Re: [asterisk-users] Problems with Outbound Calls
Two things The dev/zap problem was probably fixed by a modprobe that occurred on the reload and therefore had no relevance to the chmod. Ztdummy is created by zaptel and used in some non-analog functions AFAIK -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, February 27, 2009 6:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problems with Outbound Calls On Fri, Feb 27, 2009 at 07:12:41PM +0900, Wye-khe Kwok wrote: > We managed to find a fix through the following (For anyone who's > interested): > > > > Running /sbin/ztcfg -vv to configure Zaptel initially resulted in an > error of: > > > > Notice: Configuration file is /etc/zaptel.conf > > line 0: Unable to open master device '/dev/zap/ctl' > > > > We then Chmodded everything under /dev/zap/ , rebooted and almost fell > off our chairs when it worked! That's odd. The files under /dev/ are actually on a ramdisk. That is: they are wiped on reboot. I can't see how your chmod had any effect. What's the output of: df /dev/zap/ctl > > We were initially on the impression that Zaptel is only used with > Analogue - can anyone verify this? Also with E1, T1, (J1?), BRI and TDM over Ethernet. -- 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
Re: [asterisk-users] Problems with Outbound Calls
On Fri, Feb 27, 2009 at 07:12:41PM +0900, Wye-khe Kwok wrote: > We managed to find a fix through the following (For anyone who's > interested): > > > > Running /sbin/ztcfg -vv to configure Zaptel initially resulted in an > error of: > > > > Notice: Configuration file is /etc/zaptel.conf > > line 0: Unable to open master device '/dev/zap/ctl' > > > > We then Chmodded everything under /dev/zap/ , rebooted and almost fell > off our chairs when it worked! That's odd. The files under /dev/ are actually on a ramdisk. That is: they are wiped on reboot. I can't see how your chmod had any effect. What's the output of: df /dev/zap/ctl > > We were initially on the impression that Zaptel is only used with > Analogue - can anyone verify this? Also with E1, T1, (J1?), BRI and TDM over Ethernet. -- 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
Re: [asterisk-users] Problems with Outbound Calls
> Notice: Configuration file is /etc/zaptel.conf > line 0: Unable to open master device '/dev/zap/ctl' > We then Chmodded everything under /dev/zap/ , rebooted and almost fell off > our chairs when it worked! By right, if the problem is due to this error, you should see a permission error message in /var/log/asterisk/messages. What it means is the directory permissions might be wrong somewhere in the beginning. This may not be related to your original warning. Warning [2630]: config.c:768 process_text_line: Unknown Directive at line 231 of /etc/asterisk/../zaptel.conf >We were initially on the impression that Zaptel is only used with Analogue > – can anyone verify this? No, it is responsible for PRI channels as well. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wye-khe Kwok Sent: Friday, 27 February 2009 9:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problems with Outbound Calls Hey, thanks for the help David, Tzafrir. Lots of config tips there ☺ We managed to find a fix through the following (For anyone who’s interested): Running /sbin/ztcfg –vv to configure Zaptel initially resulted in an error of: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! We were initially on the impression that Zaptel is only used with Analogue – can anyone verify this? YK ___ -- 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] Problems with Outbound Calls
Hey, thanks for the help David, Tzafrir. Lots of config tips there :-) We managed to find a fix through the following (For anyone who's interested): Running /sbin/ztcfg -vv to configure Zaptel initially resulted in an error of: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! We were initially on the impression that Zaptel is only used with Analogue - can anyone verify this? YK ___ -- 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] Problems with Outbound Calls
thanks
2009/2/26 Tzafrir Cohen
> On Thu, Feb 26, 2009 at 05:06:56PM -0200, David fire wrote:
> > sorry but how do you know the warning is from an # ?
>
> 'Directive' is something that begins with a '#'.
>
> --
>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
>
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Re: [asterisk-users] Problems with Outbound Calls
On Thu, Feb 26, 2009 at 05:06:56PM -0200, David fire wrote: > sorry but how do you know the warning is from an # ? 'Directive' is something that begins with a '#'. -- 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
Re: [asterisk-users] Problems with Outbound Calls
sorry but how do you know the warning is from an # ?
he only post this from zaptel.conf
span = 1,1,0,esf,b8zs
bchan = 1-23
dchan = 24
loadzone = jp
defaultzone = jp
David
2009/2/26 Tzafrir Cohen
> On Thu, Feb 26, 2009 at 11:50:57AM -0200, David fire wrote:
> > hi
> > you should first solve this
> >
> > Warning [2630]: config.c:768 process_text_line: Unknown Directive at
> > line 231 of /etc/asterisk/../zaptel.conf
>
> In zaptel.conf it is perfectly legal to have lines beginning with a '#'.
