Re: [asterisk-users] Exit Dial Application
Mea culpa. Just being a bit lazy. In real use, the _X.-noanswer would be
s-NOANSWER (at least that's how it works in MY Dialplan).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Wednesday, April 15, 2009 4:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Exit Dial Application
On Wed, Apr 15, 2009 at 09:24:54AM -0500, Danny Nicholas wrote:
> This is what you "Really" want; It should work with SIP or Zap
>
> exten => _X.,1,Dial(${DIALNUM},${ARG2},tT)
> exten => _X.-NOANSWER,1,background(press5tocallback)
> exten => -X.-NOANSWER,2,waitexten(5)
Anything after a '.' in a pattern match is practically ignored.
Also note that X only matches digits.
--
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] Exit Dial Application
On Wed, Apr 15, 2009 at 09:24:54AM -0500, Danny Nicholas wrote:
> This is what you "Really" want; It should work with SIP or Zap
>
> exten => _X.,1,Dial(${DIALNUM},${ARG2},tT)
> exten => _X.-NOANSWER,1,background(press5tocallback)
> exten => -X.-NOANSWER,2,waitexten(5)
Anything after a '.' in a pattern match is practically ignored.
Also note that X only matches digits.
--
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] Exit Dial Application
If you set your ARG2 to a value like 6, the phone would only ring twice before noanswer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph Fürstaller Sent: Wednesday, April 15, 2009 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Exit Dial Application -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Danny, Thanks for your replay. Jep, that would be a possibility. But then the user has to wait until my dialtime is over. If he/she is that inpatient, then with my solution he/she can end the dialing whenever needed. But, I'll try your successtion, looks interesting. chris... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Atis, Atis Lezdins schrieb: > I think the limitation could be by analogous Zap phones, as they > probably don't support sending DTMF on unanswered channel. You could > try it opposite way - Dial from SIP phone to Zap. Noop, it's not a Zap problem. I tried it with two SIP phones, same behavior. Bit odd : / > > Regards, > Atis chris... - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße 14, 5301 Eugendorf Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 commpany (26667269) Email: c.fuerstal...@commpany.at -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAknmDNAACgkQR0exH8dhr/ZhmQCfQ4RaMsglGxx23McMbBiflsA9 y0IAoK+EBojiyPF1qj1hhITM8vzBPVmH =LkKe -END PGP SIGNATURE- ___ -- 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] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Thanks for your replay. Jep, that would be a possibility. But then the user has
to wait until my dialtime is over. If he/she is that inpatient, then with my
solution he/she can end the dialing whenever needed. But, I'll try your
successtion, looks
interesting.
chris...
Danny Nicholas schrieb:
> This is what you "Really" want; It should work with SIP or Zap
>
> exten => _X.,1,Dial(${DIALNUM},${ARG2},tT)
> exten => _X.-NOANSWER,1,background(press5tocallback)
> exten => -X.-NOANSWER,2,waitexten(5)
> exten => 5,1,goto(callback,s,1)
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Atis Lezdins
> Sent: Wednesday, April 15, 2009 9:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Exit Dial Application
>
> On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller
> wrote:
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>> Hi Danny,
>>
>> Danny Nicholas schrieb:
>>> Here's how core show application dial says you should do it:
>>> Change your dial to
>>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
>> I'm not sure if this is correct. core show application dial says:
>> Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL])
>> If I configure what you wrote, then callback is passed as URL to the
> called party.
>> "The optional URL will be sent to the called party if the channel supports
> it."
>> I don't think that's what I want.
>> What I want is: If A dials B and B doesn't answer, A can press 5 and place
> an automatic
>> callback. If B is back and places or takes a call, the automatic callback
> to A should be
>> started.
>>
>> I've found a possibility to do this via answering the call before the
> dial. But ... that's
>> not an ideal solution. I would prefer not to answer the call in the
> dialplan. Does the
>> option 'd' implies an answered channel? Or is this a Bug?
