[asterisk-users] tls is up but no audio
Hi All, I'm headbanging on this from a couple of days, begging here for some help :) I'm configuring tls on asterisk for the first time to experiment with an open (public) service idea about having asterisk accepting any sip user (with the sip.conf option 'autocreatepeer=yes') and call each other on the same server and perhaps to other asterisk servers with the same configuration. Something like 'skype for poors' for the 'average joe'. I'm using asterisk 10.7.0 on a debian squeeze dedicated server (with public ip). I've followed this tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial and got no errors but when dialing a test context: exten => _X.,1,Answer exten => _X.,n,playback(tt-weasels) exten => _X.,n,echo exten => _X.,n,Hangup() i get no audio. On the client side, I've tried with many softphones (bink, jitsi, microsip, phonerlite) on both windows and linux, on two different computers but same result. I've also enabled srtp, checked the sip debug trace, recompiled libsrtp from sources, tried different combination of parameters in sip.conf, enabled and disabled some port forwardings on the client's router but same result: all looks ok, but i get no audio. If not using tls (but the usual udp and rtp), audio works full-duplex :) Anyone had a similar problem ? Any hints ? Let me know if i can provide more info. Thanks for supporting, regards and have a nice day, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi dial termination cause ?
On Wed, 19 Jan 2011 17:03:03 +0100
Thorsten Göllner wrote:
> Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:Hi All,
>
> in an AGI script, if executing the Asterisk command Dial, I only get
> result => -1 (if the call has been answered by the callee)
> and
> result => 0 (for everything else)
>
> Question:
> how can I know if the call was not answered because of timeout or because the
> callee was busy ?
>
> (I'm using Asterisk 1.8)
>
>
> Thank you very much for supporting,
> regards and have a nice day.
> Mike
> --
> Take a look here:
> http://www.voip-info.org/wiki/view/Asterisk+variables
>
> Perhaps you can get this info from ${ANSWEREDTIME} or from ${DIALSTATUS}.
>
> -Thorsten-
Ohh great! I have forgot about them, thank you both very much!
I confirm that if using phpagi the array $agi->get_variable("DIALSTATUS")
['data'] gets populated with NOANSWER, BUSY, CANCEL, ...
Thank you again and have a nice day :)
Mike
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[asterisk-users] agi dial termination cause ?
Hi All, in an AGI script, if executing the Asterisk command Dial, I only get result => -1 (if the call has been answered by the callee) and result => 0 (for everything else) Question: how can I know if the call was not answered because of timeout or because the callee was busy ? (I'm using Asterisk 1.8) Thank you very much for supporting, regards and have a nice day. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] minimum card for dahdi timing source ?
On Tue, 05 Oct 2010 17:30:49 +0100 Paul Hayes wrote: > On 02/10/10 17:24, mancyb...@gmail.com wrote: > > Hi All, > > > > for a vicidial server which uses only voip, > > which is the minimum telephony card which would provide the required clock > > timing source for conferences to work properly ? > > > > Maybe the Digium TDM410PLF card > > without any daughter card > > would do the job ? > > > > > > Thank you very much for supporting. > > > > Have a nice week-end, > > Mike > > The cheapest device I've seen to provide a hardware timing source is the > USB "voice sync tool" from Sangoma: > > http://www.sangoma.com/products/hardware_products/specialty_tools.html > > I know of at least one person using this with Vicidial successfully. > > cheers, > Paul. Hi Paul, very interesting thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] minimum card for dahdi timing source ?
Good news, very well. Thank you very much and have a nice day, Mike On Sat, 02 Oct 2010 11:38:49 -0500 Shaun Ruffell wrote: > On 10/2/10 11:24 AM, mancyb...@gmail.com wrote: > > for a vicidial server which uses only voip, which is the minimum > > telephony card which would provide the required clock timing source > > for conferences to work properly ? > > My recommendation would be to use DAHDI 2.4.0 Just having DAHDI loaded > is enough to provide timing / mix conferences without any other > configuration (i.e., no need to load dahdi_dummy). If your server can > keep accurate wall time, then it will be able to provide adequate timing > / mixing for VOIP. > > -- > Shaun Ruffell > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] minimum card for dahdi timing source ?
Hi All, for a vicidial server which uses only voip, which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ? Maybe the Digium TDM410PLF card without any daughter card would do the job ? Thank you very much for supporting. Have a nice week-end, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] vegastream 50 BRI-s latest firmware ?
Hi All, sorry for the off topic. I own 3 vegastream 50 BRI-s 4 ISDN ports gateways with firmware version 6 and I need some advanced parameters available only from firmware version 7. I am sure that I need those parameters because changing the vega gateway with a 20$ cologne pci card in an Asterisk box results in a correct call setup. The vendor refuses to provide the firmware because the product is discontinued, also begging at the phone gave the same result: they kindly suggest to replace the units with newer ones, at 900USD each. I purchased them in the 2004. Question: maybe someone happens to have the latest firmware file and may share it with me ? Thanks and regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup after n seconds using originate ?