> With Asterisk '#' is reserved for special directives (currently only
> #include and #exec exist).
>
> The asterisk-gui has a script that creates
>
> /etc/asterisk/dahdi_guiread.conf
>
> With the following content (if the system is Zaptel and not DAHDI)
>
> [general]
> zaptel
> #include "../zaptel.conf"
>
> This makes it simple to read zaptel.conf through the manager interface
> of reading Asterisk configuration files. It also has several other
> atvantages:
>
> * The user is well aware of its existance if there are any comments in
> zaptel.conf (the message pointed out above)
> * Its format is not of a real valid configuration file ('zaptel' is a
> key without a value), which serves to provides interesting chanlanges
> to configuration parser writiers.
> * It assumes the Asterisk configuration file sits at /etc/asterisk .
> Indeed why even bother with a situation where Asterisk can only read
> Zaptel's configuration but not write it?
>
> --
> 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
>
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Re: [asterisk-users] Problems with Outbound Calls
On Thu, Feb 26, 2009 at 11:50:57AM -0200, David fire wrote:
> hi
> you should first solve this
>
> Warning [2630]: config.c:768 process_text_line: Unknown Directive at
> line 231 of /etc/asterisk/../zaptel.conf
In zaptel.conf it is perfectly legal to have lines beginning with a '#'.
With Asterisk '#' is reserved for special directives (currently only
#include and #exec exist).
The asterisk-gui has a script that creates
/etc/asterisk/dahdi_guiread.conf
With the following content (if the system is Zaptel and not DAHDI)
[general]
zaptel
#include "../zaptel.conf"
This makes it simple to read zaptel.conf through the manager interface
of reading Asterisk configuration files. It also has several other
atvantages:
* The user is well aware of its existance if there are any comments in
zaptel.conf (the message pointed out above)
* Its format is not of a real valid configuration file ('zaptel' is a
key without a value), which serves to provides interesting chanlanges
to configuration parser writiers.
* It assumes the Asterisk configuration file sits at /etc/asterisk .
Indeed why even bother with a situation where Asterisk can only read
Zaptel's configuration but not write it?
--
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
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Re: [asterisk-users] Problems with Outbound Calls
hi
you should first solve this
Warning [2630]: config.c:768 process_text_line: Unknown Directive at
line 231 of /etc/asterisk/../zaptel.conf
check what do you have in the line 231 of your zaptel.conf file.
David
2009/2/26 Wye-khe Kwok
> Hi everyone!
>
> I'm quite a newbie at this Asterisk stuff so please bear with me.
>
> We've recently decided to start training in Asterisk via AsteriskNow!
>
> Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
>
> The box we have is paired with a Digium TE110P and we've managed to get
> it to the point where incoming calls via a single DID (from NTT Japan)
> can be received and answered (INS1500 here in Japan). We're using SIP
> phones here.
>
> However, on attempting outbound calls, I've noticed the following
> message on the Live Console.
>
> Warning [2630]: config.c:768 process_text_line: Unknown Directive at
> line 231 of /etc/asterisk/../zaptel.conf
>
> The phones have no dial tones and we get nothing but silence when
> dialing no.s and hitting the 'send' button.
>
> Following are some excerpts from the Conf Files (Sorry about the spam -
> I'm not sure what's redundant)
>
> ---
>
> Sip.Conf: (Only included info that didn't start with semicolons)
>
> [general]
> context=default ; Default context for incoming calls
> allowoverlap=no ; Disable overlap dialing support.
