Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer
Hi,
This is still broken in 1.6.2.20. Please see below.
On Fri, May 06, 2011 at 04:27:42PM +, Watkins, Bradley wrote:
> >From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> >boun...@lists.digium.com] On Behalf Of Mark G Thomas
> >Sent: Friday, May 06, 2011 12:02 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U()
> >option in 1.6.2.17.2 and newer
> >
> >Hi,
> >
> >On Thu, May 05, 2011 at 05:30:04PM -0400, Paul Belanger wrote:
> >> On 11-05-05 05:14 PM, Mark G Thomas wrote:
> >> >Hi,
> >> >
> >> >I think this must be a bug introduced with 1.6.2.17.something.
> >> >
> >> >When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or
> >> >1.6.2.18, my AEL Dial() commands with the "U" options fail with the
> >following error:
> >> >
> >> >[May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-
> >existent
> >> > destination for gosub: (Context:screen, Extension:s, Priority:1)
> >> >
> >> You might want to have a look at:
> >> https://issues.asterisk.org/view.php?id=18910
> >
> >Thanks. This is it.
> >
> >If I'm reading this right, it describes the change which broke things for me,
> >but no solution applicable to my Dial() command U flag, which is invoking my
> >AEL GoSub (Macro). Switching the Dials back to the M flag doesn't fix it
> >either.
> >
> >It sure seems to me this change to AEL has had unexpected consequences in
> >terms of breaking things in dialplans.
> >
>
> I was under the impression that this had been fixed, although perhaps it's
> not yet in a release. Is there a chance you try with the latest 1.6.2 branch
> from SVN?
>
> - Brad
In AEL, Dial() with the U flag is still broken.
Reverting to a pre-1.6.2.17.2 pval.c fixes the problem.
[Aug 8 13:36:01] ERROR[24608]: app_stack.c:402 gosub_exec: Attempt to reach a
non-existent destination for gosub: (Context:screen, Extension:s, Priority:1)
My AEL dialplan segment:
--
context internals {
102 => {
Dial(${MARKCELL},30,tgU(screen));
jump s@home-menu;
};
};
macro screen() {
Wait(0.5);
Read(ACCEPT,followme/options,1,,1,20);
if( "${ACCEPT}" = "1" ) {
Background(connecting);
} else {
Set(GOSUB_RESULT=CONTINUE);
};
return;
};
Here's the dialplan it created under 1.6.2.20, but U(screen)
in Dial() tries to call screen,s which doesn't exist.
---
[ Context 'internals' created by 'pbx_ael' ]
'102' => 1. Dial(${MARKCELL},30,tgU(screen)) [pbx_ael]
2. Goto(home-menu,s,1)[pbx_ael]
[ Context 'screen' created by 'pbx_ael' ]
'~~s~~' =>1. Wait(0.5) [pbx_ael]
2. Read(ACCEPT,followme/options,1,,1,20) [pbx_ael]
3. GotoIf($[ "${ACCEPT}" = "1" ]?4:6) [pbx_ael]
4. Background(connecting) [pbx_ael]
5. Goto(7)[pbx_ael]
6. Set(GOSUB_RESULT=CONTINUE) [pbx_ael]
7. NoOp(Finish if_screen_25) [pbx_ael]
8. Return() [pbx_ael]
This works, from an earlier version, before the pval.c change:
[ Context 'screen' created by 'pbx_ael' ]
's' =>1. Wait(0.5) [pbx_ael]
2. Read(ACCEPT,followme/options,1,,1,20) [pbx_ael]
3. GotoIf($[ "${ACCEPT}" = "1" ]?4:6) [pbx_ael]
4. Background(connecting) [pbx_ael]
5. Goto(7)[pbx_ael]
6. Set(GOSUB_RESULT=CONTINUE) [pbx_ael]
7. NoOp(Finish if_screen_25) [pbx_ael]
8. Return() [pbx_ael]
--
Mark G. Thomas (m...@misty.com)
Web: http://mgtinternet.com/
Tel: +1-215-512-0112 US: 877-512-0112
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Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer
Hi, On Thu, May 05, 2011 at 05:30:04PM -0400, Paul Belanger wrote: > On 11-05-05 05:14 PM, Mark G Thomas wrote: > >Hi, > > > >I think this must be a bug introduced with 1.6.2.17.something. > > > >When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, > >my AEL Dial() commands with the "U" options fail with the following error: > > > >[May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent > > destination for gosub: (Context:screen, Extension:s, Priority:1) > > > You might want to have a look at: > https://issues.asterisk.org/view.php?id=18910 Thanks. This is it. If I'm reading this right, it describes the change which broke things for me, but no solution applicable to my Dial() command U flag, which is invoking my AEL GoSub (Macro). Switching the Dials back to the M flag doesn't fix it either. It sure seems to me this change to AEL has had unexpected consequences in terms of breaking things in dialplans. Mark -- Mark G. Thomas (m...@misty.com) Web: http://mgtinternet.com/ Tel: +1-215-512-0112 US: 877-512-0112 -- _ -- 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] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer
Hi,
I think this must be a bug introduced with 1.6.2.17.something.