Thanks for the comments, this did the trick :)
On Thu, 22 Apr 2010 13:51:35 -0700
Jim Dickenson wrote:
> One way to do what you want is to create an extension and then in your
> originate action use a local change with that extension.
>
> Action: Originate
> Channel: Local/allow_caller_id:415111:541222:3...@context
> Exten: do_echo
> Context: cfmc_cdi_private
> Priority: 1
> Variable: CfMC_ActionID=AllowCallerID
> ActionID: AllowCallerID
> Async: true
>
>
> exten => _allow_caller_id.,1,Verbose(1,allow_caller_id gets ${EXTEN})
> exten => _allow_caller_id.,n,Set(MyCallerID=${CUT(EXTEN,:,2)})
> exten => _allow_caller_id.,n,GotoIf($[${LEN(${MyCallerID})}<10]?NoCID)
> exten => _allow_caller_id.,n,Set(CALLERID(num)=${MyCallerID})
> exten => _allow_caller_id.,n(NoCID),Set(MyNumber=${CUT(EXTEN,:,3)})
> exten => _allow_caller_id.,n,Set(MyTime=${CUT(EXTEN,:,4)})
> exten => _allow_caller_id.,n,Verbose(1,${OutBoundDev} calling ${MyNumber} for
> ${MyTime} seconds)
> exten => _allow_caller_id.,n,Set(CALLERPRES()=allowed_not_screened)
> exten => _allow_caller_id.,n,Dial${OutBoundDev}/${MyNumber},${MyTime},g)
> exten => _allow_caller_id.,n,Verbose(1,${OutBoundDev} call just got status
> ${DIALSTATUS})
> exten => _allow_caller_id.,n,Hangup()
>
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com
>
> CfMC
> http://www.cfmc.com/
>
>
>
> On Apr 22, 2010, at 1:31 PM, mancyb...@gmail.com wrote:
>
> > On Thu, 22 Apr 2010 15:58:34 -0400
> > Ryan Bullock wrote:
> >
> >> Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
> >> that when creating the originate command?
> >>
> >> I don't know if it works, but it is worth a shot.
> >
> > Hi Ryan, thanks for your comment.
> >
> > Unfortunately the 'Variable' parameter is used to push data between the
> > originating script and the dialplan, not commands.
> > Example:
> > Variable: var1=23|var2=24|var3=25
> >
> > Additionally, this data can be used in the dialplan only when the call gets
> > answered or when it fails.
> > I can't find a way to inject the parameter DURING (or before) the call.
> >
> >
> > Thank you very much for supporting,
> > Mike
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
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>http://www.asterisk.org/hello
>
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Re: [asterisk-users] Hangup after n seconds using originate ?
On Thu, 22 Apr 2010 15:58:34 -0400 Ryan Bullock wrote: > Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like > that when creating the originate command? > > I don't know if it works, but it is worth a shot. Hi Ryan, thanks for your comment. Unfortunately the 'Variable' parameter is used to push data between the originating script and the dialplan, not commands. Example: Variable: var1=23|var2=24|var3=25 Additionally, this data can be used in the dialplan only when the call gets answered or when it fails. I can't find a way to inject the parameter DURING (or before) the call. Thank you very much for supporting, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup after n seconds using originate ?
Hi All, I would like to know if you can confirm that, if using origination via AMI, as documented here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate it is not possible to set the max duration of a call. I mean: what you would do with the L (limit) parameter of the command Dial, is not possible when originating. As well as using the absolute timeout, as documented here: http://www.asteriskguru.com/tutorials/timeoutabsolute_function.html can't be done when originating. Is this true ? I'm using version 1.4. Thanks for supporting, have a nice day. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup after 1 second of ringing ?
Hi All, does the Asterisk's 'Dial' command have some hooks to execute commands as soon as the 'ringing' signal is received ? For example: can a call be dropped 1 second after the called party's phone started to ring ? I'm using version 1.4. Thanks for supporting, have a nice day. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lower kernel version for mISDN
Hi All, Sunday question: does mISDN work on kernel 2.4 ? Thanks and have a nice day, Mike -- _ -- 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] Encrypted calls between mobile gsm and isdn (asterisk)
Hi All, are you aware of any solution which can encrypt calls between a mobile gsm and isdn (asterisk) ? Thanks for your attention, have a nice day. Mike -- _ -- 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] Optimization of call from server 1 to 2 and thenback to 1
Hi Danny, sorry you are correct: > Difficult to say since you don't say if you are on 1.2, 1.4 or 1.6 both Asterisk are running version 1.4.21.2 -- _ -- 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] Optimization of call from server 1 to 2 and then back to 1
Hi All, suppose this call flow: there are two Asterisk servers, they are connected through a IAX2 trunk. The users use SIP. The user A on the Asterisk server 1 calls the user B on the Asterisk server 2. They talk for a while and then the user B does an attendant transfer to the user C on the Asterisk server 1. Question: is it possible to optimize the voice flow or the music on hold flow so that it is done inside the Asterisk server 1 instead of forward and back: from server 1 to 2 and then back to 1 ? Thanks for your attention and for supporting, have a nice day. Mike -- _ -- 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] Digium hardware support ?
