[asterisk-users] AMI is not loaded
Dear All, I have the following entry in the /etc/asterisk/manager.conf file, [general] enabled = yes webenabled = yes port = 5038 bindaddr = 0.0.0.0 [admin] secret = admin read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config When I did 'reload manager' in CLI I have received following error, [Nov 7 13:13:26] ERROR[14031]: config.c:1083 process_text_line: The file 'manager.d/*.conf' was listed as a #include but it does not exist. [Nov 7 13:13:26] NOTICE[14031]: manager.c:4081 __init_manager: Unable to open AMI configuration manager.conf. Asterisk management interface (AMI) disabled. What is the problem? How can I over come this problem? Please any one help me. Thanks, Velusamy. ___ -- 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
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
Thanks Dave, I added a new SSD harddrive instead of a normal SATA harddrive as well as included my ip in the hosts file also I have included 'skip-name-resolv' as you mentioned to not to resolv and tested for around 250 concurrent calls, connection was going through fine...next week I would be testing it for some more additional calls..Thanks for all your replies. Best Regards Sandesh On Fri, Nov 6, 2009 at 8:01 AM, David Gibbons wrote: > I've seen asterisk really bog in internal networks if the mysql server has > name resolution turned on (dns issues of course). The query will be blocked > until the name resolution times out. > > Try adding this line the [mysqld] section of your my.cnf: > > [mysqld] > skip-name-resolv > > That sent a server from ~10 seconds/query down to milliseconds. > > --Dave > > > ___ > -- 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 > ___ -- 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
Re: [asterisk-users] SendJabber question sending Links
On 6/11/09 10:21 PM, Stefan Schmidt wrote: > thanks for your answer, i will try to say it in an easy way ;) > > i send now a jabber message which looks like this: > > > Customer Nr [1234] Person ABC from Company XYZ > > CRM: https://crm.x.y/getcustomer?customer=1234 > Ticket: https://rt.x.y/getticketsfromcustomer?customer=1234 > > > but what i want to have should look like this: > > > Customer Nr [1234] Person ABC from Company XYZ > > CRM [URL] > Ticket: [URL] > > > I´ve tried to send it in html style withtext and so > on, but i didnt get it working. The Problem is that the links i send has > around 400 Chars each which make the message long and hard to read. > > i hope its now clear what i want. Heh sounds like you need tinyurl.com. Or maybe just make a page a.php?1234 which loads getcustomer.php?customer=1234 Or with an Apache rewrite. It seems that what you're wanting is more on the jabber client side - you're wanting one that can receive messages and display them as pure HTML. There may be one - I don't think Adium (the client I use) does it, but if you had a look at a few different clients, maybe one will. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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
Re: [asterisk-users] which asterisk, libpri, dahdi tar file to compile
On 6 Nov 2009, at 09:11, > asterisk-1.4.26.3.tar.gz > asterisk-1.6.1.9.tar.gz > I think you need to do some more reading before you do any installing.. Steve ___ -- 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
Re: [asterisk-users] Asterisk 1.4 DISA is jumping after one digit in the DISA context
Am Friday 06 November 2009 00:17:36 schrieb Marc Lindner: > Dear list, > > I have problems with DISA on an specific server with Asterisk > 1.4.26.2. > > After starting DISA I can only press one key and DISA is jumping > direct into the context without waiting for further digits. The reason and solution is: exten => _X!,n,DISA(no-password|calls_disa) [calls_disa] exten => _X.,1,NoOp() exten => _X.,n,HangUp() if context [calls_disa] like this exten => _X!,1,NoOp() exten => _X!,n,HangUp() then DISA function is broken, after entering one digit, dialplan jump to calls_disa. I did not expected this... ___ -- 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
Re: [asterisk-users] SendJabber question sending Links
