Re: [asterisk-users] How to resell my trunk/provider to others?
Have you considered putting an advertisement in the newspaper? PaulH On 05/11/09 06:43, Carlos Cuervo wrote: Hello, I've been tasked to look for ways to resell to others the service that one of a trunk provides.. In other words, i want to configure my current Asterisk (Ver. 1.4.26.1) with Freepbx 2.6.0 so i can act as a trunk to others.. I would provide an IP to them from one of my servers and they will use that IP to connect to me and i will connect them to my trunk/provider. If possible, please provide some guidance as to where to start or a link since i searched in google with no valuable results.. Maybe am looking incorrectly. Regards, Carlos ___ -- 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] How to dial multiple extensions at once likeinaring group and put them in conference?
On 29/10/09 22:40, Matt Riddell wrote: :D I should hope not!! If everyone was as smart as me, how would I take over the world? With violence, just like everyone else! PaulH ___ -- 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
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. PaulH On 25/10/09 03:26, Olivier wrote: Hello, I'm evaluating to possibility to use chan_misdn as a short term workaround, in case latest Dahdi is not stable enough for what we are planning to do (we wish to use Junghanns and Digium BRI hardware with Asterisk 1.6) . I've read www.mISDN.org http://www.mISDN.org but still have a couple of questions : 1. Is correct that in a 2.6.27 (and up) enabled kernel, the embedded mISDN version is 2.X ? 2. Is it correct that Asterisk MUST use chan-lcr to access this mISDN software or is it still possible to install mISDN 1.X to be able to use chan_misdn ? 3. Am I correctly understanding README in Dahdi-linux when I think you can switch Digium B410P support from dahdi to chan_misdn, just editing /etc/dahdi/modules file ? 4. Would you trust chan_misdn as a valuable short term solution for ISDN BRI with Asterisk 1.6 ? Regards ___ -- 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] interfacing asterisk with a legacy PBX
On 24/10/09 00:59, Lyle Giese wrote: PATRICK KANGETHE wrote: I want to interface asterisk with a legacy pbx that has around 23 extensions through my 8 fxs card, how do i work around this? Hint: I have already terminated 8 extensions from the legacy PBX, i was thinking whether i can peer the extensions from the PBX i.e like 5 extensions be peered to one extension connecting to the fxs? How can i do this? Thanks in advance, Are you planning to get rid of the legacy PBX completely? Or is Asterisk going to be a second PBX? I am going to assume you are replacing the legacy PBX. You can setup analog extensions so that you have multiple phones on each FXS channel. But they will be like a party line. If you put 6 phones on one FXS, all 6 ring at the same time, only one person can use that extension at a time. However you can add SIP phones to Asterisk and each can have their own extension instead. It just requires cat 5 cable back to a switch for each phone. Lyle Giese LCR Computer Services, Inc. Just to add my 5 cents - connecting too many phones to an FXS port can cause problems. The term is REN - ring equivalent number, and it's used to describe the maximum phones to attach to an FSX port (from memory) PaulH ___ -- 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
Jeff LaCoursiere wrote: Steve Edwards asterisk@sedwards.com wrote: Since I'm an old-school C programmer, I use emacs as my editor. I fire up gdb (the GNU C (amongst other languages) debugger) in a window, give it a command like b main; r dummy-input-for-block-ani and I can step through my program line by line, examining and changing variables at will. Bah. If you were really old school you would use vi. [ducking!] :) j Old school? I tried to use 'ed' the other day, and failed. PaulH ___ -- 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] all our circuits are busy now
The list will need to see your dialplan or a CLI dump to help you with this. PaulH B.Masoud @ SH wrote: I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message “ all our circuits are busy now” then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? thanks ___ -- 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] how to limit the calls leaving a queue?
I have used the group function to limit the calls entering a queue for a similar reason to yourself. PaulH Niccolò Belli wrote: Hi, I explain what I want to do.. All the operators share their phones. The number of the operator isn't constant, so it's possible that two operators share all the phones. They need to move all around, so they pick up the first phone they find. If there are only few operator is very annoying for them to ear the other phones ringing while they are at the phone! So I'dd like to limit the maximum number of simultaneous calls leaving the queue, but how to do it? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] tdm outgoing
Is inbound working? Can you see action on the CLI when you send a call to the lines attached to the card? PaulH B.Masoud @ SH wrote: Hi I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls to that trunk, I am getting all circuits are busy now, do I have to do something specific?? I am using elastix. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] kill sip user
Death to all sip users! Paulh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Firefox Plugin for Sip Click2Call
http://www.noojee.com.au/Page/NoojeeClick Built for firefox. PaulH Stefan Schmidt wrote: Hello, iam searching for an Firefox plugin which can make an sip Invite and Redirect after 200 OK, so i dont have to use a softphone, just to initialise a call by clicking on a number i've found some plugins which only works with a softphone installed on the system but nothing which works good with asterisk. my other problem is that we use firefox 3.5 mostly on mac so maybe there are softphones which can do this what i search but not for this version. Have anybody an idea where i can find such a plugin? Best regards steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Help sending call to local server
This needs work, but it's about right for both of the problems -
probably a cut command to filter out the actual extensions being dialled.
PaulH
exten = _x,1,Set(state=${SIPPEER(${EXTEN}:status)})
exten = _x,n,GotoIf($[${state:0:2}=OK]?online:offline)
exten - _x.,n(online),Dial(SIP/${EXTEN})
exten
=_x.,n(offline),Dial(${Dial_technology}/${extension_to_ca...@${server_ip},30,r)
DHAVAL INDRODIYA wrote:
hey paulh,
i think this would not help
because he wants such a dial command which forwards a call to local
server if server_ip is of same server
i have same kind of problem but still dont found proper solution
in,fact i need dialing on IP base in which dialing by using IP address
will send call to remote machine or same machine
regards
Dhaval
On Fri, Sep 18, 2009 at 5:59 PM, Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au wrote:
I have used the SIPPEER function to find if a phone is local and
available before.
PaulH
Asterisk User wrote:
Hi,
I have a generalized syntax for dial application in my dialplan
where
I send calls to particular server.
Here is my dial sysntax...
exten =
_x.,1,Dial(${Dial_technology}/${extension_to_ca...@${server_ip},30,r)
I can send a call to remote server using register statement in
sip.conf or iax.conf and it works as calls get landed in particular
context of remote server.
Would you please suggest me changes to be made in .conf file(s) if I
want the calls to be landed in context of local server if
Server_ip is
the IP of a server running asterisk?
Thanking you
--ASTERISK USER
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Re: [asterisk-users] Help sending call to local server
I have used the SIPPEER function to find if a phone is local and
available before.
PaulH
Asterisk User wrote:
Hi,
I have a generalized syntax for dial application in my dialplan where
I send calls to particular server.
Here is my dial sysntax...
exten =
_x.,1,Dial(${Dial_technology}/${extension_to_ca...@${server_ip},30,r)
I can send a call to remote server using register statement in
sip.conf or iax.conf and it works as calls get landed in particular
context of remote server.
Would you please suggest me changes to be made in .conf file(s) if I
want the calls to be landed in context of local server if Server_ip is
the IP of a server running asterisk?
Thanking you
--ASTERISK USER
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Re: [asterisk-users] Simple Time of Day Branching problem
It's easier to work with the closed hours then - use a goto just for Sunday/Monday PaulH James Hankins wrote: Greetings folks, new to this, trying to get the syntax correct for a day of week routing. exten = 345,1,Answer() exten = 345,n,GotoIfTime(10:00-17:00|tuethusat|*|*?open,345,1) exten = 345,n,GotoIfTime(10:00-19:00|wedfri|*|*?open,345,1) exten = 345,n,Playback(afterhours) exten = 345,n,Hangup() I'll get an error stating incorrect day of week tuethursat, assuming none What is the correct syntax for this? We have longer hours on Wednesday and Fridays and we're closed Sunday/Monday Just trying to automate the time of day greeting etc. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Odd sip error
Has anyone seen this one before? full:[Sep 14 11:49:01] DEBUG[15771] chan_sip.c: SIP attended transfer: Error: No target channel It coincided with a failed attended transfer... Ideas? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] The identifier parameter in Dial() command
I would strongly suggest you browse: http://www.asteriskdocs.org/ Kind regards, PaulH Songtao Yu wrote: Hi All, I am new to Asterisk. Now I got one question on the identifier parameter of the Dial() command. I saw as below: exten = 20,1,Dia(Zap/3/5551234). Would you please let me know the meaning of 5551234? Thanks, Songtao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Using asterisk as the recording server
I have also seen: PSTN asterisk legacy Which also gives you a migration path PaulH Research wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] queue issue
A situation where staff want a mobile and their SIP handset to share an extension - but to make sure the mobile or SIP handset do not ring if they are speaking on the other one... PaulH Lenz Emilitri wrote: It depends on what you want to do to people who are queued; if you want them to be queued, you create a queue with only one member, and have agents log on and log off as necessary; if you don't want callers to be queued, likely I would not use a queue but woul dial the agent straight. l. PS. this is quite an unusual requirement, what is it for? 2009/9/1 Paul Hales pdha...@optusnet.com.au mailto:pdha...@optusnet.com.au I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] queue issue
Matt Riddell wrote: On 3/09/09 11:34 AM, Paul Hales wrote: Hmmm.any idea how I can use hints to monitor their mobile phones? Unless the call came in via Asterisk, you can't. The calls will - so it should be able (at the very least with the asterisk internal DB - which I don't fully trust due to reboots and the odd weird behaviour) Why not just have the desk phone accept one call (i.e. call/group/whatever limit) and then use app_followme? The issue is that both phones have to ring at the same time.And it's easy enough to stop the mobile from ringing if the SIP phone is in use, but the other way around is the challengeIt's doable, but I want to find the right solution. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] queue issue
They don't want to log in, and they want both to ring if they are free -
this is a very large site, so they need to be contactable at all times.