When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18,
my AEL Dial() commands with the "U" options fail with the following error:
[May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent
destination for gosub: (Context:screen, Extension:s, Priority:1)
Here are the segments of the ael:
---
...
Dial(${MARKCELL},30,tgU(screen));
...
macro screen() {
Wait(0.5);
Read(ACCEPT,followme/options,1,,1,20);
if( "${ACCEPT}" = "1" ) {
Background(connecting);
} else {
Set(GOSUB_RESULT=CONTINUE);
};
return;
};
And, here is the dialplan created from my above ael. The change is obvious,
but I'm not sure if there is some way I can fix this in my AEL, or if this
is just a bug that needs to be fixed.
asterisk-1.6.2.17.2 and asterisk-1.6.2.18:
---
[ Context 'screen' created by 'pbx_ael' ]
'~~s~~' =>1. Wait(0.5) [pbx_ael]
2. Read(ACCEPT,followme/options,1,,1,20) [pbx_ael]
3. GotoIf($[ "${ACCEPT}" = "1" ]?4:6) [pbx_ael]
4. Background(connecting) [pbx_ael]
5. Goto(7)[pbx_ael]
6. Set(GOSUB_RESULT=CONTINUE) [pbx_ael]
7. NoOp(Finish if_screen_139) [pbx_ael]
8. Return() [pbx_ael]
asterisk-1.6.2.16.1:
---
[ Context 'screen' created by 'pbx_ael' ]
's' =>1. Wait(0.5) [pbx_ael]
2. Read(ACCEPT,followme/options,1,,1,20) [pbx_ael]
3. GotoIf($[ "${ACCEPT}" = "1" ]?4:6) [pbx_ael]
4. Background(connecting) [pbx_ael]
5. Goto(7)[pbx_ael]
6. Set(GOSUB_RESULT=CONTINUE) [pbx_ael]
7. NoOp(Finish if_screen_28) [pbx_ael]
8. Return() [pbx_ael]
--
Mark G. Thomas (m...@misty.com)
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Re: [asterisk-users] Hang using Festival application
Hi Brian, On Wed, Mar 23, 2011 at 03:15:08PM -0400, Brian Henning wrote: > Hello, > > Suppose a dialplan such as: > > exten => 6004,1,Answer > exten => 6004,n,Wait(1) > exten => 6004,n,SayDigits(1) > exten => 6004,n,Festival(This is a test of Festival) > exten => 6004,n,Hangup ... > ... and nothing more. Nothing happens after " == Parsing ...", and the SIP > channel gets stuck open even after I physically hang up the extension (will > not respond to a hangup request, can only be eliminated by restarting > asterisk). I hear "one" in the phone and then silence. ... > These are all unmodified packages obtained via aptitude. > > What am I getting wrong? Your problem might be that you either need to patch Festival or modify the Festival configuration file: http://www.voip-info.org/wiki/view/Asterisk+festival+installation Mark -- Mark G. Thomas (m...@misty.com) Web: http://mgtinternet.com/ Tel: +1-215-512-0112 US: 877-512-0112 -- _ -- 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] Asterisk not working with Festival
Hi,
I'm experiencing the same problem, with identical symptoms.
I also noticed that after making a call attempt, I see this stuck TCP
connection pair until I stop and restart the asterisk server process.
# netstat -an | grep 1314
tcp0 0 0.0.0.0:13140.0.0.0:*
LISTEN
tcp 46 0 127.0.0.1:52206 127.0.0.1:1314
CLOSE_WAIT
tcp0 0 127.0.0.1:1314 127.0.0.1:52206
FIN_WAIT2
Mark
On Thu, Aug 12, 2010 at 02:41:50PM +0530, Davinder Kumar Meen wrote:
>I tried it but I still cannot hear any sound created from Festival()
>function. I can hear only a voice saying one which was working earlier
>as well. Here is log of asterisk console:
> -- Attempting call on SIP/011xx...@gafachi1a for
>s...@connect-to-me:1 (Retry 1)
>-- Executing [...@connect-to-me:1] Answer("SIP/gafachi1a-",
>"") in new stack
>-- Executing [...@connect-to-me:2] Wait("SIP/gafachi1a-",
>"7") in new stack
>-- Executing [...@connect-to-me:3]
>SayDigits("SIP/gafachi1a-", "'1'") in new stack
>-- Playing 'digits/1.slin' (language 'en')
>-- Executing [...@connect-to-me:4] Festival("SIP/gafachi1a-",
>"hello john") in new stack
> == Parsing '/usr/local/etc/asterisk/festival.conf': == Found
>On 11/08/10 11:22 PM, "Danny Nicholas" wrote:
>
>
> From: asterisk-users-boun...@lists.digium.com
> [[1]mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Davinder Kumar Meen
> Subject: Re: [asterisk-users] Asterisk not working with Festival
> Can anyone help please on this?
