[asterisk-users] Asterisk, 2-way radio systems, app_rpt and chan_rtpdir
Hi people, i'm writting because i would like to know is someone of you have field experience on implementing 2-way radio systems with Asterisk: * app_rpt: TIARA Technology [a.k.a. Asterisk/app_rpt project] is the integration of 2-way radio systems and reasonable telephony * chan_rtpdir: Asterisk channel driver that emulates a radio transmitter and receiver and sends the audio to rtpDir using UDP over IP digitally Please write me/the forum and tell me/us about your experiences with radio systems and asterisk. Thanks in advance. Regards, Jonathan GF ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] x100p wcfxo hangup on outgoing calss
Hi Miguel, i'm in Spain like you. For a normal operational system inmediate should be set to no. Busycount and Busydetect can be improved performance with busypattern. The pattern should be shown in the CLI, just take a look. In Spain we use Kewlstart. If you card allows it try use Julian J. Menendez patch for AnswerOnPolarity... If your provider allows it also, you won't haver further issues, just tune the echo with fxotune. Disable echotraining. Not needed. Disable fax detection. Will work, let's see ;) Regards, Jonathan GF On Jan 4, 2008 11:18 PM, Miguel A Felipe Rodríguez [EMAIL PROTECTED] wrote: I have changed the signalling of the x100p to a fxsls, now i can make outgoing calls, but now I have another problem, cant detect hangup. I post the zapara.conf and the zaptel.conf so if any has idea of waht to change y have tested changing busydetect, busypattern, callprogress, etc.. but no whet results. zaptel.conf fxsls=1 loadzone= es defaultzone = es zapata.conf [channels] language=es context=incoming switchtype=national signalling=fxs_ls usecallerid=yes cidsignalling=dtmf cidstart=polarity hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes rxgain=2.0 txgain=1.0 group=1 callgroup=1 pickupgroup=1 immediate=yes callerid=asreceived busydetect=yes busycount=6 callprogress=no faxdetect=incoming channel = 1 ___ --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] Two lines for outgoing calls
Dominik,
apart from the good responses, please get rid of the 't' in the options of
dial or you will be allowing the called party to transfer the call while you
are paying.
Regards,
Jonathan GF
On Dec 26, 2007 3:32 PM, Dominik Zalewski [EMAIL PROTECTED] wrote:
Dear All,
I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel
2.6.18.
I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm
using below context for dialing out.
[outbound-local]
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9ZXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _90900X,1,Dial(Zap/g1/${EXTEN:1},30m,tTr)
When Zap/1 is busy and I try to call, it will use Zap/2 which is fine
but there is something wrong cause I hear one ring and then a weird
sound like a noise or something and then hangup. I have to reload zaptel
modules and then everything works fine for a while.
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/200-08221590,
Zap/g1/150|30|tTr) in new stack
-- Called g1/150
-- Zap/2-1 answered SIP/200-08221590
-- Hungup 'Zap/2-1'
I even thought that second fxo module is broken so I changed it. No
results.
Any ideas?
Thank you in advance,
Dominik
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Re: [asterisk-users] How can i send my sip channel 3 to mailbox 2? Please Help!
Hey Robert,
you can't imagine how much i appreciate your post, which is most a
tutorial than a post :)
Really, many thanks for your thoughts. Take for sure i will try to
implement the options you showed me here in asap.
Thank you again!
Best regards,
Jonathan GF
On 9/4/07, Robert Lister [EMAIL PROTECTED] wrote:
On Sun, Sep 02, 2007 at 04:03:51PM +0200, Jonathan GF wrote:
Hi folks,
i'm trying to configure my extensions.conf as small as posible and for
that reason i'm using macros. The problem is that maybe I have a
misunderstood the concept for the directive mailbox in sip.conf.
What mailbox= seems to do in sip.conf is set the message waiting indicator
(MWI) light on or off when there are messages waiting in a particular
mailbox for that extension using a SIP message to the phone to update it.
It does not control anything else such as who can access a particular
mailbox etc. just which extensions get notifications of voicemail. It is not
in voicemail.conf
I suppose because asterisk can have different channel types other than SIP,
it needs configuring for the different notification methods depending on
devices. (i.e, voicemail app doesn't want to be getting involved in how to
set and unset MWI for all sorts of different channel types.)
What i'm trying is to have ONLY 2 voicemail boxes and depending which
extensions i'm dialing send the caller to one or the other, but not
send based on the called id/name, but to that mailbox i want (mailbox
1 or mailbox 2, just this).
The error i'm getting is:
WARNING[2102]: app_voicemail.c:2461 leave_voicemail: No entry in
voicemail config file for '3'
The error is correct... i don't have a voicemail box named/numbered
3 but this is the behavior i want to control. How can i send my sip
channel 3 to mailbox 2?
Essentially, you need to pass the mailbox you want to access to the
voicemail app, which is the thing in ARG1 in your macro.
So, Voicemail(u2) would play the unavailable message for mailbox 2 instead
of what you are currently passing it, which appears to be ${EXTEN}, the
dialled extension.