Matt Riddell schrieb: > On 5/11/09 9:14 PM, Stefan Schmidt wrote: >> Hello, >> >> i use sendjabber notifications when a call is answered to send the >> answering user information about the caller also with links to our CRM >> or ticket system. >> >> My problem is that i dont know how i can make a link like CRM and not >> have to use http://crm.x.y/fubar?user=1234. >> >> i´ve allready googled for this question, but i´ve only found how to xml >> format an url, but not how i can send it with sendjabber application. >> >> Does anybody have an idea how i can do this? > > It might pay to rephrase your question. > > You're trying to send a link, and what's going wrong? > thanks for your answer, i will try to say it in an easy way ;) i send now a jabber message which looks like this: Customer Nr [1234] Person ABC from Company XYZ CRM: https://crm.x.y/getcustomer?customer=1234 Ticket: https://rt.x.y/getticketsfromcustomer?customer=1234 but what i want to have should look like this: Customer Nr [1234] Person ABC from Company XYZ CRM [URL] Ticket: [URL] I´ve tried to send it in html style with text and so on, but i didnt get it working. The Problem is that the links i send has around 400 Chars each which make the message long and hard to read. i hope its now clear what i want. best regards steve ___ -- 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
[asterisk-users] which asterisk,libpri,dahdi tar file to compile
hi all, i have started asterisk from scratch level. i have downloaded asterisk,libpri,dahdi from asterisk.org [1], i am getting confused as there are more than one version of asterisk and dahdi. i had downloaded those tar in /usr/src. [ser...@localhost ~]$ cd/usr/src [ser...@localhost src]$ ll asterisk-1.4.26.3.tar.gz asterisk-1.6.1.9.tar.gz DAHDI-LINUX-2.2.0.2.TAR.GZ dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz libpri-1.4.10.2.tar.gz i am confused which tar should i compile. need ur guidance. thx a lot Links: -- [1] http://asterisk.org/ ___ -- 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
Re: [asterisk-users] OT - mISDN and B410P questions
Have you tried Digital calls over the B410P, using DAHDI and B410P as bri_net_ptmp? See https://issues.asterisk.org/view.php?id=16151 Useful when a business is E1/T1 connected, but need a BRI connection to a remote warehouse, without needing a dedicated BRI line. Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Friday, 6 November 2009 7:20 p.m. To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] OT - mISDN and B410P questions On 25/10/09 11:52 AM, Paul Hales wrote: > > I have used both misdn and dahdi_bri over the last year, and would > happy take dahdi if for no other reason that it's much easier to install. > > A patch is available to allow dahdi_bri to work with Asterisk 1.4, and > I have used that successfully. Which brings me to another question - what does Digium recommend people use on a 1.4 system with their b410p card these days? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 ___ -- 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
Re: [asterisk-users] Prevent cell phone voice mail capturing call
This comes from the book Asterisk 1.4 Professional Guide :
A practical queue
Reading the above, you may well feel that queues are pretty constrained
in their
application, however, consider this scenario—a busy executive wants any
call hitting
his desk phone to ring his desk phone, and at the same time call his
mobile/cell. You
might think it's easy ... we'll simply ring both the numbers:
Dial(SIP/200&IAX2/${mytrunk}/0774901)
There is no problem if the mobile is switched on and has a signal, but
if it's switched
off, guess what? It's going to go to voicemail straight away and perhaps
ring the desk
phone once or twice.
Using queues to cascade calls
In order to resolve the above situation, what we need to do is ring the
desk phone
and delay the call to the mobile so as to give the executive a chance to
answer the
phone on his desk (if he's there).
You could of course do this:
Exten => s,1,Dial(SIP/200,10)
Exten => s,2,Dial(SIP/200&IAX2/${mytrunk}/0774901)
The problem with the above is that you'll get a break in the ringing, so
that if the
desk phone is picked up (just as it's going to the second priority), it
will result in
a dead call.