>>
>
> I think the limitation could be by analogous Zap phones, as they
> probably don't support sending DTMF on unanswered channel. You could
> try it opposite way - Dial from SIP phone to Zap.
>
> Regards,
> Atis
>
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
This is what you "Really" want; It should work with SIP or Zap
exten => _X.,1,Dial(${DIALNUM},${ARG2},tT)
exten => _X.-NOANSWER,1,background(press5tocallback)
exten => -X.-NOANSWER,2,waitexten(5)
exten => 5,1,goto(callback,s,1)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Atis Lezdins
Sent: Wednesday, April 15, 2009 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Exit Dial Application
On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi Danny,
>
> Danny Nicholas schrieb:
>> Here's how core show application dial says you should do it:
>> Change your dial to
>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
> I'm not sure if this is correct. core show application dial says:
> Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL])
> If I configure what you wrote, then callback is passed as URL to the
called party.
> "The optional URL will be sent to the called party if the channel supports
it."
>
> I don't think that's what I want.
> What I want is: If A dials B and B doesn't answer, A can press 5 and place
an automatic
> callback. If B is back and places or takes a call, the automatic callback
to A should be
> started.
>
> I've found a possibility to do this via answering the call before the
dial. But ... that's
> not an ideal solution. I would prefer not to answer the call in the
dialplan. Does the
> option 'd' implies an answered channel? Or is this a Bug?
>
I think the limitation could be by analogous Zap phones, as they
probably don't support sending DTMF on unanswered channel. You could
try it opposite way - Dial from SIP phone to Zap.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi Danny,
>
> Danny Nicholas schrieb:
>> Here's how core show application dial says you should do it:
>> Change your dial to
>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
> I'm not sure if this is correct. core show application dial says:
> Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL])
> If I configure what you wrote, then callback is passed as URL to the called
> party.
> "The optional URL will be sent to the called party if the channel supports
> it."
>
> I don't think that's what I want.
> What I want is: If A dials B and B doesn't answer, A can press 5 and place an
> automatic
> callback. If B is back and places or takes a call, the automatic callback to
> A should be
> started.
>
> I've found a possibility to do this via answering the call before the dial.
> But ... that's
> not an ideal solution. I would prefer not to answer the call in the dialplan.
> Does the
> option 'd' implies an answered channel? Or is this a Bug?
>
I think the limitation could be by analogous Zap phones, as they
probably don't support sending DTMF on unanswered channel. You could
try it opposite way - Dial from SIP phone to Zap.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Danny Nicholas schrieb:
> Here's how core show application dial says you should do it:
> Change your dial to
> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
I'm not sure if this is correct. core show application dial says:
Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL])
If I configure what you wrote, then callback is passed as URL to the called
party.
"The optional URL will be sent to the called party if the channel supports it."
I don't think that's what I want.
What I want is: If A dials B and B doesn't answer, A can press 5 and place an
automatic
callback. If B is back and places or takes a call, the automatic callback to A
should be
started.
I've found a possibility to do this via answering the call before the dial. But
... that's
not an ideal solution. I would prefer not to answer the call in the dialplan.
Does the
option 'd' implies an answered channel? Or is this a Bug?
Chris...
>
> This will execute the macro, then dial the number. You will have to take
> the hangups out of callback.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
> Fürstaller
> Sent: Tuesday, April 14, 2009 11:50 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Exit Dial Application
>
> Hi,
>
> Thanks for your replay. But this can only be done before or after the dial,
> but I wanna do it during the dial, when user A is waiting for user B,
> answering the phone. This should be possible, right?
>
> I hope anyone knows if this is possible.
>
> Chris...
>
> Danny Nicholas schrieb:
>> I'd change callback to this
>> [callback]
>> Exten => s,1,Playback(press5msg)
>> Exten => s,n,Waitexten(5)
>> Exten => s,n,Hangup
>> exten => 5,1,agi(str_concat.sh)
>> exten => 5,n,Hangup
>
>> This will play a message, wait 5 seconds for user to press 5, then hangup
> if
>> they don't.