PaulH
Lenz Emilitri wrote:
I would have them log on with the mobile when they need it, and log
off when they don't. When the mobile is not present you would simply
dial the local extension.
You could have something like:
local/1...@agents
that does something like:
if ( DBSET(has_mobile) ) {
dial( Zap/g0/MYMOBILENUM )
} else {
dial( SIP/123 )
}
and have anothe extension set/reset the has_mobile property in the AstDB.
You could then call Local/1...@gaents directkly or make it a member of
the queue (with known issues on some version of *) :-)
l.
2009/9/2 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
A situation where staff want a mobile and their SIP handset to
share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want
callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au mailto:pdha...@optusnet.com.au
I have a _very_ specific situation where I need queues to
work in
a very
specific manner - I need the queue to only accept one call
at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this
way in the
distant past (pre 1.0)...but I am willing to be mistaken.
Is this even remotely possible?
PaulH
--
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] queue issue
Hmmm.any idea how I can use hints to monitor their mobile phones?
PaulH
Danny Nicholas wrote:
One way to do this would be to use hints and an AGI to control dialing.
Let's say you have extensions 100 and 101 and each staffer also has a cell
(555-1212 and 555-1213). When you dial 100, you want to ring 100 and
555-1212 if both are available and the same with 101 and 555-1213. This
snippet would do it:
- exten = s,1XX,Macro(ring-group,${EXTEN})
- exten = s,1XX,playback(vm-goodbye)
- exten = s,1XX,hangup
- [macro-ring-group]
- exten = s,1,AGI(checkhints.agi,${ARG1})
- exten = s,n,gotoif($[${LINESTAT} = BUSY]?inuse)
- exten = s,n,Dial(SIP/${ARG1}DAHDI/g1/${CELLLINE},60)
- exten = s,n,hangup
- exten = s,n(inuse),playback(line-in-use)
- exten = s,n,hangup
The AGI checks the hint for 100 or 101 and assigns CELLLINE to call the
cell. If either is in use, LINESTAT is set to BUSY, otherwise set to AVAIL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales
Sent: Wednesday, September 02, 2009 2:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue issue
A situation where staff want a mobile and their SIP handset to share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
I have a _very_ specific situation where I need queues to work in
a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant past (pre 1.0)...but I am willing to be mistaken.
Is this even remotely possible?
PaulH
--
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] Inquiry:How to hide Caller Id
I couldn't find any information on this brand of phone on the internet at all. PaulH hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- 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 http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] queue issue
I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] queue issue
Miguel Molina wrote: Paul Hales escribió: I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH Hi, Maybe maxlen = 1? Cheers, Hmmm - almost. Maxlen limits the amounts of calls waiting for the queue, not the amount of callers talking to queue members. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inquiry:Problem with Call Parking
But they do taste similar. PaulH Darrick Hartman wrote: Polycom sip.cfg is not the same as the Asterisk sip.conf file hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inquiry:How to hide Caller Id
Matt Riddell wrote: What is a subs? A submarine. I think. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Sticky Park
Sticky Park sounds like somewhere you go late at night wearing a plastic
raincoat.
PaulH
Mat Murdock wrote:
My company for various reasons has asked that I come up with a way to
have previously parked calls be re-parked in the same parking slot. I
have looked at setting up asterisk so that the receptionist chooses
which slot to place a call, but I think there is an easier way. That is
when I came up with the idea of Sticky Park. Here is how it would
work. A call would come in and the receptionist will park the call as
she normally does. Asterisk will the pick the first open parking slot,
let's say 702 because there is already a call on 701. Lets say that the
call parked on 701 is picked up, freeing 701. So, 701 is free and 702
has our call parked on it. Now the call on 702 rings back to the
receptionist because it has timed out. She asks the person if they
would like to continue hold and will again park the call as she normally
does. Asterisk will then re-park the call back onto 702 because that is
where it came from. The normal behavior of Asterisk would of been to
park it on 701 because it is the first free parking slot. That is why I
call it Sticky Park. So what happens if between the time she picks
up the call and re-parks it someone else parks a call on 702? Then I
think Asterisk should then pick the first available parking slot and
that call becomes stuck to that parking slot if additional re-parks are
necessary.
Here is my dialplan on how I thought I could accomplish this with
dial-plan magic.
Here is the relevant features.conf entries.
[general]
parkext = 799 ;We need to use our own 700 extension so lets get this
out of the way.
parkpos = 702-706
comebacktoorigin = no ;This causes calls that have timed out to
come to the parkedcallstimeout context at s,1.
Ok now onto my Dial Plan.
[from_internal]
include = parkedcalls ; Gotta have this or things don't work.
;I do an attended transfer to 700.
exten = 700,1,Answer()
;Just so I can see if anything has been set
exten = 700,n,NoOp(I want to be parked on: ${PARKINGEXTEN})
;Also so I can see what the state of that parking slot is.
exten = 700,n,NoOp(Device State is:
${DEVICE_STATE(park:${parkingext...@parkedcalls)})
;Check to see if PARKINGEXTEN is set. If not then this must be a new
call being park, let's let asterisk find a spot for it.
exten = 700,n,GotoIf($[${LEN(${PARKINGEXTEN})}=0]?parkcall)
;Ok Looks like this call has been parked before. Let's see if we can
repark it in the same spot again. If it not INUSE then let's park the call.
exten =
700,n,GotoIf($[${DEVICE_STATE(park:${parkingext...@parkedcalls)}=INUSE]?:parkcall)
;Previous slot is not occupied lets clear the PARKINGEXTEN variable so
that when we park the call Asterisk will find the first available slot.
exten = 700,n,Set(PARKINGEXTEN=)
;Lets park the call.
exten = 700,n(parkcall),Park()
exten = 700,n,Hangup()
[parkedcallstimeout]
exten = _SIP011XX,1,Answer()
exten = _SIP011XX,n,NoOp(Call Parked on: ${PARKINGSLOT})
exten = _SIP011XX,n,NoOp(This is who parked us: ${EXTEN})
exten = _SIP011XX,n,Set(PARKINGEXTEN=${PARKINGSLOT})
;This sets the PARKINGEXTEN to the parking slot we were parked in.
exten =
_SIP011XX,n,Dial(SIP/${EXTEN:4:4},${RINGTIMER},${INTERNAL_DIAL_OPTIONS})
;This send the call back to the person who parked it. There are a
couple of global variables I use here. Nothing unusual here.
So what is the problem? Well the problem is that the PARKINGEXTEN
variable gets reset after the dial command in parkedcallstimeout. That
makes it so I cannot find out where that call was originally parked If
I can find out how to get that little bit of information when the call
is re-parked then I think this will work. If anyone has any suggestions
on how to accomplish this I would be grateful.
Thanks,
Mat Murdock
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Re: [asterisk-users] onnecting two asterisk using B410p BRI cards
Use a standard network cable - but you have to activate the 'terminate' jumper on the NT end. - Also, the new BRI stuff in dahdi is much easier to work with than misdn. PaulH voip crazy wrote: Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the cable was pluged. Then I use an isdn crossover wire with this king of schema and the lights get blinking red again. Tx+ 3 --+ +- 3 .X Rx+ 4 --+ +- 4 . Tx- 5 --+ +--5 .X Rx- 6 --+ +--6 In both servers when I do in asterisk CLI misdn shos stacks, the port one on each machine shows Server A: BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Server B: BEGIN STACK_LIST: * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Which kind of cable should I use? Why both in ports L1Link is failed? How could I solve that? Any clue will be welcomed. Thanks in advance. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Player to listen to WAV files using an hardphone
Can I assume that you meant to add that the person's phone would be used to listen to the message? PaulH Olivier wrote: Hi, I've lastly read a Request For Quotation asking for a software option I've never heard about before. It's about a player plugin with which, when using an Outlook-like email client, you can double click on an enclosed WAV file icon to listen to a voicemail instead of using PC speakers (and microphone). Has anyone heard of such player before ? Regards ___ -- 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 -- AstriCon 2009 - October 13 - 15 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] Analog FXO or IAX DIDS for new facility?