On Mon, 7 Sep 2009 08:48:25 -0500 "Juan Cardoza" wrote: > Hello > > What is your Asterisk problem?, may be I can help you... > I had configure a T1 Card TE121 connected with and AVAYA PBX > Best regards Hi Juan, thanks for your help. I'm going to choose a 4 ports PRI digium card for this server: http://h10010.www1.hp.com/wwpc/us/en/sm/WF06b/15351-15351-241434-241646-241477-1121586-3638086-3638087.html which specs are here: http://h18004.www1.hp.com/products/quickspecs/12475_na/12475_na.html and I read that the slots are: One 64-bit/133-MHz PCI-X; two 64-bit/100-MHz PCI-X; three x8 PCI Express (x4 speed) so, since the digium PCI-E card is x1, it does not fit in the x8 but, the digium PCI cards, this one: http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE407PF-T1/E1-PRI-PCI-5-0V-HW-EC.html or this one: http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE412PF-T1/E1-PRI-PCI-3-3V.html should fit in the PCI-X slot, since PCI-X has backward compatibility toward older PCI. But I still don't understand if the PCI-X slots of the ML350 is 3.3V or 5.0V OR if it has autosense (supports both). Thanks and have a nice day. ___ -- 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] SUN and PRI ?
On Mon, 7 Sep 2009 02:43:54 +0100 Ex Vito wrote: > The system specs mention PCIe expansion slots, so your only > option is the TE420B. > > -- > exvito Hi Ex Vito, shouldn't the card be low profile ? Thanks and have a nice day. ___ -- 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] Digium hardware support ?
Hi All, does Digium provide a service support for a compatibility question about their PRI hardware ? Thanks and have a nice day. ___ -- 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] SUN and PRI ?
Hi All, on this hardware: http://www.sun.com/servers/x64/x2200/specs.xml would one of the following 4 ports PRI cards be ok ? http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE420BF-T1/E1-PRI-PCIe-HW-EC.html http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE407PF-T1/E1-PRI-PCI-5-0V-HW-EC.html http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE412PF-T1/E1-PRI-PCI-3-3V.html http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE420BF-T1/E1-PRI-PCIe-HW-EC.html it should handle 90 channels over 3 PRI lines (30 channels each) and have echo cancellation. Thanks and have a nice day. ___ -- 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] Timeout func ignored if inside a macro and when Dial cmd has limit (L). Bug ?
Hi All,
suppose this:
Dial(SIP///60/L(360)M(td|${EPOCH})
where 60 is the seconds to wait for the callee (the called party) to answer
L(360) is the absolute limit of the call once it has been answered, in ms
M(td|${EPOCH}) is the macro to execute when the call gets answered. ${EPOCH}
contains the current unixtime.
That's the macro:
[macro-td]
exten => s,1,Set(myDiff=${MATH(${EPOCH}-${ARG1},i)})
exten => s,n,NoOp(${myDiff})
exten => s,n,GotoIf($[${myDiff} < 4]?hu:he)
exten => s,n(hu),Set(TIMEOUT(absolute)=6)
exten => s,n(he),NoOp(${myDiff})
where:
exten => s,1,Set(myDiff=${MATH(${EPOCH}-${ARG1},i)})
sets the variable 'myDiff' to the difference, in seconds, between the start of
the call and when it was answered.
exten => s,n,NoOp(${myDiff})
prints the variable 'myDiff'
exten => s,n,GotoIf($[${myDiff} < 4]?hu:he)
will invoke the 4th line of the macro if the call was answered in less than 4
seconds
else it will invoke the 5th line of the macro
exten => s,n(hu),Set(TIMEOUT(absolute)=6)
should set the absolute timeout of the call from 'now' to 6 seconds
exten => s,n(he),NoOp(${myDiff})
just prints again the 'myDiff'
Problem:
the Set(TIMEOUT(absolute)=6) function gets triggered if the call has been
answered in less than 4 seconds,
the Asterisk console reports the correct hangup time prediction with a message
like:
"Channel will hangup at " ...
but the call doesn't hangup.
If the L (limit) modifier of the Dial cmd is not used, the call hangups
correctly.
Asterisk version: 1.4.26
Thanks for supporting,
have a nice day.
Mancy
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