>
> >[connect-to-me]
> >exten => s,1,Answer
> >Exten => s,n,SayDigits(`1')
> >exten => s,n,Festival(hello john)
> >exten => s,n,Hangup
>
> When you call in from your mobile, you are using a DAHDI channel
> which introduces a 3-7 second delay into the process, unless you
> have one of the "blessed" phone companies that offers call
> supervision. If you put a wait(7) in front of SayDigits, you should
> hear the call "normally".
> This is what I would suggest
> [connect-to-me]
> exten => s,1,Answer
> Exten => s,n,Gotoif($["${EXTEN}:0:3)" = "SIP"]?4:3
> Exten => s,n,wait(7)
> Exten => s,n,SayDigits(`1')
> exten => s,n,Festival(hello john)
> exten => s,n,Hangup
>
>Thanks,
>Davinder Kumar Meen
>Partner & Project Manager
>Impinge Solutions, F-250, Phase 8B, Mohali (India)
>www.impingesolutions.com
>We also provide server hosting services. Please checkout our website
>www.goforspace.com
>
> References
>
>1. mailto:asterisk-users-boun...@lists.digium.com]
> --
> _________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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--
Mark G. Thomas (m...@misty.com)
Web: http://mgtinternet.com/
Tel: +1-215-512-0112 US: 877-512-0112
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Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2
Hi, On Tue, Aug 03, 2010 at 01:49:11PM -0500, Tilghman Lesher wrote: > On Tuesday 03 August 2010 13:19:11 Mark G. Thomas wrote: > > I can't figure out what syntax to use with the Dial() "M" parameter > > for the AEL parser to interpret properly. Creating an AEL > > macro named "macro-screen()" partly works as a hack, but it must > > not turn into a gosub properly, so I get warnings about the "return;". > > Is there a reason you don't want to use the 'U' option in Dial? It was > created specifically for this purpose. Thank you! I didn't know there was a "U" option. I don't see any mention of it on the voip-info.org wiki or other Dial() documentation, but didn't check for new options in the built in documentation until just now. Mark -- Mark G. Thomas (m...@misty.com) -- _ -- 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] Dial() M parameter in 1.6.2.11-rc2
Hi,
I can't figure out what syntax to use with the Dial() "M" parameter
for the AEL parser to interpret properly. Creating an AEL
macro named "macro-screen()" partly works as a hack, but it must
not turn into a gosub properly, so I get warnings about the "return;".
Dial(...,tgM(&screen)) with the ael macro named "screen" does not work
Dial(...,tgM&screen) with the ael macro named "screen" does not work
Dial(...,tgM(screen)) with the ael macro named "screen" does not work
Dial(...,tgM(screen)) with the ael macro named "macro-screen" partly works
Is a more correct or otherwise better way to do this in AEL? Is there
some other solution? The other followme examples I've found all have
different behaviors than I want. I'm not looking for the caller to
be prompted for their name or anything, and I don't want the followme
connect to happen unless the cellphone user hits a "1" to accept the
call, or other key to ditch the call, otherwise cellphone voicemail
gets the call.
Mark
context inbound {
...
211234 => Dial(SIP/1...@cme&local/1...@internals,18,rt); // screen these
...
};
context internals {
102 => {
Dial(${CELLPHONE},30,tgM(screen)); // cellphone user gets prompted
jump s...@general-menu; // jump to IVR menu if call not accepted
};
};
// play message to cellphone before connecting inbound call
// http://www.voip-info.org/wiki/view/Asterisk+tips+findme
// http://lists.digium.com/pipermail/asterisk-dev/2005-June/013598.html
//
macro macro-screen() {// hack
Wait(0.5);
Read(ACCEPT,followme/options,1,,1,20);
if( "${ACCEPT}" = "1" ) {
Background(connecting);
} else {
Set(MACRO_RESULT=CONTINUE);
};
return; // I get AEL complaints regardless of whether this is here or not.
