Re: [asterisk-users] no ringing tone with Dial option r
On 11/03/2015 01:11 PM, John Kiniston wrote:
Have you checked your indications.conf? I've seen a missing or
misconfiguration in the zone definition cause this.
On Tue, Nov 3, 2015 at 11:07 AM, sean darcy mailto:seandar...@gmail.com>> wrote:
On 11/01/2015 12:38 PM, sean darcy wrote:
I'm not getting any ringing when I use option r with Dial:
Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com
,,rTt") in
new stack
Otherwise all works. The call goes through, good audio.
sean
FWIW, 11.18.0 on Fedora 22.
sean
AFAIK, I've never touched indications.conf . Not even sure what zone
definitions are in indications.
Also, now on 11.20.0. Same problem.
sean
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Re: [asterisk-users] no ringing tone with Dial option r
On 11/04/2015 03:40 AM, A J Stiles wrote:
On Tuesday 03 Nov 2015, sean darcy wrote:
On 11/01/2015 12:38 PM, sean darcy wrote:
I'm not getting any ringing when I use option r with Dial:
Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in
new stack
Otherwise all works. The call goes through, good audio.
sean
FWIW, 11.18.0 on Fedora 22.
sean
Make sure you have an Answer(), or some command that does an implicit
Answer(), somewhere in the dialplan before the Dial() statement with the r
option. Been bitten that way before .
Me too. I put it as the first command.
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [@internal:1] Answer("DAHDI/1-1", "") in new stack
..
-- Executing [s@DialOut:15] Dial("DAHDI/1-1",
"motif/8447/+1@voice.google.com,,Ttr") in new stack
Maybe I need to put another Answer() in the DialOut context. Would 2
Answer() cause a problem?
sean
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Re: [asterisk-users] no ringing tone with Dial option r
On 11/04/2015 03:43 AM, Bertrand LUPART - Linkeo.com wrote:
Hello,
I'm not getting any ringing when I use option r with Dial:
Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new
stack
Warning, options are the 3rd arguments.
You seem to have an extra comma and a non-closed double-quote.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Here's the actual dialplan command:
exten =>
s,n(gv),Dial(motif/${MOTIF_DEFAULT}/+1${ARG1}@voice.google.com,,rTt)
No quotes. And the options are the 3rd argument, I think.
sean
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Re: [asterisk-users] issue with bridgeConference
> On Mon, Nov 2, 2015 at 3:16 PM, hadi wrote:
> > I have configure bridgeConference. But im having some issue. I want to
> > give the ability to the user when dialing from the Conference to
> > hangup the call by sending dtmf tones without being hangup from the
> > Conference. For example if the user call some person and that person
> > not answering, the user has the ability to hangup the call by sending
> > *9 and return back the Conference, and start calling again.
> >
> > Here is my dial plan:-
> >
> > exten => 200,1,Dial(SIP/200,,Hhg)
> > exten => 200,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Hangup
> > exten => s-CONGESTION,1,Congestion exten => s-CANCEL,1, Busy exten
> =>
> > s-BUSY,1,Busy exten => s-CHANUNAVAIL,1,Playback(switchoff)
> > exten => s-CHANUNAVAIL,n,Read(number,,,sn) exten =>
> > s-CHANUNAVAIL,n,GotoIf($["${number}" = "9"]?106) exten =>
> > s-CHANUNAVAIL,106,SoftHangup(${EXTEN})
>
> I suppose by bridgeConference you mean ConfBridge?
>
> If you require assistance you'll need to describe more than what you *want
> to do*. You'll need to describe the issue you are having.
> Include dialplan and logs to demonstrate the issue.
Hi Rusty,
I solved my problem. Thank you for your support.
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Re: [asterisk-users] Find me macro - calling multiple people to get a hold of one
- Original Message -
> From: "Wiebe Cazemier"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, 6 November, 2015 10:54:48 AM
> Subject: Re: [asterisk-users] Find me macro - calling multiple people to get
> ahold of one
>
> It appears that [1] lucked out in that it may sometimes work. WaitExten()
> doesn't work in a macro and has no option to supply the context it runs in.