In australia, I would usually suggest a mix of E1 and SIP for calls - it doesn't cost any money to receive calls via E1, and redundancy is an old, valuable friend of mine. PaulH Stephen Fierbaugh (PBT) wrote: I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read Asterisk: The Future of Telephony and done a fair amount of googling. I am completely sold on Asterisk, and the new building's phones will be a mix of SIP handsets and softphones. I am confused about one thing: Should we be getting a block of analog circuits from the local telco (probably ATT), connected to the server's FXO cards for in-bound and out-bound POTS calls; or should we get a block of DIDS numbers from one of the plethora of providers available over the Internet, and then have our server connect POTS calls by IAX to the DIDS provider? We are unsure whether we are going to have separate numbers for everyone in the organization, or just 1 US phone number, with everyone in the org having their own extension number. That probably largely depends upon cost. We will have 75 people in the building. We have no data on call patterns or usage (because our legacy system belongs to our current facilities host), but we currently have 4 lines for 35 people and on unusual occasions they all get busy. An additional consideration is that we also have 300 other people scattered literally world-wide, and the next logical future step is to start providing VOIP links for them, as well. Thanks in advance for your advice. Any other suggestions, such as # of lines sizing info or reputable DIDS vendors (if that's the answer) are also appreciated. ___ -- 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 to PBX
Can I assume that your project has stalled? PaulH logan wrote: Thanks Paul. Your help is much appreciated here. I don't really understand this question - Asterisk can make calls over phone lines. And it does it well. Surely, Asterisk does that well, but Asterisk needs to have multiple phone lines for that. I thought that a traditional switchboard made that happen without multiple phone lines. BTW, in Asterisk terminology a phone line means different PSTN connections to the operator, right? Why would you guess this? We had 16 phone lines in the first business I worked in. Yeah, that's fine, but even 16 phone lines don't mean you can have 16 desk phones only or 16 simultaneous calls? Thanks I will take a look at asteriskdocs. Best Regards, Hitesh ___ -- 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] Asterisk to PBX
Some thoughts inline: logan wrote: Hi Paul, Thanks a lot for the response. I'm a novice so pardon me for the stupid questions. I thought that maybe the PSTN lines don't allow more than 1 simultaneous calls on a line, but on GSM it might be possible. I basically want to know how Asterisk can dial out calls from the lines connected to it. Ideally I want to make out as many calls from the lines connected to my Asterisk box. I don't really understand this question - Asterisk can make calls over phone lines. And it does it well. I have a few related questions, again pardon me if I'm a novice. How did PBX in days when didn't have Asterisk worked? We used to have an NEC. If a company wanted to give desk phones to all the employees then it would have a switchboard which would route the calls. Maybe. Or maybe not. Now in this case I'm guessing that the company had only one PSTN line, Why would you guess this? We had 16 phone lines in the first business I worked in. but somehow the switchboard let everyone make calls and receive calls at the same time. Because the calls never used the phone lines. So is it possible to have the switchboard and have it connect to Asterisk who can there by use these lines? I suppose.and I think attaching a vintage jack-style switchboard would be a very fun project. Paul, could you also describe a bit about hook flash? It's a way of putting a call on hold and taking one off hold - much like your descreption of how calls work on a mobile phone. You should had a read of: http://www.asteriskdocs.org/ later, PaulH ___ -- 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 to PBX
logan wrote: Thanks Paul. Your help is much appreciated here. No problem - been working on telephone systems for about 12 years now - which doesn't even make me an old hand... Surely, Asterisk does that well, but Asterisk needs to have multiple phone lines for that. I thought that a traditional switchboard made that happen without multiple phone lines. Not really - but there's something you are missing in your understanding and it will come to you soon enoughjust keep reading and asking questions. Of course, Asterisk can place many calls down a network connection/adsl/E1/DS3/etc. BTW, in Asterisk terminology a phone line means different PSTN connections to the operator, right? Once again, I don't really understand this question. Why would you guess this? We had 16 phone lines in the first business I worked in. Yeah, that's fine, but even 16 phone lines don't mean you can have 16 desk phones only or 16 simultaneous calls? We had about 40 phones. We could make 16 inbound/outbound calls, and as many internal calls as we wanted to... Thanks I will take a look at asteriskdocs. Reading is a great way to learn things. PaulH ___ -- 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 to PBX
Sadly, at the end of the day the answers will probably be no, no, no and no. PaulH logan wrote: Hi, I'm an absolute newbie and wanted to know the following. I want to have a setup where I have a PSTN line connected to my Asterisk box and want to know if it is possible to make more than one simultaneous outbound call through that VoIP gateway? Can Asterisk do this magic of concurrent calls on one PSTN line?? If I put it in other words then can I receive more than one simultaneous call on a PSTN number through Asterisk (the dialplan would forward those calls to different extensions) and the phone line still be able to receive more calls? Do I need some special hardware for the above or a simple SIPURA3000 would be good enough? Please pardon me if this is not the correct list for this question. Thanks. Best Regards, Hitesh ___ -- 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] Asterisk to PBX
logan wrote: Hi Trevor, Thanks for the response. I was thinking if a GSM to VoIP gateway can do the job of multiple outgoing calls. I could be wrong but seems like cellphones do allow you to make multiple calls at a time (only one is active or one active conference). If we assume that I have a layer on my Nokia phone which allows me to have more than one active call then when my Asterisk system tries to make a call through the GSM-VoIP gateway then it would never get a busy and the layer on phone would do the job of facilitating calls. Is this a possibility that anyone has tried? Are you talking about using hook flash to change between active calls? Or the more interesting facilities available on the 3G (and beyond) networks? regards, PaulH ___ -- 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] How to ask questions the smart way
Always a great readthanks. PaulH Alex Balashov wrote: Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's classic How to Ask Questions the Smart Way to the OpenSIPS-users mailing list[1], I'm going to repost it here: http://www.catb.org/~esr/faqs/smart-questions.html As Adrian said, This a good read for those who show up on mailing lists without any guidance about how to ask the right questions and then complain that nobody answers their questions as they want. I think there's never a wrong time and a wrong place on a public high-volume mailing list for all the participants to take a moment and meditate on this issue a little bit. [1] http://www.openser.org/pipermail/users/2009-July/006873.html ___ -- 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] is Asterisk reliable for a call center application??
Yes, but not as your first Asterisk implementation. PaulH gergis.rasmy wrote: i am asked to implement a call center of 50 seats for my company , and i was wondering if Asterisk can fit this as a relaibale and low price system is it mature enough for this task?? best regards Gers ___ -- 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] Bug or Not?
'One touch park' was designed to work around this issue. PaulH Danny Nicholas wrote: Hi gang, When I try to park a call using blind-transfer (#1), the caller hears the lot instead of the transferring party. Attended transfer and blind transfer from the phone buttons (Polycom 501) work fine, so this isn’t a showstopper, just a “WHY??”. Thanks for you attention. Danny Nicholas ___ -- 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] Queue Issue (1.4.21.1)
The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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] Queue Issue (1.4.21.1)
I think the handling of this may have improved in later versions of Asterisk - is an upgrade an option? (I tested this with a newer version of Asterisk recently, and it behaved how you were hoping it would behave) PaulH Kev Szaszvari wrote: The strange thing is, Queue calls are working as per expected. If they get a call from the queue they wont get another until the 1st call is done. Its only when the agent received a direct call or a internal call from another staff member, the queue continues to ring their phone. - Original Message - From: Kev Szaszvari [mailto:k...@mailcall.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:36:32 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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 This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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 This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want
Re: [asterisk-users] Learn Asterisk
I can definitely recommend the 'sit down and play with it' website. Worked for me. PaulH David @ULC wrote: What the best website and book to start learning asterisk ? ___ -- 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] agent login status visual clue on Polycom?
