tel:%2B91-9172341457
Software Engineer
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Re: [asterisk-users] How to know how many user is connected
Google linux commands for the purpose.
Not sure about preemptively disconnecting sockets . I think there are
commands like ss in linux which you can use. You need to collect info from
AMI and then use combination of linux commands via php directly to
disconnect anyone (if possible).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, August 25, 2011 6:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many user is connected
Hi Ahmed,
Just realized that maybe youre talking about disconnecting any other
AMI/manger connected user from another manager connection
hhmmm
I dont
think so. Check AMI commands from asterisk wiki. If not, you may need system
command in your AMI connection to close some other socket
Yes I was looking for this :)
Please tell me how to close other socket from current sockets.
one more thing in my case it may be possible that
root 127.0.0.1 may be more then one then how to close them individually?
On Thu, Aug 25, 2011 at 5:09 PM, Gohar Ahmed gohar.ah...@vopium.com wrote:
Just realized that maybe youre talking about disconnecting any other
AMI/manger connected user from another manager connection
hhmmm
I dont
think so. Check AMI commands from asterisk wiki. If not, you may need system
command in your AMI connection to close some other socket.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gohar Ahmed
Sent: Thursday, August 25, 2011 4:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to know how many user is connected
What I understood: you need to disconnect the AMI socket.
1) I want to disconnect connected manager into Asterisk. Is it possible ?
ß Close the $socket after you get the response.
What I understood: you need to maintain the socket until some button is
pressed to stop AMI
2) I want to maintain this socket connection until we disconnect it from web
page. ß Close the $socket on particular action from web-page. This ones
tricky btw maintain a while loop and break loop on a condition toggled by
web-page)
See php section for other examples.
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, August 25, 2011 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many user is connected
Hi List,
Thanks now I am able to get all values from asterisk CLI but I want 2 more
things .
1) I want to disconnect connected manager into Asterisk. Is it possible ?
2) I want to maintain this socket connection until we disconnect it from web
page.
On Thu, Aug 25, 2011 at 1:57 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
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
Im 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
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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to know how many user is connected
What I understood: you need to disconnect the AMI socket.
1) I want to disconnect connected manager into Asterisk. Is it possible ?
ß Close the $socket after you get the response.
What I understood: you need to maintain the socket until some button is
pressed to stop AMI
2) I want to maintain this socket connection until we disconnect it from web
page. ß Close the $socket on particular action from web-page. This ones
tricky btw maintain a while loop and break loop on a condition toggled by
web-page)
See php section for other examples.
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, August 25, 2011 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many user is connected
Hi List,
Thanks now I am able to get all values from asterisk CLI but I want 2 more
things .
1) I want to disconnect connected manager into Asterisk. Is it possible ?
2) I want to maintain this socket connection until we disconnect it from web
page.
On Thu, Aug 25, 2011 at 1:57 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
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
Im 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
-- _
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to know how many user is connected
Just realized that maybe youre talking about disconnecting any other
AMI/manger connected user from another manager connection
hhmmm
I dont
think so. Check AMI commands from asterisk wiki. If not, you may need system
command in your AMI connection to close some other socket.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gohar Ahmed
Sent: Thursday, August 25, 2011 4:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to know how many user is connected
What I understood: you need to disconnect the AMI socket.
1) I want to disconnect connected manager into Asterisk. Is it possible ?
ß Close the $socket after you get the response.
What I understood: you need to maintain the socket until some button is
pressed to stop AMI
2) I want to maintain this socket connection until we disconnect it from web
page. ß Close the $socket on particular action from web-page. This ones
tricky btw maintain a while loop and break loop on a condition toggled by
web-page)
See php section for other examples.
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, August 25, 2011 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many user is connected
Hi List,
Thanks now I am able to get all values from asterisk CLI but I want 2 more
things .
1) I want to disconnect connected manager into Asterisk. Is it possible ?
2) I want to maintain this socket connection until we disconnect it from web
page.
On Thu, Aug 25, 2011 at 1:57 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
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
Im 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
-- _
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
-
Thanks and regards
[asterisk-users] Information required about ADM
Hello All, I'm a newbie and just started working on asterisk.I have recently installed ADM and I want to know about ADM (Asterisk Desktop Manager) like its benchmarks ,issues or bugs, compatibility with which asterisk version e.t.c. and then any Good web-based CRM recommendations. thanx in anticipation. Regards Gohar _ More than messages–check out the rest of the Windows Live™. http://www.microsoft.com/windows/windowslive/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