What you can do is check to see if a voicemail mailbox exists for a
particular extension before you try it, and if no mailbox exists (i.e, you
have not configured it in voicemail.conf) then you can do something else.)
Something like this will check to see if a mailbox exists before trying it,
if not then default to mailbox 2:
exten = s,1,MailboxExists(${ARG1},j)
exten = s,2,Voicemail(u2)
exten = s,3,Hangup
exten = s,102,Voicemail(u${ARG1})
exten = s,103,Hangup
Note, you can also check the variable ${VMBOXEXISTSSTATUS} for one of
SUCCESS or FAILED if you don't like the old style priority jumping,
which can get a bit awkward if you have to renumber things, this is the
'newer' way to do it, something like:-
exten = s,1,MailboxExists(${ARG1},j)
exten = s,2,Goto(s-${VMBOXEXISTSSTATUS},1}
exten = s-FAILED,Voicemail(u2)
exten = s-FAILED,Hangup
exten = s-SUCCESS,Voicemail(u${ARG1})
exten = s-SUCCESS,Hangup
Of course, how you work out when somebody accesses your voicemail to listen
to messages depends on how you are authenticating them into voicemail in the
first place. You might just prompt for the mailbox number and/or PIN, or you
can drop them straight into the right mailbox using a similar technique.
If it gets more exotic than your two mailboxes, then you could use astdb
entries to work out which mailbox is associated with a particular extension,
which is more elaborate but might be worth doing for ease of configuration.
(In that you are not hardcoding stuff into extensions.conf for every
extension)
astdb is asterisk's builtin database, which is really handy for this kind of
thing (Unless you have millions of mailboxes which is an entirely different
database proposition!)
$ asterisk -r
asterisk*CLI database put 3 mailbox 2
asterisk*CLI database show 3
/3/mailbox : 2
(That is to say, for the extension 3, we want mailbox 2)
Then, to see that db variable in where you need it in the dialplan,
would look like this:-
${DB(${EXTEN}/mailbox)} (where ${EXTEN} is 3, this would return 2)
or ${DB(${ARG1}/mailbox)} in the case of your macro.
This will look in the astdb for that mailbox variable you set up and
use that instead of hardcoding it into the dialplan.
Suppose your voicemail access extension is 444 and you want a passwordless
login from the extension based on what you have set in the astdb for that
extension, based on caller ID of the incoming extension:-
; passwordless login
exten = 444,1,VoiceMailMain(${DB(${CALLERID(num)}/mailbox)}|s)
exten = 444,n,Hangup
(Yes, I know, it's a bit fugly bracket hell, but it's worth it!)
You could combine this of course with MailboxExists to drop them into some
default mailbox, or prompt for a mailbox number, or if there is a mailbox
for that extension and no translation is required
Re: [asterisk-users] Rechazo de llamada en triangulacion de asterisk.
Guillermo, el username deberia de ser igual al nombre del canal, es decir, si [pbx1] entonces username = pbx1 Saludos, Jonathan GF On 9/3/07, Guillermo Rodriguez [EMAIL PROTECTED] wrote: Hola Alex, He puesto el username y aun asi me da el fallo de autentificacion :-( Creo que lo mas facil sería cambiar el name de los telefonos.. aunque seria lo mas complicado ya que tendria que cambiar el name de todos mis clientes... y realmente no sera precisamente una solucion viable. El Jueves, 30 de Agosto de 2007 17:52, Alex Balashov escribió: Guillermo, Me parece que la cosa aqui es que el nombre del usuario debe ser el mismo en el URI del fuente que en el el proceso de autentificacion. Traiga poner username= en la configuracion asi: On Thu, 30 Aug 2007, Guillermo Rodriguez wrote: [pbx1] name=test1 callerid=200 host=dynamic nat = yes type friend secret= test1 username=... Y diganos lo que pasa. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 ___ --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] Dificult macro, please advise
Hi,
BRIEF RESUME:
Is there any other way to obtain the same result but being easier to
configure?? Thanks!
EXTENDED RESUME:
i've configured a, rather difficult, macro that even for me without
being documented is difficult. I ask for the help of the experts to
know if the functionality it apports can be achieved better in another
way.
What i'm trying is to enable call a channel (e.g. SIP/3) and being
able to leave a message only on box 2 while any other call to any
other user (e.g.SIP/1 or SIP/2) and leave a message only on mailbox 1.
This is inteded for a SOHO environment.
I have defined only two mailboxes, the 1'st for personal ussage and
the 2'nd for profesional usage.
The macro that now allows me to do that is the following:
EXT_CALLER=
EXT_STUDIO=SIP/3
exten = _X,1,Macro(diallocal|SIP/${EXTEN}|20,Tr)
[macro-diallocal]
exten = s,1,Dial(${ARG1}|${ARG2}|${ARG3})
exten = s,2,Set(EXT_CALLED=${ARG1})
exten =
s,3,GotoIf($[${EXT_CALLED}=${EXT_STUDIO}]?s-STUDIO-${DIALSTATUS},1:s-REST-${DIALSTATUS},1)
exten = s-STUDIO-NOANSWER,1,Voicemail(u2)
exten = s-STUDIO-NOANSWER,2,Hangup()
exten = s-STUDIO-BUSY,1,Voicemail(b2)
exten = s-STUDIO-BUSY,2,Hangup()
exten = _s-STUDIO-.,1,Hangup()
exten = s-REST-NOANSWER,1,Voicemail(u1)
exten = s-REST-NOANSWER,2,Hangup()
exten = s-REST-BUSY,1,Voicemail(b1)
exten = s-REST-BUSY,2,Hangup()
exten = _s-REST-.,1,Hangup()
Is there any other way to obtain the same result but being easier to configure??