From memory, it is doable but this is a feature that Polycom never quite finished writing. PaulH On Fri, 2009-06-19 at 10:58 +0200, Louis-David Mitterrand wrote: Hi, Is there a way on Polycom phones to show an agent whether he is logged in or not? ___ -- 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] Basic Config
Farooq Hussain wrote: Dear All, Please help as I new to Asterisk. I want know something what is Trunk what it will do. And I want to create a Dialplan like bellow: 1. It ask to dial a extension. 2. User will dial a extension. 3. User will be routed to that extension. I also want to connect my asterisk phone with my Analog Telephone which device will need to do this. And how I will created a dialup plan for it Any time spent reading this book will be well spent: http://downloads.oreilly.com/books/9780596510480.pdf PaulH ___ -- 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] visp multiaccount + firewall configuration problem
Alex Samad wrote: Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it did not seem to have any problem. now I have placed another box to act as a firewall in front of the asterisk box and I can't seem to register both lines. the sip account details are the same except for the username + id. so same destination ip. I would guess what I would really like to do is set a bindport for a particular account. port = 5061 PaulH ___ -- 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] Faxing issues
Todd S wrote: What's the bets way to verify T.38 is being used on both incoming an outgoing transaction? 3 to 1 in favour of not working. ;) PaulH ___ -- 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] PSTN Connection
Digium PSTN cards seem to work. PaulH Manoj Panicker - FOES wrote: Hi Which is the best interface card to connect* PSTN* line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj ___ -- 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] Open source SIP client
Not true. I am always wrong. (wait...is that a paradox?) PaulH ContactTel Business wrote: Niecly said.. hoeever, these list are not for astrix users, butt for bashing, didnot you realise this ? It had where 4 years more , know that this is fluent in this site. Translated as in : this list is a bash fest since i can remember back in 2004, everyone is right, no one is wrong, everyone is a god, and so on. However you made a point that will get tossed back in the “pit of endless replies” however good a point it was. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno *Sent:* May-18-09 7:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Open source SIP client It seems like a few people including me DID understand what Dhaval meant, or maybe some people used they common sense and their intelligence to understand what somebody who's english is not the primary language wanted to say and put some effort to guide or help someone in the community getting to the right direction instead of trying to put him down. I think a few others need to consider investigating more deeply the basic mechanics of understanding written English, or should themselves research what some collections of syllables intend to convey. I also think if they were that good, why not provided some english tutoring instead of putting people down. Good luck in you research Dhaval! On Mon, May 18, 2009 at 9:46 AM, Scott Gifford sgiff...@suspectclass.com mailto:sgiff...@suspectclass.com wrote: DHAVAL INDRODIYA dhaval.it01...@gmail.com mailto:dhaval.it01...@gmail.com writes: can anybody help me to give Opensource SIP client information which can be modified as per our requirment Hello Dhaval, We have tried several open-source SIP phones on Linux. We have had the best luck with Twinkle Phone: http://www.xs4all.nl/~mfnboer/twinkle/index.html http://www.xs4all.nl/%7Emfnboer/twinkle/index.html It has lots of hooks where you can stick your own scripts to modify its behavior. We also had pretty good luck with SFLphone: http://www.sflphone.org/ There is a list of open source clients on voip-info that includes these two. It might be a good starting point: http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software Good luck! Scott. ___ -- 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
Re: [asterisk-users] Problem with Asterisk + TDM410 FXO
I think you have your line types mixed up - FXS is for phones, FXO is for lines. An analogue passthorugh setup _is_ doable, just not overly recommended. PaulH Alex Samad wrote: Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to a fax line and used just in case the internet connection is down. I have tested the pstn line connection with a soft phone and it seems to be working fine. I need some help on how to tell asterisk to ignore the line for incoming ! when I connect the PABX to the FXO ports I ran into a problem. It seems to register okay, I pick up the handset on the pabx and select line 1 and i can hear a dial tone (same with line2) - this is the same what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in use. But I can't hear anything from the pabx - no dtmf tones and thus can't dial! when I try dialing in from the internet to asterisk then to ZAP/g1 the pabx can see the ring and I can pick up the phone I can hear the other end, but they can't hear me. I don't believe its a firewall issue as I can't dial from the pabx okay some print outs # zaptel_hardware pci::05:02.0 wctdm24xxp+ d161:8005 Wildcard TDM410P # ztcfg -vv Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. # cat /etc/zaptel.conf fxsks=4 fxoks=1,2,3 loadzone=au defaultzone=au /etc/asterisk/zapata.conf # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$' [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ks rxwink=300; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 Group=1 signalling=fxo_ks context=in-pbx channel=1-2 Group=2 echocancel=yes signalling=fxs_ks context=in-pstn channel=4 Group=3 signalling=fxo_ks context=in-spare channel=3 the thing that has me beet is that it work with the spa9000 I would expect it to just sort of work with the digium card. the os is debian amd64 2.6.26 #dpkg -l asteri* | grep ^ii ii asterisk1:1.4.21.2~dfsg-3 Open Source Private Branch Exchange (PBX) ii asterisk-barbarast.com 0.0.0-1 asterisk setup for hme1.samad.com.au ii asterisk-doc1:1.4.21.2~dfsg-3 Source code documentation for Asterisk ii asterisk-sounds-extra 1.4.7-1 Additional sound files for the Asterisk PBX ii asterisk-sounds-main1:1.4.21.2~dfsg-3 Core Sound files for Asterisk (English) #dpkg -l zapt* | grep ^ii ii zaptel 1:1.4.11~dfsg-3 zapata telephony utilities ii zaptel-modules-2.6.22-2-amd64 1:1.4.11~dfsg-3+2.6.22-4 zaptel modules for Linux (kernel 2.6.22-2-am ii zaptel-modules-2.6.26-2-amd64 1:1.4.11~dfsg-3+2.6.26-15 zaptel modules for Linux (kernel 2.6.26-2-am ii zaptel-source thanks Alex ___ -- 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] Problem with Asterisk + TDM410 FXO
Alex Samad wrote: On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote: I think you have your line types mixed up - FXS is for phones, FXO is for lines. sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is that a attached fxs presents internally as a fxo I have a pstn line attached to the FXO and I have my pabx attached to 2 FXS ports, which signal as fxo into asterisk (I could be wrong about that). By reading your configs below, you could be right - ports 1,2 and 3 are FXS, while 4 is FXO. What happens if you make a call in from the old fax line and send that over to the old PABX? Does that work OK? You could also buy some IP phones or put softphones around. That would solve the problem (you said that a softphone worked OK) PaulH An analogue passthorugh setup _is_ doable, just not overly recommended. PaulH Alex Samad wrote: Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to a fax line and used just in case the internet connection is down. I have tested the pstn line connection with a soft phone and it seems to be working fine. I need some help on how to tell asterisk to ignore the line for incoming ! when I connect the PABX to the FXO ports I ran into a problem. It seems to register okay, I pick up the handset on the pabx and select line 1 and i can hear a dial tone (same with line2) - this is the same what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in use. But I can't hear anything from the pabx - no dtmf tones and thus can't dial! when I try dialing in from the internet to asterisk then to ZAP/g1 the pabx can see the ring and I can pick up the phone I can hear the other end, but they can't hear me. I don't believe its a firewall issue as I can't dial from the pabx okay some print outs # zaptel_hardware pci::05:02.0 wctdm24xxp+ d161:8005 Wildcard TDM410P # ztcfg -vv Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. # cat /etc/zaptel.conf fxsks=4 fxoks=1,2,3 loadzone=au defaultzone=au /etc/asterisk/zapata.conf # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$' [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 Group=1 signalling=fxo_ks context=in-pbx channel=1-2 Group=2 echocancel=yes signalling=fxs_ks context=in-pstn channel=4 Group=3 signalling=fxo_ks context=in-spare channel=3 the thing that has me beet is that it work with the spa9000 I would expect it to just sort of work with the digium card. the os is debian amd64 2.6.26 #dpkg -l asteri* | grep ^ii ii asterisk1:1.4.21.2~dfsg-3 Open Source Private Branch Exchange (PBX) ii asterisk-barbarast.com 0.0.0-1 asterisk setup for hme1.samad.com.au ii asterisk-doc1:1.4.21.2~dfsg-3 Source code documentation for Asterisk ii asterisk-sounds-extra 1.4.7-1 Additional sound files for the Asterisk PBX ii asterisk-sounds-main1:1.4.21.2~dfsg-3 Core Sound files for Asterisk (English) #dpkg -l zapt* | grep ^ii ii zaptel 1:1.4.11~dfsg-3 zapata telephony utilities ii zaptel-modules-2.6.22-2-amd64 1:1.4.11~dfsg-3+2.6.22-4 zaptel modules for Linux (kernel 2.6.22-2-am ii zaptel-modules-2.6.26-2-amd64 1:1.4.11~dfsg-3+2.6.26-15 zaptel modules for Linux (kernel 2.6.26-2-am ii zaptel-source thanks Alex ___ -- 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
Re: [asterisk-users] Problem with Asterisk + TDM410 FXO
Have you tried plugging analog phones into the FXS ports in the Asterisk box? That should let you know what the Asterisk is really doing with it's FXS ports. PaulH Alex Samad wrote: On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote: I think you have your line types mixed up - FXS is for phones, FXO is for lines. sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is that a attached fxs presents internally as a fxo I have a pstn line attached to the FXO and I have my pabx attached to 2 FXS ports, which signal as fxo into asterisk (I could be wrong about that). An analogue passthorugh setup _is_ doable, just not overly recommended. PaulH Alex Samad wrote: Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to a fax line and used just in case the internet connection is down. I have tested the pstn line connection with a soft phone and it seems to be working fine. I need some help on how to tell asterisk to ignore the line for incoming ! when I connect the PABX to the FXO ports I ran into a problem. It seems to register okay, I pick up the handset on the pabx and select line 1 and i can hear a dial tone (same with line2) - this is the same what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in use. But I can't hear anything from the pabx - no dtmf tones and thus can't dial! when I try dialing in from the internet to asterisk then to ZAP/g1 the pabx can see the ring and I can pick up the phone I can hear the other end, but they can't hear me. I don't believe its a firewall issue as I can't dial from the pabx okay some print outs # zaptel_hardware pci::05:02.0 wctdm24xxp+ d161:8005 Wildcard TDM410P # ztcfg -vv Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. # cat /etc/zaptel.conf fxsks=4 fxoks=1,2,3 loadzone=au defaultzone=au /etc/asterisk/zapata.conf # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$' [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 Group=1 signalling=fxo_ks context=in-pbx channel=1-2 Group=2 echocancel=yes signalling=fxs_ks context=in-pstn channel=4 Group=3 signalling=fxo_ks context=in-spare channel=3 the thing that has me beet is that it work with the spa9000 I would expect it to just sort of work with the digium card. the os is debian amd64 2.6.26 #dpkg -l asteri* | grep ^ii ii asterisk1:1.4.21.2~dfsg-3 Open Source Private Branch Exchange (PBX) ii asterisk-barbarast.com 0.0.0-1 asterisk setup for hme1.samad.com.au ii asterisk-doc1:1.4.21.2~dfsg-3 Source code documentation for Asterisk ii asterisk-sounds-extra 1.4.7-1 Additional sound files for the Asterisk PBX ii asterisk-sounds-main1:1.4.21.2~dfsg-3 Core Sound files for Asterisk (English) #dpkg -l zapt* | grep ^ii ii zaptel 1:1.4.11~dfsg-3 zapata telephony utilities ii zaptel-modules-2.6.22-2-amd64 1:1.4.11~dfsg-3+2.6.22-4 zaptel modules for Linux (kernel 2.6.22-2-am ii zaptel-modules-2.6.26-2-amd64 1:1.4.11~dfsg-3+2.6.26-15 zaptel modules for Linux (kernel 2.6.26-2-am ii zaptel-source thanks Alex ___ -- 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
Re: [asterisk-users] Problem with Asterisk + TDM410 FXO
Alex Samad wrote: On Thu, May 14, 2009 at 03:31:18PM +1000, Paul Hales wrote: Have you tried plugging analog phones into the FXS ports in the Asterisk box? good ideal, but trying to find an old style phone the site has a commander PABX with digital handsets. I will see if I can track one down :) A You could use the fax machine (if it has a handset). Failing that, you will have one at home. This is most likely just a question of getting some settings right, and an analog handset will be a quick way to check how close you are. PaulH ___ -- 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] Problem with Asterisk + TDM410 FXO
Alex Samad wrote: On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote: What happens if you make a call in from the old fax line and send that over to the old PABX? Does that work OK? not sure what you are asking here. I have checked an incoming call through the FXO(PSTN) through to a FXS port (pabx) Testing a phone call from the outside world, into the fax line, into the asterisk box and then to the PABX. This avoids all networking. PaulH ___ -- 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] Understanding Codecs
I got very excited when I read the title of this email - I was hoping someone had learnt to speak g729. Ah well. PaulH Adrian Marsh wrote: Hi, I’m having problems with an asterisk server that’s not offering Codecs for ulaw and alaw as it should. I’ve three servers in total: a1, a2 and “b” A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP *Adding codec 0x4 (ulaw) to SDP* *Adding codec 0x8 (alaw) to SDP* Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 15554 Adding codec 0x2000 (amr) to SDP Adding non-codec 0x1 (telephone-event) to SDP Its not building ulaw or alaw into its list. Server B doesn’t support AMR, so rejects the call. (I’ve no idea about the 0x4000 error – but I see it on both the good and bad servers, so I don’t think its related). The odd thing is that the sip.conf files for A1 and A2 are exactly the same (save IP info). The build of the Asterisk server is from a 1.4.15 private build to add AMR, but, it’s the same source built on both A1 and A2. I’m trying to figure out why A2 isnt offering ulaw and alaw. The codec seems ok, and is listed in the show codecs: 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 8192 (1 13) (0x2000) audio amr (AMR) But I cant see why its not transcoding across to ulaw/alaw. Thanks, Adrian ___ -- 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] Building a System.