};
--
Mark G. Thomas (m...@misty.com)
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Re: [asterisk-users] SPA8000 outbound CID problem
Hi, >On 24 June 2010 19:54, Mark G. Thomas <[1]m...@misty.com> wrote: > > Hi, > I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to > both a local Asterisk server and also with a trunk directly to > a VOIP provider. Everything works great, except I'm having a problem > setting the outbound caller ID to a value different from the > SIP username/authname. Perhaps my wording above was unclear. My Asterisk server sends RPID to the VOIP provider just fine. I am having problems getting the SPA8000 to send RPID to the VOIP provider. I realize this is a SPA8000 problem and not an Asterisk problem, but since many people use Asterisk in conjunction with ATA devices like the SPA8000, I was hoping someone here would have either a solution or suggestions. As per below, I can see from my Asterisk server that the SPA8000 is sending a configurable SIP username/authname, and that's being treated as the CID. Since the SPA8000 is at a different office from the Asterisk server, I need to get the SPA8000 RPID directly to the VOIP provider working, I don't want to have to route all the calls through Asterisk to fix the outbound caller ID. > The SPA8000 has SIP setting for Display Name, User ID, Password, > and Auth ID, as well as a "Use Auth ID" checkbox. It's running 6.1.3 > firmware, which looks to be the latest, and supports SIP trunking, > though > even if I don't use trunking, I have the same obstacle if I > configure it > per-line instead of per-trunk. > Inbound CID works fine. When VOIP calls come in via the provider or > Asterisk, the SPA generates CID on it's analog ports. > > The problem is that the outbound caller ID number seems to come from > the SIP "User ID" setting, which is also the SIP authentication > name. > If I instead put the SIP account id into the "Auth ID" field and > check > the "Use Auth ID" box, Asterisk reports: > Registration from 'John Smith <[2]sip:jsm...@our.sip.gateway.com>' > failed for > '1.2.3.4' - Username/auth name mismatch. > Sure, I can overide the CID number on our Asterisk server, but I > don't > have that ability with the VOIP provider's Asterisk server. The > outbound > caller ID always looks like "John Smith " instead of > "John Smith <211212>" no matter how I try to set these fields. > I take it the SIP username and auth name need to match, so that > leaves me > with the question of how to configure a CID number that doesn't > necessarily > match the SIP user/auth name. Is this a limitation of this device, > or > is there some other option I'm overlooking? > Mark On Fri, Jun 25, 2010 at 08:25:02AM +0100, dotnetdub wrote: >Ask your upstream provider if they support remote party ID. IF they do >you can set sendrpid=yes in your sip.conf and set your outbound CID on >an extension or trunk level. >HTH I am not having a problem with Asterisk sending RPID. I know the upstream provider is honoring it, because it works fine with calls from the Asterisk server to the VOIP provider. sendrpid=yes in sip.conf works fine, and the VOIP provider honors it, sending the desired CID on dialed calls. The SPA8000 has no sendrpid=yes or similar named setting. I can't figure out how to get the SPA8000 to send a configured RPID for outbound SIP VOIP calls. Mark -- Mark G. Thomas (m...@misty.com) Web: http://mgtinternet.com/ Tel: +1-215-512-0112 US: 877-512-0112 -- _ -- 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] SPA8000 outbound CID problem
Hi, I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to both a local Asterisk server and also with a trunk directly to a VOIP provider. Everything works great, except I'm having a problem setting the outbound caller ID to a value different from the SIP username/authname. The SPA8000 has SIP setting for Display Name, User ID, Password, and Auth ID, as well as a "Use Auth ID" checkbox. It's running 6.1.3 firmware, which looks to be the latest, and supports SIP trunking, though even if I don't use trunking, I have the same obstacle if I configure it per-line instead of per-trunk. Inbound CID works fine. When VOIP calls come in via the provider or Asterisk, the SPA generates CID on it's analog ports. The problem is that the outbound caller ID number seems to come from the SIP "User ID" setting, which is also the SIP authentication name. If I instead put the SIP account id into the "Auth ID" field and check the "Use Auth ID" box, Asterisk reports: Registration from 'John Smith ' failed for '1.2.3.4' - Username/auth name mismatch. Sure, I can overide the CID number on our Asterisk server, but I don't have that ability with the VOIP provider's Asterisk server. The outbound caller ID always looks like "John Smith " instead of "John Smith <211212>" no matter how I try to set these fields. I take it the SIP username and auth name need to match, so that leaves me with the question of how to configure a CID number that doesn't necessarily match the SIP user/auth name. Is this a limitation of this device, or is there some other option I'm overlooking? Mark -- Mark G. Thomas (m...@misty.com) Web: http://mgtinternet.com/ Tel: +1-215-512-0112 US: 877-512-0112 -- _ -- 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] SIP equivalent of zap "c" option
Hi,
On Tue, Apr 13, 2010 at 03:37:37PM +0100, Julian Lyndon-Smith wrote:
> At the moment, we have a feature where if someone's sip extension is
> called, we also make another call to their mobile. We use the "c"
> option in the zap dialstring so that the user has to press "#" after
> answering to confirm the call (this prevents things like the
> answermachine grabbing the call if the mobile is switched off).
>
> We are now looking to move towards a sip provider to take all of our
> ISDN calls, so instead of using zap / isdn to call the mobile, we will
> be routing the call over a SIP trunk. Is there any feature of SIP that
> we can use in order to duplicate this functionality (i.e. have to
> press # to confirm the call)
You could do something like this:
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
It works well for me. If I answer then hit "1" on my cellphone I
get the call, otherwise it goes to Asterisk VM, and never to
the cellphone VM.