A neater solution is to use the queue application to kickoff multiple
calls for
you with delays where required. Look at the following example of queues
in
extensions.conf:
; Call Sales
exten => 1,1,NoOp(calling sales)
exten => 1,2,Queue(salesQ)
[Queues.conf]
[salesQ]
joinempty = yes
member => Local/*...@call_nik_mobile
member => Local/1...@call_sales/n
member => Local/1...@call_200/n
[..extensions.conf]
[call_sales]
exten => 1,1,Dial(${SALES},30,ortT)
exten => 1,2,VoiceMail(2...@default,su)
[call_200]
exten => 1,1,wait(5)
exten => 1,2,Dial(SIP/200&SIP/201,30,ortT)
[call_nik_mobile]
exten => *35,1,wait(10)
exten => *35,2,Dial(SIP/${mytrunk}/0774960)
exten => *35,3,Hangup()
In this example, an inbound call is routed to [salesQ].
Within [Queues.conf], we've defined three static members to call. The
queue
application executes all three of them at the same time. In essence, now
we have
three independent threads running for a single inbound call.
The net result is that the phones defined in the ${Sales} ring group are
called, and
ten seconds later, the extension 200 starts to ring. In the meantime,
the sales phones
continue to ring uninterrupted. Finally, the mobile starts to ring.
The above shows how you can stagger calls to devices without
interrupting the
ring process.
Jonas.
On Thu, 2009-11-05 at 20:25 -0600, Darrick Hartman wrote:
> Russell Horn wrote:
> > Hi,
> >
> > I've a DID number that gets passed to three internal phones and a cell
> > phone via my outbound IAX trunk. If the cell phone is off or out of
> > coverage, its voice mail captures the call.
> >
> > What's the best way to avoid this? Is there a recommended way to force
> > the cell phone user to press 1 before the call is passed there ala
> > google voice? Or is there another way to detect the presence of the
> > answering machine rather than a human?
> >
> > Thanks,
> >
> > Russell.
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Re: [asterisk-users] odbc to ms-sql server
If you are want CDR's to go to MS-SQL try cdr_tds. Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand Sent: 06 November 2009 07:04 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] odbc to ms-sql server Gotcha! Missed libtool! :) -Original Message- From: Neeraj Chand Sent: Friday, 6 November 2009 6:43 PM To: 'asterisk-users@lists.digium.com' Subject: RE: odbc to ms-sql server Hi all, I'm trying to set up an odbc connection to a ms-sql server from an asterisk 1.6.1 install My problem is that I cannot get asterisk to build func_odbc & res_odbc.so I installed yum -y install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel And then went on to reconfigure / recompile asterisk after a ./configure --with-odbc=/usr/lib/ I get ### checking for mandatory modules: UNIXODBC... ok configure: creating ./config.status And then when I go to make menuselect; [XXX]Res_odbc [XXX] func_odbc [XXX] cdr_odbc Can anyone help out with what I am missing? [I've gotten to a stage where tsql and isql connections to my sql db work, however, getting odbc right is making me pull my hair out a bit] ___ -- 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 ___ -- 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
[asterisk-users] [VUC] Friday Nov 6 @ 12 Noon EST: Village Telco
Hello from http://VUC.me or voipusersconference.org This week on the VoIP Users Conference we welcome the Village Telco Project [http://www.villagetelco.org/ ] self-described as "an easy-to-use, scalable, standards-based, wireless, local, do-it-yourself, telephone company toolkit". This is an inspiring project with the stated ambition: "... to render local telephony in developing countries to be so cheap as to be virtually free. Thanks to advances in Open Source telephony software and the dramatic decrease in the cost of wireless broadband technology, we think this is entirely possible." They are developing open hardware as well: "In a nutshell, the Village Telco needs an affordable device to connect customers to the meshed WiFi network. The Mesh Potato will dramatically reduce the cost of a Village Telco startup" Please join us this week to learn more about this original and worthy project. We have a very wide range of ways you can be a part of our community, thanks to the efforts of members of the same community: Your local time of next conference: http://VUC.me/next Audio live communication channels: SIP g722 wideband 200...@login.zipdx.com SIP g711 7463#2262...@proxy.ideasip.com Skype Call vuc.me or if your bandwidth is limited, low def using skype:ld.vuc.me POTS +1 567 252 2286 Text channels: IRC #voip-users-conference Google Wave search for VUC or try http://VUC.me/nextwave ___ -- 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