> The '1' extension defined there is never called.
>
> You can use Background() instead, but you have to play something that is long
> enough so people have time to press a button; then you don't need
> WaitExten(). Background() has documented behavior with regards to macros. It
> calls extensions in the calling context, except if you give the context as
> option:
>
> exten =>
>
> s,n,Background(emergency&lines-complaining-customers&press-1&press-1&press-1&press-1&press-1&press-1&press-1,,,macro-screen)
>
> But then still, more problems. The 1 and 4 extensions work, but 2 doesn't; it
> hangs up. I really don't get that.
>
> =
> exten => 1,1,Verbose(0,"1 pressed")
> exten => 1,n,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to
> connect the caller
>
>
> exten => 4,1,Verbose(0,"4 pressed")
> exten => 4,n,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to
> connect the caller
>
> ; This doesn't seem to work. It goes to hangup, in the default context
> exten => 2,1,Verbose(0,"2")
> exten => 2,n,Set(MACRO_RESULT=CONTINUE)
> =
>
> And I still have the problem that if I hang up instead of pressing 1, the
> caller gets connected to that disconnected line. There doesn't seem to be
> any extension I can put in my macro that responds to the hangup. Not e
> (exception), i (invalid), h (hangup), etc.
>
> Normally, Background() restarts the current context with the result on the
> stack. But when you hang up, it just SEEMS to return with a non-zero exit
> code, yet not continue with the 's' extension. Is it possible to make it
> continue with the dial plan?
I got it working. I set the MACRO_RESULT to CONTINUE in the start extension,
and then set it to something else when I don't want to connect the caller:
[globals]
WIEBE_MOBILE = 10digitnr
ANOTHER_MOBILE = 10digitnr
[default]
exten => _XX,1,Dial(SIP/sip2phoneprovider/${EXTEN:4},40,M(screen))
exten => _XX,2,Hangup
[macro-screen]
exten => s,1,Set(MACRO_RESULT=CONTINUE) ; default action -> not connecting
exten => s,n,Wait(2)
exten =>
s,n,Background(emergency&lines-complaining-customers&press-1&press-1&press-1&press-1&press-1&press-1&press-1&press-1&press-1&press-1&press-1&press-1&press-1,,,macro-screen)
exten => 1,1,Verbose(0,"1 pressed")
exten => 1,n,Set(MACRO_RESULT=SANTA)
[findme]
exten => s,1,Set(CALLERID(all)="Bla" <123456789>)
same => n,Playback(please-wait-connect-oncall-eng)
same => n,Dial(LOCAL/${WIEBE_MOBILE}&LOCAL/${ANOTHER_MOBILE})
same => n,Playback(vm-nobodyavail)
exten => t,1,Playback(vm-nobodyavail)
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Re: [asterisk-users] PJSIP with registratrion to DNS SRV records fail with PJLIB_UTIL_EDNSNOANSWERREC
On Wed, Nov 4, 2015 at 9:19 AM, Daniel Tryba wrote: > > I finally thought it might be a good time to start looking at the pjsip > implementation in Asterisk 13. But trying to register to a sip cluster > that uses SRV records fails randomly with: > > [Nov 4 15:50:59] WARNING[31330]: pjsip:0 : tsx0x7f075c006 Failed to > send Request msg REGISTER/cseq=17800 (tdta0x7f075c0058f0)! err=320047 > (No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC)) > [Nov 4 15:50:59] WARNING[31330]: res_pjsip_outbound_registration.c:735 > schedule_retry: No response received from 'sip:sip.itco.nl' on > registration attempt to 'sip:tr...@sip.itco.nl', retrying in '60' > > [Nov 4 15:51:59] WARNING[31330]: pjsip:0 : tsx0x7f075c006 Failed to > send Request msg REGISTER/cseq=17801 (tdta0x7f075c0058f0)! err=320047 > (No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC)) > [Nov 4 15:51:59] WARNING[31330]: res_pjsip_outbound_registration.c:735 > schedule_retry: No response received from 'sip:sip.itco.nl' on > registration attempt to 'sip:tr...