You don't need freepbx. In all honestly, building a system from scratch isn't too bad if you having some decent (or indecent?) linux karma. If you don't, it's going to be a rather unfun time. PaulH John F. Ervin wrote: So, people have recommended building a system from scratch, start with a CentOS base and installing asterisk and all of the other utilities. I've only used Trixbox for my business system. I'm wondering what surprises I'd run into. Right now, I know I'd need the OS, Asterisk, something like FreePBX, I have a x100p card so I'd need Zaptel, does that come with asterisk? Fax support, seems to work with Trixbox, but I've heard that it needs to be loaded. Voicemail etc.? I mean, I don't know exactly what you'd need because almost everything I need comes with the Trixbox build. Are there (??) instructions for people who are experienced at the Trixbox level but wish to move on? -- John F. Ervin *Central Florida TeleSource, LLC.** *4270 Aloma Ave #124-69C Winter Park, FL 32792 (W) 407-679-6238 (F) 866-566-1282 (F) 321-445-0781 jer...@jervin.com mailto:jer...@jervin.com http://jervin.com/cft ___ -- 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] Building a System.
George Kwabenah Appiah wrote: Are there (??) instructions for people who are experienced at the Trixbox level but wish to move on? If you'd like to get a solid foundation on Asterisk and how the various pieces fit together, I suggest you invest a couple hours and go through the O'Reilly book: Asterisk - The Future of Telephony (free PDF / HTML of entire book at http://astbook.asteriskdocs.org/). That's great advice - it's a great book. PaulH ___ -- 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] How to get PBX's clock with AMI?
I am still not sure what you are askingis it something to do with NTP? PaulH Daniel - Asterisk wrote: I guess it was a problem with my connection, here the complete question.. Dear all, I wanna know what can I do to get the PBX's clock from an external AMI server, especially with Asterisk-Java Library. Thanks by your answers. Elder Arohuanca Lagos t. +51 1 994149553 On Tue, Apr 28, 2009 at 11:00 AM, Steve Howes st...@geekinter.net mailto:st...@geekinter.net wrote: On 28 Apr 2009, at 16:49, Daniel - Asterisk wrote: Dear all, I wanna know what can I do to get the PBX's clock from You sir, are made of fail. ___ -- 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
Re: [asterisk-users] How to get PBX's clock with AMI?
Steve Howes wrote: On 28 Apr 2009, at 16:49, Daniel - Asterisk wrote: Dear all, I wanna know what can I do to get the PBX's clock from You sir, are made of fail. I had to admit, I laughed. PaulH ___ -- 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] Changing menuselect values from CLI and not TUI
Is a drive image out of the question? PaulH David Klaverstyn wrote: Hi All, I’m in the process of writing an install script and I would like to change some settings for the install process but I don’t want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the iLBC codec. menuselect codec ilbc on Regards David. ___ -- 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] Best Practice Advice?
I would upgrade to the latest 1.4, if stable is what is needed. PaulH Gabriel - IP Guys wrote: Dear All, I have a asterisk setup that is currently running on version 1.4.15 – I wish to upgrade or migrate this instance to the current asterisk stable, 1.6.0.6. It is my intention to build a FC8 box, then install asterisk from source, and begin to migrate over the configuration. The thing is, this sounds so simple in my head, and I’ve had enough issues with asterisk, to know that life isn’t simple! What I plan to do, is to copy the old configuration over to a box running FC8 – and then compile and run asterisk 1.4.15 – and gradually upgrade it, until I reach 1.6.0.6 – Any input on this matter will be appreciated. Thank you --- Mr Gabriel ___ -- 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] Advice
If there is an asterisk users group in your area, visit and ask lots of questions. PaulH Roland Roland wrote: Hi all, a few month ago I got the task of setting up asterisk for my company. I had 94 employee to set this up for ... I never heard of asterisk before to b honest, so after researching a bit.. I started with a digium card with ZAP though that didn’t work out as the card were flawed.. so ended up setting up SIP for everyone using a SIP callcentric accounts as well as sipura for pstn lines.. now it's working at it's minimal state.. but as am out of the heat of pressure from management.. so now It's time to learn about asterisk the right way as I had lots of help from this mailing list as well as the IRC channel that I'm not sure I could do it again on my own.. so not to add more to my email, I'm seeking advice about the proper way to learn about asterisk from A to Z if possible... any advice would be appreciatedSmile emoticon thanks in advance, Roland ___ -- 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] What is the one thing that polycom can do...
Karl Fife wrote: On the landing page of the Polycom web site there's a We're listening nanosurvey, asking what is the one thing Polycom can do to improve their products. The link points here: http://polycom.zuberance.com/survey.htm I wrote a sentence about tweaking the user interface on the IP Soundpoint series phones, so that one can escape any level of any menu with repeated pressing of the same softkey, rather than having to hunt for the appropriate label (a moving target). I only mention it because many have voiced the same complaint on the Voip Users Conference weekly conference/podcast. While not the one 'big picture' item that polycom should be focusing on, one could reasonably argue it would rank right up there in terms of 'bang for the buck' because it would be such an easy tweak. If this irks you as well, go make yourself heard. If seveeral people say the same thing, perhaps they'll do something about it. Maybe they actually ARE listening :-) -Karl I would love to see the agent login/logout stuff working - but that's just me. PaulH ___ -- 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] usb-phones
The Asterisk console is pretty goodbut there was a text version of one of the softphones once (sjphone, if I remember correctly) PaulH Hans Witvliet wrote: While reading the thread about recommending usb-phones... Once in a while, i'm in a data-centre, no normal phones, and too much concrete shielding wireless phones. So i was thinking to use one of those usb-phones, and plug it into one of my servers there. But what i read from the thread, i seems that you need a graphical environment, while all of the servers are strictly cli-only. Is there a cli-based phone (besides the asterisk-console), that can use a usb-audio-device? Afaicr,those usb-phones present themselves as an plain usb-audio device. hw ___ -- 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] strategy ringall
Are both of your agents logged in? What does the CLI show? PaulH edw...@web.de wrote: Hi, I have a problem with queue strategy. But only 1 of my agents ring, when someone call. my queue.conf: [MyQueue] strategy=ringall member = Agent/201 member = Agent/202 announce-holdtime = yes joinempty = strict leavewhenempty=yes my extension.conf: exten=8708464,1,Answer exten=8708464,n,Ringing ;exten=8708464,n,Wait(2) exten=8708464,n,Queue(MyQueue50) I have 1 voip telephone and 1 x-lite. Is this a bug? --- edwin ___ -- 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] music-on-hold kicks in and disconnects/interrupt the call
Joseph wrote: I'm using Asterisk 1.4.22.1 When I'm on active call it happens many times the call gets interrupted by music-on-hold without my pressing any button. MOH just kicks in and int erupt the call and I have no way of getting the call back. Did anybody experienced anything like this? No - do you have any dialplan code or cli output to show for this excitement? PaulH ___ -- 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] queued?