[whatever]
exten => s,n,Dial(${EXT}&${CCME}&local/1...@internals,20,rt)
[internals]
exten => 101,1,Dial(${MARKCELL},30,tgM(screen))
exten => 101,n,Goto(main-menu,s,1) ; if not answered and accepted
[macro-screen]
exten => s,1,Wait(0.5)
exten => s,n,Read(ACCEPT,followme/options,1,,1,20)
exten => s,n,GotoIf($["${ACCEPT}" = "1"]?yes:no)
exten => s,n(yes),Background(connecting)
exten => s,n,Goto(end)
exten => s,n(no),Set(MACRO_RESULT=CONTINUE)
exten => s,n(end),NoOp
Mark
--
Mark G. Thomas (m...@misty.com)
Web: http://mgtinternet.com/
Tel: +1-215-512-0112 US: 877-512-0112
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Re: [asterisk-users] duration of rfc2833 generated dtmf
Hi, Setting the minimum was necessary in my case, and did affect tones to and from SIP devices as well as the SIP provider, though this was some time ago and you may have different results with your setup than I did with mine. Mark On Wed, Apr 15, 2009 at 11:31:06AM -0400, John covici wrote: > Well, this solution seemed not to work for me, maybe because I did not > set the minimum and also if I am using a sip phone or ATA, the > solution would not apply -- correct me if I am wrong on either of > these. > > on Wednesday 04/15/2009 Mark G. Thomas(m...@misty.com) wrote > > Hi, > > > > On Mon, Apr 13, 2009 at 05:32:45PM -0400, John covici wrote: > > > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > > > however I would like to increase the duration of the tone, its pretty > > > short and some IVR's are unhappy or don't detect it. I did poke > > > around, but it looks like when RFC2833 is used, it actually generates > > > rtp packets of some sort, so I have no idea how to increase that > > > duration. > > > > > > Any assistance would be appreciated. > > > > I had a similar problem. > > > > Adding "dtmf" to the "console =>" line in logger.conf is tremendously > > helpful in diagnosing the dtmf behavior. > > > > My successful work-around was to recompile asterisk with the following > > adjustments. Asterisk then extends the duration of the short tones. I'm > > puzzled why these aren't a run-time configuration settings, since I'd > > think this would be a common problem. > > > > [r...@sylvester asterisk-1.4.24]# diff main/channel.c_orig main/channel.c > > 91c91 > > < #define AST_DEFAULT_EMULATE_DTMF_DURATION 100 > > --- > > > #define AST_DEFAULT_EMULATE_DTMF_DURATION 150 > > 94c94 > > < #define AST_MIN_DTMF_DURATION 80 > > --- > > > #define AST_MIN_DTMF_DURATION 150 > > > > I also later got my provider (Vitelity) to provision my service > > on a different server of theirs, which then also seemed to improve > > both their RFC2833 DTMF reliability and duration. > > > > -Mark -- Mark G. Thomas (m...@misty.com) voice: 215-591-3695 http://mail-cleaner.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
Re: [asterisk-users] duration of rfc2833 generated dtmf
Hi, On Mon, Apr 13, 2009 at 05:32:45PM -0400, John covici wrote: > Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, > however I would like to increase the duration of the tone, its pretty > short and some IVR's are unhappy or don't detect it. I did poke > around, but it looks like when RFC2833 is used, it actually generates > rtp packets of some sort, so I have no idea how to increase that > duration. > > Any assistance would be appreciated. I had a similar problem. Adding "dtmf" to the "console =>" line in logger.conf is tremendously helpful in diagnosing the dtmf behavior. My successful work-around was to recompile asterisk with the following adjustments. Asterisk then extends the duration of the short tones. I'm puzzled why these aren't a run-time configuration settings, since I'd think this would be a common problem. [r...@sylvester asterisk-1.4.24]# diff main/channel.c_orig main/channel.c 91c91 < #define AST_DEFAULT_EMULATE_DTMF_DURATION 100 --- > #define AST_DEFAULT_EMULATE_DTMF_DURATION 150 94c94 < #define AST_MIN_DTMF_DURATION 80 --- > #define AST_MIN_DTMF_DURATION 150 I also later got my provider (Vitelity) to provision my service on a different server of theirs, which then also seemed to improve both their RFC2833 DTMF reliability and duration. -Mark -- Mark G. Thomas (m...@misty.com) voice: 215-591-3695 http://mail-cleaner.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
[asterisk-users] sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?
Hi,
I've been noticing a lot of these messages lately:
"NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?"
Is something broken? I'm running asterisk-1.4.22.1.
They seem to happen in a number of different places where a beep or
recording is played, such as when someone leaves voicemail or when
an AGI script I have plays a time announcement -- lots of different places.
.
-- Executing [...@main-menu:2] Wait("SIP/redacted-09501e28", "1") in new
stack
-- Executing [...@main-menu:3] VoiceMail("SIP/redacted-09501e28", "10|s")
in new stack
-- Playing 'beep' (language 'en')
[Jan 14 12:34:18] NOTICE[10030]: sched.c:220 ast_sched_add_variable: Scheduled
event in 0 ms?
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/10/tmp/EGIKgI
format: wav49, 0x9489080
-- x=1, open writing: /var/spool/asterisk/voicemail/default/10/tmp/EGIKgI
format: gsm, 0x94feaf8
-- x=2, open writing: /var/spool/asterisk/voicemail/default/10/tmp/EGIKgI
format: wav, 0x950c6e0
-- User hung up
.
.
-- Executing [...@main-menu:3] VoiceMail("SIP/redacted-09500428", "10|s") in
new stack
-- Playing 'beep' (language 'en')
[Jan 14 13:23:05] NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled
event in 0 ms?