@sip.itco.nl', retrying in '60' > > At 15:52:59 the register succeeds somehow. > > Attached is a pcap of the DNS request and the responses (capture filter: > port 53 or port 5060 or port 5061). Unlike the warning says the > responses are there. > > Does anybody have a hint of what is going on/what I do wrong? > > pjsip.conf: > [transport-udp] > type=transport > protocol=udp > bind=0.0.0.0 > > [transport-tcp] > type=transport > protocol=tcp > bind=0.0.0.0 > > [tryba] > type=endpoint > transport=transport-udp > context=tryba > disallow=all > allow=alaw > outbound_auth=tryba_auth > force_rport=yes > direct_media=no > ice_support=yes > auth=tryba_auth > > [tryba_auth] > type=auth > auth_type=userpass > password=** > username=tryba > > [tryba_register] > transport=transport-udp > type=registration > server_uri=sip:sip.itco.nl > client_uri=sip:tr...@sip.itco.nl > contact_user=tryba > outbound_auth=tryba_auth > expiration=180 > For those wandering web-searching souls: https://issues.asterisk.org/jira/browse/ASTERISK-25528 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://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
Re: [asterisk-users] issue with bridgeConference
On Mon, Nov 2, 2015 at 3:16 PM, hadi wrote:
> I have configure bridgeConference. But im having some issue. I want to give
> the ability to the user when dialing from the Conference to hangup the call
> by sending dtmf tones without being hangup from the Conference. For example
> if the user call some person and that person not answering, the user has the
> ability to hangup the call by sending *9 and return back the Conference, and
> start calling again.
>
> Here is my dial plan:-
>
> exten => 200,1,Dial(SIP/200,,Hhg)
> exten => 200,n,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Hangup
> exten => s-CONGESTION,1,Congestion
> exten => s-CANCEL,1, Busy
> exten => s-BUSY,1,Busy
> exten => s-CHANUNAVAIL,1,Playback(switchoff)
> exten => s-CHANUNAVAIL,n,Read(number,,,sn)
> exten => s-CHANUNAVAIL,n,GotoIf($["${number}" = "9"]?106)
> exten => s-CHANUNAVAIL,106,SoftHangup(${EXTEN})
I suppose by bridgeConference you mean ConfBridge?
If you require assistance you'll need to describe more than what you
*want to do*. You'll need to describe the issue you are having.
Include dialplan and logs to demonstrate the issue.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] How to encode plus sign in REGEX function in dialplan?
On Thu, Nov 5, 2015 at 12:49 AM, Recursive wrote:
> Dear all,
>
> I have made a fairly complex dialplan where I am using the REGEX function in
> many places. This works so far, but I wasn't able to solve the following
> problem. What I would like to do is the following (please note that this is
> normal regex syntax and obviously not what the REGEX function expects, but I
> hope it shows the idea):
>
> same => n(A1), GotoIf($[${REGEX("^\+49.*" ${EXTEN})}]?:A2)
>
> This line should make Asterisk jump to label A2 if the extension begins with
> +49. Since the plus sign is a special char in regexes, I have escaped it with
> \ as usual. But that does not work; the pattern is not matched and the goto
> is not executed when the extension begins with +49.
>
> What I already have tried:
>
> 1) same => n(A1), GotoIf($[${REGEX("^\\+49.*" ${EXTEN})}]?:A2)
>
> 2) same => n(A1), GotoIf($[${REGEX("^\\\+49.*" ${EXTEN})}]?:A2)
>
> 3) same => n(A1), GotoIf($[${REGEX("^+49.*" ${EXTEN})}]?:A2)
>
> 4) same => n, Set(REPAT=^+49.*)
>same => n(A1), GotoIf($[${REGEX(${REPAT} ${EXTEN})}]?:A2)
>
> 5) same => n, Set(REPAT="^+49.*")
>same => n(A1), GotoIf($[${REGEX(${REPAT} ${EXTEN})}]?:A2)
>
> 6) same => n, Set(REPAT=^+49.*)
>same => n(A1), GotoIf($[${REGEX("${REPAT}" ${EXTEN})}]?:A2)
>
> 7) same => n, Set(REPAT="^+49.*")
>same => n(A1), GotoIf($[${REGEX("${REPAT}" ${EXTEN})}]?:A2)
>
> Neither of these worked.