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
>> Fuerstaller
>> Sent: Tuesday, April 14, 2009 5:04 AM
>> To: Asterisk Users Mailing List
>> Subject: [asterisk-users] Exit Dial Application
>
>> Hi,
>
>> I' try to implement an automatic callback mechanism, just for local SIP
>> calls.. Callback
>> on busy and on no answer. If the other party doen't answer, it should be
>> possible to press
>> 5 to place an callback.
>
>> Here is my dial:
>> exten => _X.,1,Set(EXITCONTEXT=callback)
>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>
>> And here the script for callback.
>> [callback]
>> exten => 5,1,agi(str_concat.sh)
>> exten => 5,n,Hangup
>
>> If I call someone and press 5, nothing happens. What could be a problem?
>> DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted
> correctly,
>> I can enter
>> the voicmail menue.
>
>> I'm using Asterisk 1.4.21.1.
>
>> Any successions are very appreciated.
>
>> Chris...
>
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- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
Here's how core show application dial says you should do it:
Change your dial to
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
This will execute the macro, then dial the number. You will have to take
the hangups out of callback.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fürstaller
Sent: Tuesday, April 14, 2009 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial,
but I wanna do it during the dial, when user A is waiting for user B,
answering the phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
> I'd change callback to this
> [callback]
> Exten => s,1,Playback(press5msg)
> Exten => s,n,Waitexten(5)
> Exten => s,n,Hangup
> exten => 5,1,agi(str_concat.sh)
> exten => 5,n,Hangup
>
> This will play a message, wait 5 seconds for user to press 5, then hangup
if
> they don't.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
> Fuerstaller
> Sent: Tuesday, April 14, 2009 5:04 AM
> To: Asterisk Users Mailing List
> Subject: [asterisk-users] Exit Dial Application
>
> Hi,
>
> I' try to implement an automatic callback mechanism, just for local SIP
> calls.. Callback
> on busy and on no answer. If the other party doen't answer, it should be
> possible to press
> 5 to place an callback.
>
> Here is my dial:
> exten => _X.,1,Set(EXITCONTEXT=callback)
> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>
> And here the script for callback.
> [callback]
> exten => 5,1,agi(str_concat.sh)
> exten => 5,n,Hangup
>
> If I call someone and press 5, nothing happens. What could be a problem?
> DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted
correctly,
> I can enter
> the voicmail menue.
>
> I'm using Asterisk 1.4.21.1.
>
> Any successions are very appreciated.
>
> Chris...
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- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Atis Lezdins schrieb:
>
> Ok, at first glance the app_macro looks suspicious, can You try
> calling dial without Macro?
Tried it without macro -> same behavior.
>
> If unsuccessful, You could enable debug level 2, it will tell way much
> more of everything, including DTMF events etc. Btw, does DTMF work at
> all for this Zap/ line? You could verify that by using Read before
> Dial.
Called Read, entered numbers, echoed them correctly.
Then I tried something ... different. I Answered the call before calling the
macro. And voila it's working. Do I have to answer the channel before Dial
option 'd' is working? It's a bit odd, cause the dial duration starts counting
and I hear a
'beep'. That's not ideal : / I've attached a full.log.
>
> Regards,
> Atis
>
chris...