Thanks in advance.
Best regards,
Jonathan GF
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Re: [asterisk-users] Dificult macro, please advise
Atis, thanks for the quick post. I tried, probably wrong, to make a simple macro for all local switching, but i realized it became hard to mantain and can divert to errors in the future. I think i will go towards your proposal. Thanks for the input :) Jonathan GF On 9/3/07, Atis [EMAIL PROTECTED] wrote: On 9/3/07, Jonathan GF [EMAIL PROTECTED] wrote: Hi, BRIEF RESUME: Is there any other way to obtain the same result but being easier to configure?? Thanks! Why don't you use asterisk extension masks? ___ --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] How can i send my sip channel 3 to mailbox 2? Please Help!
Hi folks,
i'm trying to configure my extensions.conf as small as posible and for
that reason i'm using macros. The problem is that maybe I have a
misunderstood the concept for the directive mailbox in sip.conf.
Under my knowledge configuring the mailbox directive to the mailbox I
want would be enought to leave an retreive messages in that voicemail
box. Of course it seems to be that i was wrong :/
What i'm trying is to have ONLY 2 voicemail boxes and depending which
extensions i'm dialing send the caller to one or the other, but not
send based on the called id/name, but to that mailbox i want (mailbox
1 or mailbox 2, just this).
The error i'm getting is:
WARNING[2102]: app_voicemail.c:2461 leave_voicemail: No entry in
voicemail config file for '3'
The error is correct... i don't have a voicemail box named/numbered
3 but this is the behavior i want to control. How can i send my sip
channel 3 to mailbox 2?
I'm a bit stuck and would appreciate so much your help.
The block that is causing me headache is that:
SIP.CONF
[3]
context = internal
type= friend
username= 3
secret = pwd3
callerid= Studio 3
host= dynamic
nat = no
mailbox = 2
qualify = yes
canreinvite = no
callgroup = 2
pickupgroup = 2,1
dtmfmode= rfc2833
-- EXTENSIONS.CONF -
[internal]
exten = _x,1,Macro(diallocal|${EXTEN}|SIP/${EXTEN}|15)
[macro-diallocal]
exten = s,1,Dial(${ARG2}|${ARG3}|Tr)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Hangup()
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Hangup
exten = _s-.,1,Hangup
- VOICEMAIL.CONF --
[zonemessages]
europe=Europe/Madrid|'vm-received' Q 'digits/at' R
[default]
1 = 1,Main Phone,,,saycid=yes|delete=no|tz=europe
2 = 2,The Studio,,,saycid=yes|delete=no|tz=europe
All configuration (sip, extensions, voicemail, etc...) is available @
http://www.surestorm.com/asterisk/ for those that want to help.
Thanks in advance.
Best regards,
Jonathan GF
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Re: [asterisk-users] Nokia cell connected to Asterisk
Thank you all for your post, i've found them quite interesting and will give work for some time :) Thanks again. Cheers, Jonathan GF On 8/20/07, Eric Chamberlain [EMAIL PROTECTED] wrote: Using the phone itself as a GSM-SIP gateway is not possible with the native VoIP application, but it looks like it should be possible with a custom application for the phone. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Remco Barendse Sent: Monday, August 20, 2007 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nokia cell connected to Asterisk Has anyone ever tried using a Nokia phone with SIP client as channel for Asterisk? I mean i would like to receive calls to the mobile on asterisk and use the Nokia phone to place calls to cell destinations. I have enough Nokia E60's to do that and it would circumvent the need for chan_bluetooth or something similar!! :) On Mon, 20 Aug 2007, Steve Totaro wrote: Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Thanks, Steve Jonathan GF wrote: Thanks Steve and Mitcheloc, in fact i was think in something more obsolet like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, *mitcheloc* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com http://www.snapanumber.com ___ --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] Asterisk in Soekris 5501: Is Astlinux the only able solution?
Hello, i would like the forum to help or advice me if my feeling is correct or not. Is Astlinux the only distribution able to run on Soekris 5501 hardware or other can run also (trixbox, freepbx, o maybe a manual installation of asterisk). My question is easy: i'd need to install it on that hardware for a very small office and the further administrator do not understand so much about Asterisk, although they can handle unix/linux boxes. Any help would be really appreciated. Thank in advance for you help. Regards, Jonathan GF ___ --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] Nokia cell connected to Asterisk
Thanks Steve and Mitcheloc, in fact i was think in something more obsolet like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, mitcheloc [EMAIL PROTECTED] wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --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] Nokia cell connected to Asterisk
Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --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