>
> Actually, the REGEX function is not able to handle normal regular
> expressions. To make things worse, there doesn't seem to be any
> documentation. Could anybody please point me to documentation or tell me how
> write that very simple pattern?
>
> Thank you very much,
>
> Recursive
>
> P.S. This happens in Asterisk 13.6.0 - I haven't tested with other versions.
The documentation for that function is available at the CLI "core show
function REGEX" and is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_REGEX
It should be able to handle typical regular expression. I don't see
anything wrong with what you are doing. Please file a bug at
issues.asterisk.org/jira. Do include a debug log on the issue
captured when Asterisk attempts to execute these extensions.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Thanks,
--
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Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] Find me macro - calling multiple people to get a hold of one
- Original Message -
> From: "Wiebe Cazemier"
> To: asterisk-users@lists.digium.com
> Sent: Wednesday, 4 November, 2015 5:24:38 PM
> Subject: [asterisk-users] Find me macro - calling multiple people to get a
> hold of one
>
> Hi list,
>
> We're trying to set up a phone number that customers can call to get a hold
> of anyone of a group of sysadmins (and not their voice mails!). We found the
> findme example ([1]) that makes the callees press 1 to accept the call. It
> almost works, but it doesn't work correctly when one of the callees, the
> sysadmins, hangs up after accepting the call.
>
> We're using this 'screen' macro:
>
> ==
> [default]
> exten => _XX,1,Dial(SIP/bla/${EXTEN:4},40,M(screen))
> exten => _XX,2,Hangup
>
> [macro-screen]
> exten => s,1,Wait(1)
> exten => s,n,Background(press-1)
> exten => s,n,WaitExten(10) ; the value is the Wait time before we assume the
> call is not accepted
> exten => 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to
> connect the caller
> exten => t,1,Playback(weasels-eaten-phonesys) ; if you're too late with
> pressing 1
> exten => t,n,Set(MACRO_RESULT=CONTINUE)
>
> [findme]
> exten => s,1,Set(CALLERID(all)="Alarm" <911>)
> same => n,Playback(please-wait-connect-oncall-eng)
> same => n,Dial(LOCAL/${WIEBE_MOBILE})
> same => n,Playback(vm-nobodyavail)
> exten => t,1,Playback(vm-nobodyavail)
> =
>
> First of all, what is MACRO_RESULT? I can't seem to find anything about that.
> Googling for it yields basically nothing.
>
> But the biggest problem is when the callee answers, then hangs up. The person
> calling is connected to the phone that hangs up, instead of hearing
> 'vm-nobodyavail'. This seems to be because there is nothing that sets
> MACRO_RESULT in that event (it's only set on 't', timeout).
>
> I tried adding:
>
> exten => h,1,Verbose(0,"The callee hung up")
> exten => h,n,Set(MACRO_RESULT=CONTINUE)
>
> to handle the hangup (h), but it's not doing that.
>
> WaitExten() pushes the result back on the stack and restarts the context,
> right? So what is the result when the person hangs up?
>
> Regards,
>
> Wiebe
>
>
>
> [1] http://www.voip-info.org/wiki/view/Asterisk+tips+findme
>
It appears that [1] lucked out in that it may sometimes work. WaitExten()
doesn't work in a macro and has no option to supply the context it runs in. The
'1' extension defined there is never called.
You can use Background() instead, but you have to play something that is long
enough so people have time to press a button; then you don't need WaitExten().
Background() has documented behavior with regards to macros. It calls
extensions in the calling context, except if you give the context as option:
exten =>
s,n,Background(emergency&lines-complaining-customers&press-1&press-1&press-1&press-1&press-1&press-1&press-1,,,macro-screen)
But then still, more problems. The 1 and 4 extensions work, but 2 doesn't; it
hangs up. I really don't get that.