Any idea what this means? And why they are different? - Extension Changed 22142[default] new state Idle for Notify User 31001 (queued) Extension Changed 22142[default] new state Idle for Notify User 30060 - I have googled and searched, and can't find anything on this subject. Does anyone have an suggestions? PaulH ___ -- 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] Good phone near $125
SIP wrote: I believe SNOM 300s do PoE (might have to check that, though) and are around $100. We've little experience with them, but we use an office full of Snom 320s, and we're nothing but pleased with them. Good speaker, good handset, lots of excellent options. And reasonably priced. N. The first generation of Snom 300's did _not_ support POE - but later models did. PaulH ___ -- 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] 428 Loop Detected
I am probably missing something, being a newbie. I have a 4 port fxs/fxo (2/2) card. My land line is going to one of the FXO port and my home phone is connected to one of the FXS port. I want to be able to call my phone number from external phone (cell phone) and have my home phone ring. And if I do not pick up the phone in 10 secs I want the voicemail to pickup the call. I do have a dialtone when pick up my phone that is attached to the FXS port of my asterisk server A printout from the CLI would be helpful - but I think you have your contexts crossed over. (call from outside hitting internal, instead of from-pstn) PaulH ___ -- 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] automatic call bridging when destination is available feature
I can't force them to use star codes to set DND in astdb). Once again, someone who underestimates the power of physical violence. PaulH ___ -- 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] DAHDI and B410P (BRI)
I wish it was available too - I have just had to back dahdi out of a system and revert to misdn after a whole day of testing. PaulH Andrew Thomas wrote: I have LibPri installed and working (.../wPRI). So, if I understand Tzafrir correctly - DAHDI support for the B410P isn't available in 1.4 at all. Looks like I'm going back to mISDN. Cheers Andy -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Jose Luis Villalon -- Sent: 09 March 2009 18:07 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] DAHDI and B410P (BRI) -- -- Hi -- -- What it's the result of execute -- -- strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI -- Telephony' -- -- It's LibPri install before of Dahdi package? -- -- JL. -- -- El Lun, 9 de Marzo de 2009, 6:36 pm, Andrew Thomas escribió: -- Hi all, -- -- -- I am having trouble setting the signalling method for the B410P -- using -- DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or -- 'bri_net' - but it doesn't mind having 'pri_cpe' etc. -- -- -- ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling -- method -- 'bri_net' -- -- -- Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 -- Asterisk - 1.4.23.1 -- Libpri - 1.4.9 -- -- -- I have set the spans up with no problems (well, dahdi_cfg doesn't -- complain) - it's just my chan_dahdi.conf file I need to fix now. -- -- Thanks -- Andy -- -- -- -- ___ -- -- 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 ___ -- 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] Outlook integration?
Noojeeclick? http://www.noojee.com.au/Page/NoojeeClick ADM? (asterisk desktop manager?) PaulH Alan Lord (News) wrote: Dean Collins wrote: ADA Forums: http://forums.digium.com/index.php?c=8 ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe ADA Administrators Guide: http://dl1.digium.com/ADA1.1/ADA_Admin_Manual.pdf Thanks for the links. I hadn't seen that before. The product is kind of interesting, but does anyone know of something similar for non-windows desktops? Thanks Al ___ -- 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] Configuring asterisk to revert call back to forwarder if exten is busy
Can I assume that you want this only for blind transfers?
I have done this previously, but I lost my copy of the work (and it was
a proof of concept only)
It involved the ${BLINDTRANSFER} variable, which catches the number that
made the blind transfer and making macro-stdexten (or your equivalent)
dial that variable in the case of the dial status being treated as BUSY.
To get a 'busy' will involve single line phones, or disabling call
waiting on the phone receiving the call.
regards,
PaulH
James Mutuku wrote:
Hellos,
I want to configure asterisk so that if exten A transfers a call to
exten B, and B is either busy or the call is not answered, the call
returns back to A. Is this possible?
Please help
James
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Re: [asterisk-users] Dialing with cli
Have a look at 'call files' on voip-info.org Great fun, especially for load testing. PaulH Joseph L. Casale wrote: Any way to initiate a call and execute a playback of an audio file from the cli? My only chance to debug or make changes is usually when no one's at the office including me! Thanks! jlc ___ -- 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] asterisk 1.6.0.5 and IM
I tried to get some of that stuff going a while ago, and it just didn't work - like the polycom user status stuff (at lunch) Asterisk sees the bits of info but doesn't want to handle it. PaulH lord_f...@iinet.net.au wrote: hi all, i have 2 x-lite version 3.0 softphones configured on extension 9000 and 9005. i have one call the other and then try and send an IM between them using the x-lite IM facility. the asterisk console shows the message... WARNING[27193]: chan_sip.c:11866 receive_message: Received message to s9...@hhh from c9...@hhh;tag=717de473, dropped it... when i look at the code theres a comment in there saying Message outside of a call, we do not support that . but both phones say they're connected, any ideas? is there some config file option i need to set? thanks, fleg ___ -- 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] Call from '6000' to extension rejected because extension not found
Please read this book: http://downloads.oreilly.com/books/9780596510480.pdf PaulH Chuck Coleman wrote: Call from '6000' to extension 'xx' rejected because extension not found. ___ -- 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] DTMF Forwarking Problems.
Check features.conf - all the codes are set in that file. PaulH Catalin S. wrote: Hello ppl, I have a problem with my asterisk when i want to call some destination through my peers and I must enter DTMF digits to select some extension/conference number or password to access some features.Every numbers is accepted but when i must press # key my asterisk interpret it like transfer options. I want to know how can i activate and deactivate transfer mode of # key on my desired peers. Thank you very much, Catalin. ___ -- 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] check if not human
I should get all my non-human friends to call this number...or do you want to call non-humans? PaulH Edwin Quijada wrote: NVLineDetect , I dont find it in the web for asterisk 1.4 Anybody has a link that works? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Date: Fri, 20 Feb 2009 09:31:41 -0800 From: nt_aster...@yahoo.com To: asterisk-users@lists.digium.com CC: nt_jnew...@yahoo.com Subject: Re: [asterisk-users] check if not human NVGenderDetect is new, but you can find NVLineDetect on the web. *From:* David fire ddf...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Thursday, February 19, 2009 3:00:14 PM *Subject:* Re: [asterisk-users] check if not human NVLineDetect, NVGenderDetect what is that? amd info voip-info.org or asterisk.org support asterisk book. i bougth one to support the cause!!! David 2009/2/19 Asterisk Asterisk nt_aster...@yahoo.com You can probably use combo of NVLineDetect, NVGenderDetect, and AMD (NVMachineDetect). *From:* Edwin Quijada listas_quij...@hotmail.com *To:* Asterisk Asterisk asterisk-users@lists.digium.com *Sent:* Thursday, February 19, 2009 12:55:05 PM *Subject:* Re: [asterisk-users] check if not human How can I detect how many ring a call to hangup? Where I can find info about AMD? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Get Windows Live and get whatever you need, wherever you are. Start here. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. See how Windows® connects the people, information, and fun that are part of your life http://clk.atdmt.com/MRT/go/119463819/direct/01/ ___ -- 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] HDD FULLL
If you are being paid to work on an Asterisk system, you are in over your head. You are defrauding your boss and most likely will give him and everyone in the company a bad impression of Asterisk. Continuing to answer your questions will only continue to enable you. Please take a step back, buy some books, take some courses, practice on your own systems on your own time. I will continue to read your posts, but only for comic relief. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ That was the funniest thing I have read in a while - thanks! PaulH ___ -- 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] zaptel telephone cards and asterisk in another pc
Ignacio wrote: Hello, I have some zaptel cards, and I would like to install them in some user's computers. Is there any way to connect those cards with asterisk server (which is in another computer)? All manuals I have read explain how to connect asterisk and zaptel cards in the same computers, but not on different ones. Sometimes things that look the same can be different, and things that look different can be the same. PaulH ___ -- 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] zaptel telephone cards and asterisk in another pc
Why do you need so many Asterisk installs? With the ability of Asterisk to handle hundreds of lines/phones/etc, the need for several Asterisk server is generally for very specific situations. PaulH Ignacio wrote: Jeff I will take a more depth look at those linksys devices this weekend but I think they could be very interesting. Tzafrir, what I like to avoid is installing an asterisk server in every user computer. I think that is useless I want only one server to mantain. On Fri, Feb 20, 2009 at 7:55 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Fri, Feb 20, 2009 at 07:11:04PM +0100, Ignacio wrote: Thank you very much for your fast answer Eric. I was trying to avoid to have to install as many asterisk as pcs I have. But I think there is no way to do it. I only have seen something like network block device, but not sure if it is going to work and quite difficult to configure properly. Anyway I think the fast and easier way will be installing and asterisk in every client. I guess you can use TDMoE. But I'm not really sure it will give you a lower overhead. Specifically, why is it that you want to avoid installing Asterisk there? The requirements of an Asterisk system for a few analog channels and a few uncompressed SIP/IAX channels are rather minimal. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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
Re: [asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)
The old classic is to say something like ' your callerid is blocked, please get out your credit card' PaulH Alfred Monticello wrote: I'm thinking of starting a partyline, where people call in and talk to other people. For record keeping and billing purposes, I'd like to go by the callers telephone number. This method works fine as long as the caller doesn't have callerid blocked, but what are my options if they do block their number? I know there must be a way to report it, because there is a service provider here in my area that if I call and block my number, they are still able to obtain it. I know that when dialing a toll-free number, that the number is reported regardless. But what about regular non-toll free numbers? Does anybody have any ideas how I can do this? Are there any providers out there that offer this service over PRI or some other method? Thank you in advance. ___ -- 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] Slow hangup - Australia - analog - incoming calls
I have had to install a TDM800 in a site, as the telco has held off installing ISDN indefinitely.. It's all fine except for the fact that it takes ages to hang up the line (6 or more rings), and sometimes doesn't even bother. This is only on incoming calls - outgoing calls work perfectly. Is there any good tricks for a fast and accurate hangup detect in this situation? 'callprogress=yes' helped but gave us random hangups! Changing 'busycount' didn't help, and the fact that 1 call in 8 doesn't hang up at all. Is polarity reversal the only way to go? PaulH ___ -- 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 - Trixbox
I don't think scary is a strong enough wordterrifying? horrifying? abominable? PaulH Steve Totaro wrote: Your carrier is running Trixbox? That is scary. Anyways, they are obviously routing calls to the wrong machine. If your side worked properly before and now does not, then they have to explain why. That would be my stance anyways. Thanks, Steve On Mon, Feb 2, 2009 at 10:18 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Mike Hammett asterisk-us...@ics-il.net Sent: Thursday, January 29, 2009 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.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
Re: [asterisk-users] Quiet 24 port POE gig switch
This whole thread is getting stupid and I'd hope the people involved would desist from this O/T drivel. If you want a switch go to the shop, hand over some money and buy one... Like every one else does and they're perfectly happy with their purchase. The O.P. is not going to change the world and quite frankly the designer/manufacturer of the product knows a lot more about the subject then they do It's really just a lot of hot air. (ducks and runs for cover) PaulH ___ -- 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] Quiet 24 port POE gig switch
My memory of a HP procurve (a 2626 PWR from memory) was that it was quite noisy - have they changed? PaulH OCG Technical Support wrote: Check out the HP ProCurve Switch 2610-24-PWR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: February 1, 2009 6:58 AM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch I can find FANLESS 24 port PoE 10/100 That's an achievement in itself. Can you post details - I have quite a few locations where that might be useful... TIA. Regards, Chris ___ -- 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
Re: [asterisk-users] Callback / Camp / Extention Free notify?