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/10/tmp/o9SnHz
format: wav49, 0x94feaf8
-- x=1, open writing: /var/spool/asterisk/voicemail/default/10/tmp/o9SnHz
format: gsm, 0x9489080
-- x=2, open writing: /var/spool/asterisk/voicemail/default/10/tmp/o9SnHz
format: wav, 0x9523bf8
-- User hung up
.
-- Launched AGI Script /var/lib/asterisk/agi-bin/talking-clock.agi
-- AGI Script Executing Application: (PlayTones) Options:
(!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!523/20,!0/980,!415/1500)
-- Playing 'at-tone-time-exactly' (escape_digits=) (sample_offset 0)
[Jan 14 13:35:07] NOTICE[10271]: sched.c:220 ast_sched_add_variable: Scheduled
event in 0 ms?
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/30' (language 'en')
[Jan 14 13:35:08] NOTICE[10271]: sched.c:220 ast_sched_add_variable: Scheduled
event in 0 ms?
-- Playing 'digits/5' (language 'en')
[Jan 14 13:35:09] NOTICE[10271]: sched.c:220 ast_sched_add_variable: Scheduled
event in 0 ms?
-- Playing 'vm-and' (language 'en')
-- Playing 'digits/15' (language 'en')
-- Playing 'seconds' (language 'en')
-- AGI Script Executing Application: (PlayTones) Options:
(!0/500,!523/20,!0/980,!523/20,!0/980,!415/1500)
== Spawn extension (from-pots1, s, 2) exited non-zero on 'Zap/5-1'
-- Hungup 'Zap/5-1'
.
--
Mark G. Thomas (m...@misty.com)
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Re: [asterisk-users] CISCO 7940 United_States/7960-tones.xml
Mikel, On Thu, Jan 08, 2009 at 12:52:02AM +1100, Mikel Lindsaar wrote: > I have a smartnet contract for this phone, and have searched high and > low for this file on the Cisco website. > > I need: > > United_States/7960-tones.xml > English_United_States/7960-font.xml > > Every road seems to lead to the Call manager express downloads... I > don't have a CME, so that's basically useles. > > Can anyone point me in the right direction? Those files aren't directly included in the CME downloads. I think their contents must be included in the binary phone load or internal to CME. Using CME, if one sets "cnf-file location flash:", then does a "create cnf-files", they are then written out to the CCME flash, however they default to being on "system:", not the tftp server flash. Have you tried resetting your phone to factory defaults -- "**#" to unlock the settings menu? You might not actually need these files. Mark -- Mark G. Thomas (m...@misty.com) http://mail-cleaner.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
[asterisk-users] SIP host=dynamic help needed for CCME
Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test setup (to avoid NAT complications, etc...) Here's the local static one, which works fine: sip.conf: -- [general] context=default allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes ; [ccme-inbound] type=friend host=10.5.7.130 qualify=yes context=from-ccme allow=all insecure=port,invite canreinvite=no ; [ccme-outbound] type=friend host=10.5.7.130 qualify=yes context=from-ccme trustrpid=yes sendrpid=yes allow=all canreinvite=no dtmfmode=rfc2833 And, in CME: - dial-peer voice 200 voip session protocol sipv2 incoming called-number 211212 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 101 voip description softphones 4-N destination-pattern 4[0-9] monitor probe icmp-ping session protocol sipv2 session target dns:sylvester.home.misty.com dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua no remote-party-id registrar dns:sylvester.home.misty.com expires 3600 secondary sip-server dns:sylvester.home.misty.com I think if I want to use host=dynamic in sip.conf on asterisk, I need to do something like this in CME: --- dial-peer voice 101 voip destination-pattern [1-2][0-9] session protocol sipv2 session target dns:sylvester.home.misty.com dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua authentication username foobar password 7 060F06233B583F4B00 realm NOTSURE registrar dns:sylvester.home.misty.com expires 3600 sip-server dns:sylvester.home.misty.com And, maybe for sip.conf, something like this: --- [foobar] type=friend context=from-ccme host=dynamic secret=notthis username=foobar dtmfmode=rfc2833 But, I'm really not getting far with this. There are tons of examples online of asterisk configurations to initiate connections to static hosts such as SIP providers, and CCME examples using static hosts, but I can't find anything like what I'm doing, even though it seems to me like a common kind of thing to set up. Any help would be greatly appreciated. Mark -- Mark G. Thomas (m...@misty.com) http://mail-cleaner.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
[asterisk-users] sip MWI Messages-Waiting: always reports no messages
Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like "[EMAIL PROTECTED]" in sip.conf, but that doesn't help any. I also tried the same with the mailbox containing messages under [default], but still no luck. I see messages like this if I do "sip set debug ip 10.5.7.130" - Reliably Transmitting (no NAT) to 10.5.7.130:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.7.21:5060;branch=z9hG4bK44627853;rport From: "asterisk" ;tag=as7d9b65d4 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 84 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) --- - from sip.conf: -- [29] insecure=port,invite context=ccme type=friend host=r2.home.misty.com qualify=yes dtmfmode=rfc2833 canreinvite=no nat=no mailbox=29 fromuser=777 vmexten=777 username=29 from voicemail.conf: --- [home] ; testing 29 => 1234,Joe Test,[EMAIL PROTECTED] --- [EMAIL PROTECTED] asterisk]# ls /var/spool/asterisk/voicemail/home/29/INBOX msg.gsm msg.WAV msg0001.wav msg0002.txt msg0003.gsm msg0003.WAV msg0004.wav msg.txt msg0001.gsm msg0001.WAV msg0002.wav msg0003.txt msg0004.gsm msg0004.WAV msg.wav msg0001.txt msg0002.gsm msg0002.WAV msg0003.wav msg0004.txt -- Mark G. Thomas ([EMAIL PROTECTED]) voice: 215-591-3695 http://mail-cleaner.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
Re: [asterisk-users] Dial timeout to cell phones
Hi,
On Tue, Sep 02, 2008 at 03:22:21PM -0400, John Novack wrote:
>
> Steve Repo wrote:
> > Hello,
> >
> > I'm new to asterisk and i'm having a really good time configuring it.