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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[Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Answer'
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing
[s-d...@macro-dialone:10] Answer("Zap/31-1", "") in new stack
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing
[s-d...@macro-dialone:11] Set("Zap/31-1", "_EXITCONTEXT=callback") in new stack
[Apr 15 00:02:34] DEBUG[8816] app_macro.c: Executed application: Set
[Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Set'
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing
[s-d...@macro-dialone:12] Set("Zap/31-1", "orig_exten=236") in new stack
[Apr 15 00:02:34] DEBUG[8816] app_macro.c: Executed application: Set
[Apr 15 00:02:34] DEBUG[8816] pbx.c: Launching 'Dial'
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Executing
[s-d...@macro-dialone:13] Dial("Zap/31-1", "SIP/236|30|dtT") in new stack
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Asked to create a SIP channel with
formats: 0x8 (alaw)
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Allocating new SIP dialog for (No
Call-ID) - INVITE (With RTP)
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Setting NAT on RTP to On
[Apr 15 00:02:34] DEBUG[8816] acl.c: # Testing 10.10.5.1 with 10.10.0.0
[Apr 15 00:02:34] DEBUG[8816] rtp.c: Channel 'Zap/31-1' has no RTP, not doing
anything
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_DEPTH.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable orig_exten.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable
EXITCONTEXT.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable calls.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable peer.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ARG2.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable DIALNUM.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFNA.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFBS.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CFIM.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable COUNT.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ARG1.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_PRIORITY.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_CONTEXT.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable MACRO_EXTEN.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable
start.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Copying soft-transferable variable
intern.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable CALLEDTON.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable ANI2.
[Apr 15 00:02:34] DEBUG[8816] channel.c: Not copying variable
TRANSFERCAPABILITY.
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Outgoing Call for 236
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Call to peer '236' is 1 out of 10
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: Our T38 capability (0), joint T38
capability (0)
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: ** Our capability: 0x10e
(gsm|ulaw|alaw|g729) Video flag: False
[Apr 15 00:02:34] DEBUG[8816] chan_sip.c: ** Our prefcodec: 0x8 (alaw)
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- Called 236
[Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel SIP/236-081df8b0 to read
format slin
[Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel Zap/31-1 to write format
slin
[Apr 15 00:02:34] DEBUG[8816] channel.c: Set channel Zap/31-1 to read format
g729
[Apr 15 00:02:34] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '45c32fd1024523451dba563865cd0...@xxx.at' Request
102: Found
[Apr 15 00:02:34] VERBOSE[8816] logger.c: -- SIP/236-081df8b0 is ringing
[Apr 15 00:02:34] DEB
Re: [asterisk-users] Exit Dial Application
On Tue, Apr 14, 2009 at 11:11 PM, Christoph Fürstaller wrote: > Thanks for the hint. I've looked aht the full log. I've attached a snipplet > from the file. But I can't see anythin which can help me. Very interesting, > but not helpful for me : / Is it possible to deactivate the 'd' option? Or > what else could cause > my problem? > Ok, at first glance the app_macro looks suspicious, can You try calling dial without Macro? If unsuccessful, You could enable debug level 2, it will tell way much more of everything, including DTMF events etc. Btw, does DTMF work at all for this Zap/ line? You could verify that by using Read before Dial. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Atis Lezdins schrieb:
> That's CLI interface output, log should have timestamps and much more
> detail in it.
>
> Check /var/log/asterisk/full (assuming default install location).
> You'll need to enable "full" line in logger.conf, restart Asterisk and
> issue "core set verbose 3" and "core set debug 1" in CLI.
Thanks for the hint. I've looked aht the full log. I've attached a snipplet
from the file. But I can't see anythin which can help me. Very interesting, but
not helpful for me : / Is it possible to deactivate the 'd' option? Or what
else could cause
my problem?
>
>
> Regards,
> Atis
thanks for your help,
chris...
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing
[s-d...@macro-dialone:11] Set("Zap/31-1", "EXITCONTEXT=callback") in new stack
[Apr 14 22:49:25] DEBUG[7867] app_macro.c: Executed application: Set
[Apr 14 22:49:25] DEBUG[7867] pbx.c: Launching 'Set'
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing
[s-d...@macro-dialone:12] Set("Zap/31-1", "orig_exten=236") in new stack
[Apr 14 22:49:25] DEBUG[7867] app_macro.c: Executed application: Set
[Apr 14 22:49:25] DEBUG[7867] pbx.c: Launching 'Dial'
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Executing
[s-d...@macro-dialone:13] Dial("Zap/31-1", "SIP/236|30|dtT") in new stack
[Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Asked to create a SIP channel with
formats: 0x8 (alaw)
[Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Allocating new SIP dialog for (No
Call-ID) - INVITE (With RTP)
[Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Setting NAT on RTP to On
[Apr 14 22:49:25] DEBUG[7867] acl.c: # Testing 10.10.5.1 with 10.10.0.0
[Apr 14 22:49:25] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing
anything
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_DEPTH.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable orig_exten.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable EXITCONTEXT.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable calls.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable peer.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ARG2.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable DIALNUM.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFNA.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFBS.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CFIM.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable COUNT.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ARG1.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_PRIORITY.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_CONTEXT.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable MACRO_EXTEN.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Copying soft-transferable variable
start.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Copying soft-transferable variable
intern.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable CALLEDTON.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable ANI2.