Quick solution that comes into mind:
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if (${DIALSTATUS}=BUSY) {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h = {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if (${call_to}!=) {
Set(DB(camp/${exten_copy}/call_to)=);
System(call_to ${exten_copy} ${call_to});
}
}
So, in case if phone2 is busy, store callerid of phone1 in database,
so when phone2 will hangup it will triger a script call_to which
however can originate call trough manager or call-file.
Of course you will need some additional handling in case if multiple
callers decide to camp, or diferent protocols are used, etc.
You could call a batch script from the dialplan that parses the output
of 'show hints' with a simple grep to find the status of the individual
in question.
PaulH
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Re: [asterisk-users] Callback / Camp / Extention Free notify?
Yes, the more expensive ones do. The majority do not. Linksys phones. I have a Snom 320 and an Aastra 480i on my desk, and one of the reasons I love them (especially the Aastra) is the BLF features. Its not so much knowing if the user is busy or not, its the ability to be automatically notified once the user becomes available. * *No problem - it will be doable, it's just how much effort will be needed to get it working, and then how much more effort to get it 'perfect'. I know that a lot of people have been through exactly what you are going through with regards to legacy features - I had to write a piece of dialplan code to return blind transfers back to the person who started the transfer if the extension they were calling did not answer...just like the old phone system they had...because attended transfers were too hard. later, PaulH ** ___ -- 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] I need help
There are no good Mexican restaurants near my house. PaulH Jose P. Espinal wrote: And your problem is... ? Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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
Re: [asterisk-users] soft ATA on linux with zaptel?
Brian J. Murrell wrote: Slightly OT, but I'm wondering if anyone here has come across a soft ATA. That is, software that will perform the functions of a basic POTS line ATA on Linux with a zaptel driven card. I have a Linux machine with a zaptel card in it and I want to have another Linux machine running Asterisk utilize the zaptel card in the first Linux machine to make outgoing and receive incoming calls. I realize I could make Asterisk do this job, but it seems pretty heavy- weight for just that purpose -- of bridging a POTS line to a SIP (or IAX) connection. Ideas? b. Is a hard-ATA (such as the linksys 3xxx) really out of the question? If you figure out how much you are worth an hour, it might be cheaper to work, earn money and buy the ATA. PaulH ___ -- 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] Dumb question: retrieve values from OS-level commands?
Not sure if this is still valid - I used it on a project quite a while ago: http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks PaulH Ken D'Ambrosio wrote: Hi, all. I want to execute a script, and return the value of said (Python) script to the dialplan. I thought something like exten = 1,1,Set(MyWorkingDir=System(/bin/pwd)) might work, but apparently not. I also looked into AGI stuff, but that doesn't quite seem to be the right approach. Surely there's *some* way to do this... Any suggestions? Thanks! -Ken ___ -- 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] sip based fax
Google T38 - it's a big subject. PaulH amir...@namche.com wrote: Hi all, Can we configure sip based outgoing fax on asterisk or we must need zap channel attached with it? Thanx in Advance. Amir Shrestha ___ -- 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] FXS Help Needed...
I have used the xorcom usb units for fax a few times, and they work pretty well. PaulH Gregory Malsack wrote: Hello All, I have a need to connect an analog device to an asterisk server. The analog device has 4 analog lines going into it (it’s a fax solution). The fax solution answers the analog call, then listens for dtmf. The dtmf code that is played tells the fax device what email address to send the fax to. All calls on our system come into the server through a PRI. The faxes come in over a PRI, the current phone system routes the faxes to the device, then sends the dtmf, then bridges the fax transmission. Does anyone know how I can do this on an asterisk system? I have the PRI card, and have an 8 port fxs card in the system as well. Is it as easy as picking up the line and dialing the 4 digit dtmf, just like it was an fxo port? Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.552 / Virus Database: 270.10.6/1888 - Release Date: 1/12/2009 7:04 AM ___ -- 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] Not Dialing 9
This has worked fine for me (as far as I know). Is there some flaw I am not seeing? I see a lot of small businesses that require a 9 to dial out, even though they don't have very many extensions. Couldn't they do what I did and not have to dial 9? Many older systems _cannot_ process the call based on what is dialled (in Asterisk this is called 'pattern matching') - so the first digit (ie: 9) tells the phone system that the user is about to dial an outside number. PaulH ___ -- 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] Allison Smith, Music-on-Hold Parody--outstanding.
Andrew Joakimsen wrote: On Wed, Dec 31, 2008 at 22:09, Paul Hales pdha...@optusnet.com.au wrote: Karl Fife wrote: Allison Smith just created a hysterical parody music on hold Parody. Whatever you were doing, stop, and dial this number to listen to it: 360-519-5689. 2 minutes. I just gave her a few ideas, but she took it and ran with it--she chose the audio and did the mix-down and everything. Really funny!! -Karl Any chance of us non-us citizens hearing it? (podcast, download...) I put up a recording here: http://app5.netjdn.com/~joako/karl.wav I hope Karl doesn't mind. Excellent work - from yourself, Allison and Karl. PaulH ___ -- 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] Allison Smith, Music-on-Hold Parody--outstanding.