> >
> > I'd like to VoIP-to-PSTN call my SIP number (${MYSIP}) first and then
> > my cell phone (123456) and then finally to my voicemail.
> >
> > Here's my dialplan.
> >
> > exten => s,1,Answer()
> > exten => s,n,Dial(${MYSIP},20)
> > exten => s,n,Dial(Zap/g0/123456,10)
> > exten => s,n,Voicemail([EMAIL PROTECTED])
> > exten => s,n,Hangup()
> >
> > When I receive calls, my IP phone (SIP) rings for 20 seconds as
> > expected. However, asterisk does not terminate calls to my cell phone
> > after 10 seconds and keeps ringing.
> >
> > The call to my cell phone is then answered by cell phone voicemail
> > instead of asterisk voicemail.
> >
> > Any ideas how to go about this?
> >
> > Thanks!
> > Steve
> >
> Zap channels are considered answered once dialing is complete, so your
> "10" second time fails
> No answer supervision on Zap
>
> John Novack
It was challenging to figure this out, since a lot of the online
examples seem to work differently, depending on older versions of Asterisk.
I wanted to ring my cellphone (via SIP provider) and deskphone (via Zap)
simultaneously, but didn't want the call to end up with the cellphone
voicemail, so press "1" on my cellphone if I want to accept the call
there. I even see the original caller ID of the inbound caller on my
cellphone, since I'm out-dialing via a SIP provider.
[inbound]
exten => 211212,1,Playtones(ring) ; play fake ring so caller doesn't wonder
exten => 211212,n,Dial(Zap/g10&local/[EMAIL PROTECTED],,) ; ring FXS and
cell
; http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
;
[internals]
exten => 101,1,Dial(${MARKCELL},30,tM(screen)) ; play message before connecting
; http://www.voip-info.org/wiki/view/Asterisk+tips+findme
; play message to cellphone before connecting inbound call
; http://lists.digium.com/pipermail/asterisk-dev/2005-June/013598.html
;
[macro-screen]
exten => s,1,Wait(0.5)
exten => s,n,Read(ACCEPT,inbound,1,,1,20)
exten => s,n,GotoIf($["${ACCEPT}" = "1"]?yes:no)
exten => s,n(yes),Background(connecting)
exten => s,n,Goto(end) ; Continue on in dialplan to bridge the call
exten => s,n(no),Set(MACRO_RESULT=CONTINUE) ; Hangup the called party and
continue on in the dialplan
exten => s,n(end),NoOp
--
Mark G. Thomas ([EMAIL PROTECTED])
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Re: [asterisk-users] Very loud noise on TDM400
Hi, I had a quirky problem with a TDM844E where after some number of hours or days, I'd suddenly get loud noise in one direction, and only one direction, on the FXS ports. If I picked up a connected phone set, I'd hear dialtone, but my DTMF wouldn't break dialtone, and callers wouldn't be able to hear my voice over the noise they heard. Once the noise lock-up started, I'd have to reboot the server to make it go away. I'm pretty sure I've isolated this to an issue with the 12VDC supply coming from the computer, and solved it with adding some filtering - a 47uF electrolytic capacitor paralleled with a 2.2uF tantalum cap in parallel with the 12V connection. I suspect noise on the 12V power can cause the card to go into some sort of oscillation/noise state. I think they have a separate power supply hook-up you can use instead of getting power from the server's supply. That's probably easier for those less inclined to mess around with capacitors, a soldering iron, and heatshrink tubing. Your problem sounds like it could possibly be similar. Mark On Fri, Jul 25, 2008 at 03:32:46PM -0500, Carlos Chavez wrote: > I am having a problem with and Asterisk installation where two ports > connected to a TDM400 card will have a very loud noise when you try to > dial. The server has an OpenVox D110P, a TDM04B and a Xorcom Astribank > 8 fxs. It is running Zaptel 1.4.11 and Asterisk 1.4.18. > > The problem always happens with two ports (34 and 35) which are > connected to two GSM gateways. They will work fine for a week and then > when you try to dial a mobile number through them you will hear a very > loud noise (like a TV tuned to a channel with no signal at top volume). > To resolve this I have to reboot the server. > > Three weeks ago we reinstalled the server from scratch to see if the > problem would go away but after almost two weeks without interruption > today the noise came back. If you plug the GSM gateway to a regular > phone you get a clear dial tone and you can send and receive calls. > When connected to the card you will get the loud noise when dialing out > and incoming calls will just ring and ring. > > Any ideas why these two ports are misbehaving? > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez Prats > Director de Tecnología > +52-55-91169161 ext 2001 > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark G. Thomas ([EMAIL PROTECTED]) voice: 215-591-3695 http://mail-cleaner.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vitelity dtmfmode=rfc2833 started working!
Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked "sip show peer" and saw that Vitelity for inbound was now reporting "DTMFmode : rfc2833" (it didn't used to), so switched my ountbound dtmfmode to rfc2833 and my problems went away! Yay! It looks like Vitelity now supports rfc2833 on SIP channels. I thought others might be interested in knowing this, as at least in my case it broke things until I changed my settings, and I see this has been a prior source of frustration for many others. Mark -- Mark G. Thomas ([EMAIL PROTECTED]) voice: 215-591-3695 http://mail-cleaner.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Call Screener
Hi Ryan,
On Wed, Jul 09, 2008 at 05:54:28PM -0400, Ryan M. Colbert wrote:
> I'm trying to build a simple accept/reject screening app for inbound calls
> that * forwards to my cell phone. Basically I want * to announce the caller
> ID and then let me press 1 to accept the call or 2 to reject the call and
> send the outside party to voicemail.
I'm doing something similar, ringing a Zaptel port and calling a cellphone,
as per below. I had success using Read() to get the accept/reject.
Note that you need to Set(MACRO_RESULT=CONTINUE) if you do NOT want to accept
the call on the cellphone, and hence want the diaplan to CONTINUE.
In my case, I don't need to announce the caller ID since the cellphone
simply displays it, and I'm letting the human or analog voicemail on the
zaptel port take the call if the cellphone user doesn't press 1.
> I've been messing around with variation of the script below... can anyone
> tell me what I'm doing wrong? It's got to be something obvious that I've
> overlooked.
>
> Thanks!!!
>
> [main]
> exten => s,1,Answer
> exten => s,n,Ringing
> exten => s,n,Wait(1)
> exten => s,n,Dial(SIP/[EMAIL PROTECTED],120,gM(screen))
> exten => s,n,PlayBack(vm-goodbye)
> exten => s,n,Hangup
>
> [macro-screen]
> exten => s,1,Wait(1)
> ;exten => s,n,SayDigits(${CALLERID(num)})
> exten => s,n,Set(TIMEOUT(digit)=5)
> exten => s,n,Set(TIMEOUT(response)=30)
> exten => s,n,Background(accept-reject)
>
> exten => 1,1,Set(MACRO_RESULT=CONTINUE)
> exten => 2,1,PlayBack(vm-goodbye)
> exten => 2,2,Hangup
>
> exten => s,6,Wait(10)
> exten => i,1,Goto(TT_VO,s,1)
...
[inbound]
exten => 211212,1,Playtones(ring)
exten => 211212,n,Dial(${PTNR}&local/[EMAIL PROTECTED],,t)
[internals]
exten => 101,1,Dial(${MARKCELL},30,tM(screen))
[macro-screen]
exten => s,1,Wait(0.5)
exten => s,n,Read(ACCEPT,inbound,1,,1,20)
exten => s,n,GotoIf($["${ACCEPT}" = "1"]?yes:no)
exten => s,n(yes),Background(connecting)
exten => s,n,Goto(end)
exten => s,n(no),Set(MACRO_RESULT=CONTINUE)
exten => s,n(end),NoOp
--
Mark G. Thomas ([EMAIL PROTECTED])
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[asterisk-users] Centos-5.2 and zaptel-1.4.11 do not get along well
Hi, After doing a "yum update" on my previously Centos-5.1 system, now zaptel-1.4.11 fails to build with this below. I figured I better caution people because it sucks being down, and I haven't figured out how to install the previously compiled modules into the new kernel or any other work around. ... LD [M] /opt/src/asterisk/zaptel-1.4.11/kernel/wcte12xp/wcte12xp.o CC [M] /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o In file included from /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/xpd.h:26, from /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.c:27: /opt/src/asterisk/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types for ‘bool’ include/linux/types.h:36: error: previous declaration of ‘bool’ was here make[4]: *** [/opt/src/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 make[3]: *** [/opt/src/asterisk/zaptel-1.4.11/kernel/xpp] Error 2 make[2]: *** [_module_/opt/src/asterisk/zaptel-1.4.11/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-92.1.6.el5-i686' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/opt/src/asterisk/zaptel-1.4.11' make: *** [all] Error 2 -- Mark G. Thomas ([EMAIL PROTECTED]) voice: 215-591-3695 http://mail-cleaner.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