[Apr 14 22:49:25] DEBUG[7867] channel.c: Not copying variable
TRANSFERCAPABILITY.
[Apr 14 22:49:25] DEBUG[7867] chan_sip.c: Outgoing Call for 236
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- Called 236
[Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel SIP/236-08219bb0 to read
format slin
[Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel Zap/31-1 to write format
slin
[Apr 14 22:49:25] DEBUG[7867] channel.c: Set channel Zap/31-1 to read format
g729
[Apr 14 22:49:25] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request
102: Found
[Apr 14 22:49:25] VERBOSE[7867] logger.c: -- SIP/236-08219bb0 is ringing
[Apr 14 22:49:25] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing
anything
[Apr 14 22:49:25] DEBUG[7867] chan_zap.c: Requested indication 3 on channel
Zap/31-1
[Apr 14 22:49:26] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request
102: Found
[Apr 14 22:49:26] VERBOSE[7867] logger.c: -- SIP/236-08219bb0 is ringing
[Apr 14 22:49:26] DEBUG[7867] rtp.c: Channel 'Zap/31-1' has no RTP, not doing
anything
[Apr 14 22:49:27] DEBUG[7616] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '65fc078c0c6cd24f5c068b770dabc...@xxx.at' Request
102: Found
[Apr 14 22:49:27] VERB
Re: [asterisk-users] Exit Dial Application
On Tue, Apr 14, 2009 at 9:14 PM, Christoph Fürstaller
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi Atis,
>
> No problem : ) I tried it again, here is the log output:
> -- Executing [...@from-pbx:1] Set("Zap/31-1", "EXITCONTEXT=callback") in
> new stack
> -- Executing [...@from-pbx:2] Dial("Zap/31-1", "SIP/236||d") in new stack
> -- Called 236
> -- SIP/236-0825f928 is ringing
> -- SIP/236-0825f928 is ringing
> -- SIP/236-0825f928 is ringing
> -- SIP/236-0825f928 is ringing
That's CLI interface output, log should have timestamps and much more
detail in it.
Check /var/log/asterisk/full (assuming default install location).
You'll need to enable "full" line in logger.conf, restart Asterisk and
issue "core set verbose 3" and "core set debug 1" in CLI.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
No problem : ) I tried it again, here is the log output:
-- Executing [...@from-pbx:1] Set("Zap/31-1", "EXITCONTEXT=callback") in
new stack
-- Executing [...@from-pbx:2] Dial("Zap/31-1", "SIP/236||d") in new stack
-- Called 236
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
-- SIP/236-0825f928 is ringing
Nothing happens. I adopted my [callback] context:
[callback]
exten => 1,1,Verbose(hello)
exten => s,1,Verbose(s)
exten => i,1,Verbose(i)
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
But nothing happens, if I dial 1, 5, or everything else. I have no clue what's
wrong here.
chris...
Atis Lezdins schrieb:
>> Thanks for your replay. But in my 1st post, I mentioned my dial statement:
>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>>
>> As you can see, there is a d to exit the dial application. And one priority
>> earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
>> doesn't : /
>>
>
> Oh, sorry, missed that part :)
>
> Try enabling "full" log in logger.conf, set verbosity to 3 and debug
> to 1, and see what goes in it.