=
exten => 1,1,Verbose(0,"1 pressed")
exten => 1,n,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to
connect the caller
exten => 4,1,Verbose(0,"4 pressed")
exten => 4,n,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to
connect the caller
; This doesn't seem to work. It goes to hangup, in the default context
exten => 2,1,Verbose(0,"2")
exten => 2,n,Set(MACRO_RESULT=CONTINUE)
=
And I still have the problem that if I hang up instead of pressing 1, the
caller gets connected to that disconnected line. There doesn't seem to be any
extension I can put in my macro that responds to the hangup. Not e (exception),
i (invalid), h (hangup), etc.
Normally, Background() restarts the current context with the result on the
stack. But when you hang up, it just SEEMS to return with a non-zero exit code,
yet not continue with the 's' extension. Is it possible to make it continue
with the dial plan?
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[asterisk-users] bad performance centos6 ->centos7
hi, i'm evaluating performance of centos7 i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 with 500calls (7sec alaw, simple dialplan, pass trough - sipp generators/asterisk receiver with answer/playback) scenario - sipp generators - asterisk - asterisk receiver (i wrote ansible scenario for this if you are interested) then i reinstalled system to centos7 x86_64/distro kernel 3.10, asterisk 11.20.0 and run the test again there is big performance hit https://dl.dropboxusercontent.com/u/44105720/context_interrupts.PNG https://dl.dropboxusercontent.com/u/44105720/cpu.PNG https://dl.dropboxusercontent.com/u/44105720/load.PNG any ideas what tweaks can help? (it looks like the main problem is in interrupts from network card) your experience with centos7? any experience with kernel 4.2 from http://elrepo.org/linux/kernel/el7/x86_64/RPMS/ ? -- --- Marek Cervenka === -- _ -- 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] Re-Invite to Native Bridge
Hello all, My Asterisk is between my ITSP and a SIP phone. I cannot do direct media between the provider and the SIP phone, but I would like Asterisk to locally RTP bridge the two channels using native_rtp. Example: > Bridge cfb56606-6b40-4da7-a6fa-6499e183cdbb: switching from simple_bridge > technology to native_rtp > Locally RTP bridged 'PJSIP/5iwrlBee9oKCMAs-' and > 'PJSIP/ekiga_outbound-0001' in stack > Locally RTP bridged 'PJSIP/5iwrlBee9oKCMAs-' and > 'PJSIP/ekiga_outbound-0001' in stack My SIP phone supports G722 and PCMA, as does the ITSP provider. But depending on the other party in a call, the ITSP may only offer PCMA. So with my current setting (allow=!all,g722,alaw) I run into this situation when the ITSP doesn't offer G722: ITSP <- PCMA -> Asterisk <- G722 -> SIP phone Obviously that's not optimal. Transcoding needs to take place. I'm looking for a way to get Asterisk to re-INVITE/UPDATE the SIP phone, to renegotiate the codec, to avoid transcoding. I searched the web and found that there are pre-bridge handlers. But they are executed on the called party channel. So that wouldn't help when doing an outbound call, plus I'm not sure how I could extract enough information from the other channel to know if a reinvite is needed (e.g. the codec the other channel is using). I also found the Media Format Rewrite article on the wiki: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite In the General Rules paragraph point 3 states (quote): Prior to entering a bridge, a dialplan function can be used to set whether or not that channel will attempt to make itself compatible with whatever is in the bridge with it. If a channel enters a bridge that has another channel in it with a format it supports, it will attempt to switch the channel to the bridged channel's format to facilitate native bridging. Note that this has no bearing in multi-party bridges, where everyone is transcoded. There's another paragraph called "Re-Invite to Native Bridge" with an example where Alice and Bob have a differently ordered set of codecs and Alice's channel is set to re-INVITE back to native bridging if possible. After the re-INVITE Asterisk switches to a native bridge. Well, that is exactly what I want :) I installed Asterisk 13.6.0 hoping that I could get it to do this. But until now I haven't found out how. Does anybody know if this feature from the Media Format Rewrite article is already available? Kind regards, Sebastian -- _ -- 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