Karl Fife wrote: Allison Smith just created a hysterical parody music on hold Parody. Whatever you were doing, stop, and dial this number to listen to it: 360-519-5689. 2 minutes. I just gave her a few ideas, but she took it and ran with it--she chose the audio and did the mix-down and everything. Really funny!! -Karl Any chance of us non-us citizens hearing it? (podcast, download...) PaulH ___ -- 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] top posting again [was: Re: CDR Design]
In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! I am strongly against anyone posting anything with their bottom. later, PaulH ___ -- 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] Zaptel / TDM400P card stopped working
This may be an obvious thing, but you didn't mention checking whether or not the card was still seated in the slot properly after the move. I know from experience that when you move offices, even if you take all the precautions possible, a card can get bumped just enough to jostle the connections loose. Even if the card appears to be seated correctly I'd take it out and re-seat it. Unfortunately it looks like you may have compounded the problem by removing and reinstalling the zaptel packages. It looked like the card was still there - from memory the lspci command said it was. PaulH ___ -- 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] Zaptel / TDM400P card stopped working
Langdon Stevenson wrote: Paul Hales wrote: It looked like the card was still there - from memory the lspci command said it was. PaulH That is correct, lspci shows the card is there. I have also tried moving the card to a different slot to be sure. Langdon So - the current state of play is: card = yes drivers = no As a stop gap, have you tried building the drivers from source? PaulH ___ -- 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] Zaptel / TDM400P card stopped working
Langdon Stevenson wrote: Yes, that is the current state of play and yes, it looks like I will have to build from source. I haven't done this before and am pretty busy at the moment, so it will take me a while. I will post back when I have done so. Thanks for the input (to all who have contributed), it is much appreciated. Regards, Langdon Building the drivers from source will only take you 10 minutes - not a huge hassle. PaulH ___ -- 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] Zaptel / TDM400P card stopped working
Have you tried loading the zaptel driver for your card manually? PaulH Langdon Stevenson wrote: Hi I have a Dell PE2300 with a Digium TDM400P line card in it (with one module to handle an inbound phone line). This is running on a Fedora 8 system with Asterisk 1.4.21.2-1.fc8 This system has been working nicely for about 12 months. After a recent move of office and relocation of the server Asterisk is back on line, but the TDM line card has stopped working. I have spent half a day working through Google search results, but no luck so far. The command: lspci -v produces: snip 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at e400 [size=256] Memory at f9ffd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel modules: hisax The IRQ is not in use by any other device, so there is no conflict (this seems to be a common problem). The card has always been detected as a Tiger3XX. What stands out here to me is: Kernal modules: hisax I don't believe that this was the case when I first installed the card (but it was over a year ago, so I may be wrong). The hisax driver is blacklisted in /etc/modprobe.d/blacklist. The command: lsmod produces: Module Size Used by xt_dscp 6465 0 rfcomm 32721 0 l2cap 21953 9 rfcomm bluetooth 47013 6 rfcomm,l2cap autofs420933 2 fuse 47837 1 tun12613 0 sunrpc154785 3 nf_conntrack_netbios_ns 6593 0 iptable_nat 8777 0 nf_nat 18393 1 iptable_nat iptable_mangle 6849 0 nf_conntrack_ipv4 11849 5 iptable_nat,nf_nat xt_state6209 2 nf_conntrack 51221 5 nf_conntrack_netbios_ns,iptable_nat,nf_nat,nf_conntrack_ipv4,xt_state ipt_REJECT 6977 2 ipt_LOG 9285 4 iptable_filter 6849 1 ip_tables 14033 3 iptable_nat,iptable_mangle,iptable_filter xt_tcpudp 6977 33 ip6t_REJECT 7617 2 ip6table_filter 6593 1 ip6_tables 15057 1 ip6table_filter x_tables 15557 9 xt_dscp,iptable_nat,xt_state,ipt_REJECT,ipt_LOG,ip_tables,xt_tcpudp,ip6t_REJECT,ip6_tables ipv6 238277 25 ip6t_REJECT dm_multipath 18505 0 parport_pc 26725 0 parport32173 1 parport_pc floppy 52229 0 i2c_piix4 11473 0 i2c_core 20949 1 i2c_piix4 pcspkr 6593 0 e100 33997 0 mii 8385 1 e100 dcdbas 10465 0 sr_mod 17541 0 cdrom 33249 1 sr_mod sg 31605 0 ata_piix 19397 0 libata131937 1 ata_piix raid1 22593 2 dm_snapshot18661 0 dm_zero 5825 0 dm_mirror 19521 0 dm_log 12229 1 dm_mirror dm_mod 48265 8 dm_multipath,dm_snapshot,dm_zero,dm_mirror,dm_log aic7xxx 101753 15 scsi_transport_spi 23233 1 aic7xxx sd_mod 26329 20 scsi_mod 123917 6 sr_mod,sg,libata,aic7xxx,scsi_transport_spi,sd_mod raid456 121681 1 async_xor 7361 1 raid456 async_memcpy6209 1 raid456 async_tx9869 3 raid456,async_xor,async_memcpy xor18633 2 raid456,async_xor ext3 110281 2 jbd41045 1 ext3 mbcache10309 1 ext3 uhci_hcd 22993 0 ohci_hcd 22853 0 ehci_hcd 32845 0 The command: ztcfg -v produces: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected The command: ls -al /dev/ | grep zap produces nothing So, I am left wondering what has changed and why the Zaptel drivers are no longer loading. Can anyone suggest to me how I might go about troubleshooting this issue? Langdon ___ -- 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] Zaptel / TDM400P card stopped working
h...I haven't used the RPM's before, so I can only guess that the RPM's are doing something not quite right. Is the asterisk-zaptel or the zaptel rpm supposed to provide the drivers? Does rpm -qf filename show the correct kernel version? If that fails, you could download the source files from the Asterisk site and build them yourself. PaulH Langdon Stevenson wrote: Hi Paul Thanks for the reply. I have removed and re-installed all of the Fedora Zaptel packages with Yum. I have the following installed: asterisk-zaptel 1.4.12.1-1.fc8 zaptel.i386 1.4.12.1-1.fc8 zaptel-devel.i386 1.4.12.1-1.fc8 zaptel-lib.i386 1.4.12.1-1.fc8 zaptel-utils.i386 1.4.12.1-1.fc8 The command: modprobe wctdm produces: FATAL: Module wctdm not found. The command: modprobe zaptel produces: FATAL: Module zaptel not found. Is there anything else that I should be doing? Regards, Langdon Paul Hales wrote: Have you tried loading the zaptel driver for your card manually? PaulH Langdon Stevenson wrote: Hi I have a Dell PE2300 with a Digium TDM400P line card in it (with one module to handle an inbound phone line). This is running on a Fedora 8 system with Asterisk 1.4.21.2-1.fc8 This system has been working nicely for about 12 months. After a recent move of office and relocation of the server Asterisk is back on line, but the TDM line card has stopped working. I have spent half a day working through Google search results, but no luck so far. The command: lspci -v produces: snip 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0001 Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at e400 [size=256] Memory at f9ffd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel modules: hisax The IRQ is not in use by any other device, so there is no conflict (this seems to be a common problem). The card has always been detected as a Tiger3XX. What stands out here to me is: Kernal modules: hisax I don't believe that this was the case when I first installed the card (but it was over a year ago, so I may be wrong). The hisax driver is blacklisted in /etc/modprobe.d/blacklist. The command: lsmod produces: Module Size Used by xt_dscp 6465 0 rfcomm 32721 0 l2cap 21953 9 rfcomm bluetooth 47013 6 rfcomm,l2cap autofs420933 2 fuse 47837 1 tun12613 0 sunrpc154785 3 nf_conntrack_netbios_ns 6593 0 iptable_nat 8777 0 nf_nat 18393 1 iptable_nat iptable_mangle 6849 0 nf_conntrack_ipv4 11849 5 iptable_nat,nf_nat xt_state6209 2 nf_conntrack 51221 5 nf_conntrack_netbios_ns,iptable_nat,nf_nat,nf_conntrack_ipv4,xt_state ipt_REJECT 6977 2 ipt_LOG 9285 4 iptable_filter 6849 1 ip_tables 14033 3 iptable_nat,iptable_mangle,iptable_filter xt_tcpudp 6977 33 ip6t_REJECT 7617 2 ip6table_filter 6593 1 ip6_tables 15057 1 ip6table_filter x_tables 15557 9 xt_dscp,iptable_nat,xt_state,ipt_REJECT,ipt_LOG,ip_tables,xt_tcpudp,ip6t_REJECT,ip6_tables ipv6 238277 25 ip6t_REJECT dm_multipath 18505 0 parport_pc 26725 0 parport32173 1 parport_pc floppy 52229 0 i2c_piix4 11473 0 i2c_core 20949 1 i2c_piix4 pcspkr 6593 0 e100 33997 0 mii 8385 1 e100 dcdbas 10465 0 sr_mod 17541 0 cdrom 33249 1 sr_mod sg 31605 0 ata_piix 19397 0 libata131937 1 ata_piix raid1 22593 2 dm_snapshot18661 0 dm_zero 5825 0 dm_mirror 19521 0 dm_log 12229 1 dm_mirror dm_mod 48265 8 dm_multipath,dm_snapshot,dm_zero,dm_mirror,dm_log aic7xxx 101753 15 scsi_transport_spi 23233 1 aic7xxx sd_mod 26329 20 scsi_mod 123917 6 sr_mod,sg,libata,aic7xxx,scsi_transport_spi,sd_mod raid456 121681 1 async_xor 7361 1 raid456 async_memcpy6209 1 raid456 async_tx9869 3 raid456,async_xor,async_memcpy xor18633 2 raid456,async_xor ext3 110281 2 jbd41045 1 ext3 mbcache10309 1 ext3 uhci_hcd 22993 0 ohci_hcd 22853 0 ehci_hcd 32845 0
Re: [asterisk-users] 'dialer' application to trigger call betweenhardphone and number
There are a few web-based ones - is that an option at all? PaulH Danny Nicholas wrote: This sounds like a job for a VB.NET programmer. The program would run like a DDE server and ftp a call file to your asterisk server on the desired action. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Karl Fife *Sent:* Monday, December 08, 2008 3:04 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] 'dialer' application to trigger call betweenhardphone and number Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via hotkey, or a right-click menu or by clicking a system tray icon. There are scads of outlook-only options (no thanks), and I've found and tried the Asterisk Dialer 1.0, which I don't like because it depends on Yahoo widgets (heavy) AND it requires nearly as many discreet actions to dial a number as just typing them on the phone itself. Ideal would be something very 'efficient' with at most two or three discreet actions needed to dial-- (i.e. 1:Select, 2:Hotkey--done!) Any ideas? Any Happy customers? ___ -- 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