> (Default is yes)
> bindport=5060 ; UDP Port to bind to (SIP standard port
> is 5060)
> bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
> to all)
> srvlookup=yes ; Enable DNS SRV lookups on outbound
> calls
>
> language=jp ; Default language setting for all
> users/peers
>; This may also be set for individual
> users/peers
>
>
> Extensions.Conf:
>
> [general]
> static = yes
> writeprotect = no
> autofallthrough = yes
> clearglobalvars = no
> priorityjumping = no
>
> [globals]
> span_1 = Zap/g1
>
> [dundi-e164-canonical]
>
> [dundi-e164-customers]
>
> [dundi-e164-via-pstn]
>
> [dundi-e164-local]
> include => dundi-e164-canonical
> include => dundi-e164-customers
> include => dundi-e164-via-pstn
>
> [dundi-e164-switch]
> switch => DUNDi/e164
>
> [dundi-e164-lookup]
> include => dundi-e164-local
> include => dundi-e164-switch
>
> [macro-dundi-e164]
> exten => s,1,Goto(${ARG1},1)
> include => dundi-e164-lookup
>
> [iaxtel700]
> exten =>
> _91700XXX,1,Dial(IAX2/${iaxin...@iaxtel.com/${EXTEN:1...@iaxtel
> )
>
> [iaxprovider]
>
> [trunkint]
> exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
> exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunkld]
> exten => _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
> exten => _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunklocal]
> exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [trunktollfree]
> exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>
> [international]
> ignorepat => 9
> include => longdistance
> include => trunkint
>
> [longdistance]
> ignorepat => 9
> include => local
> include => trunkld
>
> [local]
> ignorepat => 9
> include => default
> include => parkedcalls
> include => trunklocal
> include => iaxtel700
> include => trunktollfree
> include => iaxprovider
>
> [macro-stdexten]
> exten => s,1,Dial(${ARG2},20)
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(${ARG1},u)
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(${ARG1},b)
> exten => s-BUSY,2,Goto(default,s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> [macro-stdPrivacyexten]
> exten => s,1,Dial(${ARG2},20|p)
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
> exten => s-DONTCALL,1,Goto(${ARG3},s,1)
> exten => s-TORTURE,1,Goto(${ARG4},s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> [macro-page]
> exten => s,1,ChanIsAvail(${ARG1}|js)
> exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
> exten => s,n(autoanswer),Set(_ALERT_INFO="RA")
> exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)
> exten => s,n,NoOp()
> exten => s,n,Dial(${ARG1}||)
> exten => s,n(fail),Hangup
>
> [page]
> exten => _X.,1,Macro(page,SIP/${EXTEN})
>
> [default]
> exten => 6050,1,VoiceMailMain
> exten = 7000,1,Goto(voicemenu-custom-1|s|1)
> exten => 6000,1,MeetMe(${EXTEN}|MI)
> exten = 3010,1,Goto(ringroups-custom-1|s|1)
> exten = 3020,1,Goto(ringroups-custom-2|s|1)
> exten = 6005,1,Queue(${EXTEN})
>
> [voicemenu-custom-1]
> include = default
> comment = Welcome
> alias_exten = 7000
> exten = s,1,Answer
> exten = s,2,Wai
[asterisk-users] Problems with Outbound Calls
Hi everyone!
I'm quite a newbie at this Asterisk stuff so please bear with me.
We've recently decided to start training in Asterisk via AsteriskNow!
Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
The box we have is paired with a Digium TE110P and we've managed to get
it to the point where incoming calls via a single DID (from NTT Japan)
can be received and answered (INS1500 here in Japan). We're using SIP
phones here.
However, on attempting outbound calls, I've noticed the following
message on the Live Console.
Warning [2630]: config.c:768 process_text_line: Unknown Directive at
line 231 of /etc/asterisk/../zaptel.conf
The phones have no dial tones and we get nothing but silence when
dialing no.s and hitting the 'send' button.
Following are some excerpts from the Conf Files (Sorry about the spam -
I'm not sure what's redundant)
---
Sip.Conf: (Only included info that didn't start with semicolons)
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
language=jp ; Default language setting for all
users/peers
; This may also be set for individual
users/peers
Extensions.Conf:
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[globals]
span_1 = Zap/g1
[dundi-e164-canonical]
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
switch => DUNDi/e164
[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch
[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
[iaxtel700]
exten =>
_91700XXX,1,Dial(IAX2/${iaxin...@iaxtel.com/${EXTEN:1...@iaxtel)
[iaxprovider]
[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten => _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
ignorepat => 9
include => local
include => trunkld
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[macro-stdexten]
exten => s,1,Dial(${ARG2},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${ARG1},u)
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Voicemail(${ARG1},b)
exten => s-BUSY,2,Goto(default,s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)
exten => a,1,VoicemailMain(${ARG1})
[macro-stdPrivacyexten]
exten => s,1,Dial(${ARG2},20|p)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${ARG1})
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Voicemail(b${ARG1})
exten => s-BUSY,2,Goto(default,s,1)
exten => s-DONTCALL,1,Goto(${ARG3},s,1)
exten => s-TORTURE,1,Goto(${ARG4},s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)
exten => a,1,VoicemailMain(${ARG1})
[macro-page]
exten => s,1,ChanIsAvail(${ARG1}|js)
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA")
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)
exten => s,n,NoOp()
exten => s,n,Dial(${ARG1}||)
exten => s,n(fail),Hangup
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
[default]
exten => 6050,1,VoiceMailMain
exten = 7000,1,Goto(voicemenu-custom-1|s|1)
exten => 6000,1,MeetMe(${EXTEN}|MI)
exten = 3010,1,Goto(ringroups-custom-1|s|1)
exten = 3020,1,Goto(ringroups-custom-2|s|1)
exten = 6005,1,Queue(${EXTEN})
[voicemenu-custom-1]
include = default
comment = Welcome
alias_exten = 7000
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Background(thank-you-for-calling)
exten = s,4,Background(if-u-know-ext-dial)
exten = s,5,Background(otherwise)
exten = s,6,Background(to-reach-operator)
exten = s,7,Background(pls-hold-while-try)
exten = s,8,WaitExten(6)
[numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls
exten =
_9011XXX!,1,Macro(trunkdial,${span_1}/${EXTEN:1},${span_1_cid})
comment = _9011XXX!,1,International,standard
exten =
_9256XXX!,1,Macro(trunkdial,${span_1}/${EXTEN:4},${span_1_cid})
comment = _9256XXX!,1,Local,stand
Re: [asterisk-users] Problems with outbound calls through VSP
Update:
I was able to obtain another VSP to try and rule out Broadvoice. Seems
that either my Broadvoice settings, or something on their end is causing
the brief screech noise upon playing the first sound.
However, with this new VSP I still have the AMD (Answering Machine
Detect) problem where it locks up unless I play some sound before
calling AMD. So my modified question is, has anyone ever had a problem
with AMD through a VSP (SIP, in this case). And it does *not* lock up
when calling phones local to the server.
Christopher Robinson wrote:
Bear with me this is a bit long winded. I am having some issues
making automated outbound calls over Broadvoice from my Asterisk 1.4.2
server. For reference, none of the below issues happen when I make
the calls to VoIP phones attached to the Asterisk server. What I am
trying to do is call, using a .call file, out via the SIP trunk we
have setup, and when the party picks up use AMD to detect if it's
reached a human or machine. If it's human then one message will be
played, and if machine another will be played theoretically after the
answering machine/voicemail is done playing. By the way, I'd like to
mention that this is not at all for spamming, or telemarketing. This
is an appointment reminder service.
from extensions.conf:
[mycontext]
exten => 899,1,Answer
exten => 899,2,Wait(2)
exten => 899,3,AMD
exten => 899,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
exten => 899,n(mach),WaitForSilence(2500)
exten => 899,n,Playback(were-sorry)
exten => 899,n,Hangup
exten => 899,n(humn),WaitForSilence(500)
exten => 899,n,Playback(welcome)
exten => 899,n,Hangup
The call goes out fine. When I pick it up AMD basically locks up,
although not exactly because as you can see below it does recognize
the HANGUP. However, it will not recognize my voice or dead air no
matter how long I stay on the call to try. If I just let my voicemail
pickup it does the same thing...takes forever for the call to
terminate. Again, this all works as expected when I make the call to
a SIP phone attached to the Asterisk server.
-- Attempting call on SIP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 1)
> Channel SIP/sip.broadvoice.com-08bad080 was answered.
-- Executing [EMAIL PROTECTED]:1]
Answer("SIP/sip.broadvoice.com-08bad080", "") in new stack
-- Executing [EMAIL PROTECTED]:2]
AMD("SIP/sip.broadvoice.com-08bad080", "") in new stack
-- AMD: SIP/sip.broadvoice.com-08bad080 (null) (Fmt: 4)
-- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
[800] totalAnalysisTime [5000] minimumWordLength [100]
betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]
-- AMD: HANGUP
I did find a solution to this "lock up". That was to play a bit of
silence at any point before I actually call AMD (even before Answer
works):
[mycontext]
exten => 899,1,Playback(silence/1)
exten => 899,2,Answer
Although I don't particularly like this solution, as I'm just patching
the problem that I still don't understand, plus it adds a little more
delay that confuses the called party.
Also, when I tried this I realized yet another issue, which could be
the underlying cause of the whole thing. No matter what sound it is,
no matter if I use AMD or not, the very first sound that I play
results in a short "screech" sound before it is played. This happens
every time without fail. If I were to guess, I would say that there
is some data in the audio channel that is not audio data, and is being
represented with that screech sound...but of course that's just a guess.
Any help would be greatly appreciated. Below are some relevant
configuration settings:
sip.conf:
[general]
context=testusers ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
externip=xx.xx.xx.xx
localnet=192.168.1.0/255.255.255.0
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
pedantic=no
register =>
[EMAIL PROTECTED]:mysecret:[EMAIL PROTECTED]
[sip.broadvoice.com]
allow=ulaw
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=716XXX
secret=mysecret
username=716XXX
insecure=very
context=from_broadvoice
authname=716XXX
dtmf=inband
dtmfmode=inband
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=yes
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[asterisk-users] Problems with outbound calls through VSP
Bear with me this is a bit long winded. I am having some issues making
automated outbound calls over Broadvoice from my Asterisk 1.4.2 server.
For reference, none of the below issues happen when I make the calls to
VoIP phones attached to the Asterisk server. What I am trying to do is
call, using a .call file, out via the SIP trunk we have setup, and when
the party picks up use AMD to detect if it's reached a human or
machine. If it's human then one message will be played, and if machine
another will be played theoretically after the answering
machine/voicemail is done playing. By the way, I'd like to mention that
this is not at all for spamming, or telemarketing. This is an
appointment reminder service.
from extensions.conf:
[mycontext]
exten => 899,1,Answer
exten => 899,2,Wait(2)
exten => 899,3,AMD
exten => 899,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
exten => 899,n(mach),WaitForSilence(2500)
exten => 899,n,Playback(were-sorry)
exten => 899,n,Hangup
exten => 899,n(humn),WaitForSilence(500)
exten => 899,n,Playback(welcome)
exten => 899,n,Hangup
The call goes out fine. When I pick it up AMD basically locks up,
although not exactly because as you can see below it does recognize the
HANGUP. However, it will not recognize my voice or dead air no matter
how long I stay on the call to try. If I just let my voicemail pickup
it does the same thing...takes forever for the call to terminate.
Again, this all works as expected when I make the call to a SIP phone
attached to the Asterisk server.
-- Attempting call on SIP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 1)
> Channel SIP/sip.broadvoice.com-08bad080 was answered.
-- Executing [EMAIL PROTECTED]:1]
Answer("SIP/sip.broadvoice.com-08bad080", "") in new stack
-- Executing [EMAIL PROTECTED]:2]
AMD("SIP/sip.broadvoice.com-08bad080", "") in new stack
-- AMD: SIP/sip.broadvoice.com-08bad080 (null) (Fmt: 4)
-- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
[800] totalAnalysisTime [5000] minimumWordLength [100]
betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]
-- AMD: HANGUP
I did find a solution to this "lock up". That was to play a bit of
silence at any point before I actually call AMD (even before Answer works):
[mycontext]
exten => 899,1,Playback(silence/1)
exten => 899,2,Answer
Although I don't particularly like this solution, as I'm just patching
the problem that I still don't understand, plus it adds a little more
delay that confuses the called party.
Also, when I tried this I realized yet another issue, which could be the
underlying cause of the whole thing. No matter what sound it is, no
matter if I use AMD or not, the very first sound that I play results in
a short "screech" sound before it is played. This happens every time
without fail. If I were to guess, I would say that there is some data
in the audio channel that is not audio data, and is being represented
with that screech sound...but of course that's just a guess.
Any help would be greatly appreciated. Below are some relevant
configuration settings:
sip.conf:
[general]
context=testusers ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
externip=xx.xx.xx.xx
localnet=192.168.1.0/255.255.255.0
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
pedantic=no
register =>
[EMAIL PROTECTED]:mysecret:[EMAIL PROTECTED]
[sip.broadvoice.com]
allow=ulaw
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=716XXX
secret=mysecret
username=716XXX
insecure=very
context=from_broadvoice
authname=716XXX
dtmf=inband
dtmfmode=inband
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=yes
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