>
> Regards,
> Atis
>
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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=7V9F
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Re: [asterisk-users] Exit Dial Application
>
> Thanks for your replay. But in my 1st post, I mentioned my dial statement:
> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>
> As you can see, there is a d to exit the dial application. And one priority
> earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
> doesn't : /
>
Oh, sorry, missed that part :)
Try enabling "full" log in logger.conf, set verbosity to 3 and debug
to 1, and see what goes in it.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
doesn't : /
Chris...
Atis Lezdins schrieb:
> CLI> core show application Dial
>
> d- Allow the calling user to dial a 1 digit extension while waiting
> for
>a call to be answered. Exit to that extension if it exists in the
>current context, or the context defined in the EXITCONTEXT
> variable,
>if it exists.
>
> Regards,
> Atis
>
> On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller
> wrote:
> Hi,
>
> Thanks for your replay. But this can only be done before or after the dial,
> but I wanna do it during the dial, when user A is waiting for user B,
> answering the phone. This should be possible, right?
>
> I hope anyone knows if this is possible.
>
> Chris...
>
> Danny Nicholas schrieb:
>>>> I'd change callback to this
>>>> [callback]
>>>> Exten => s,1,Playback(press5msg)
>>>> Exten => s,n,Waitexten(5)
>>>> Exten => s,n,Hangup
>>>> exten => 5,1,agi(str_concat.sh)
>>>> exten => 5,n,Hangup
>>>>
>>>> This will play a message, wait 5 seconds for user to press 5, then hangup
>>>> if
>>>> they don't.
>>>>
>>>> -Original Message-
>>>> From: asterisk-users-boun...@lists.digium.com
>>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
>>>> Fuerstaller
>>>> Sent: Tuesday, April 14, 2009 5:04 AM
>>>> To: Asterisk Users Mailing List
>>>> Subject: [asterisk-users] Exit Dial Application
>>>>
>>>> Hi,
>>>>
>>>> I' try to implement an automatic callback mechanism, just for local SIP
>>>> calls.. Callback
>>>> on busy and on no answer. If the other party doen't answer, it should be
>>>> possible to press
>>>> 5 to place an callback.
>>>>
>>>> Here is my dial:
>>>> exten => _X.,1,Set(EXITCONTEXT=callback)
>>>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>>>>
>>>> And here the script for callback.
>>>> [callback]
>>>> exten => 5,1,agi(str_concat.sh)
>>>> exten => 5,n,Hangup
>>>>
>>>> If I call someone and press 5, nothing happens. What could be a problem?
>>>> DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
>>>> I can enter
>>>> the voicmail menue.
>>>>
>>>> I'm using Asterisk 1.4.21.1.
>>>>
>>>> Any successions are very appreciated.
>>>>
>>>> Chris...
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>
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>
>>
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asterisk-users mailing list
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>>
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Re: [asterisk-users] Exit Dial Application
CLI> core show application Dial
d- Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
if it exists.
Regards,
Atis
On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> Thanks for your replay. But this can only be done before or after the dial,
> but I wanna do it during the dial, when user A is waiting for user B,
> answering the phone. This should be possible, right?
>
> I hope anyone knows if this is possible.
>
> Chris...
>
> Danny Nicholas schrieb:
>> I'd change callback to this
>> [callback]
>> Exten => s,1,Playback(press5msg)
>> Exten => s,n,Waitexten(5)
>> Exten => s,n,Hangup
>> exten => 5,1,agi(str_concat.sh)
>> exten => 5,n,Hangup
>>
>> This will play a message, wait 5 seconds for user to press 5, then hangup if
>> they don't.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
>> Fuerstaller
>> Sent: Tuesday, April 14, 2009 5:04 AM
>> To: Asterisk Users Mailing List
>> Subject: [asterisk-users] Exit Dial Application
>>
>> Hi,
>>
>> I' try to implement an automatic callback mechanism, just for local SIP
>> calls.. Callback
>> on busy and on no answer. If the other party doen't answer, it should be
>> possible to press
>> 5 to place an callback.
>>
>> Here is my dial:
>> exten => _X.,1,Set(EXITCONTEXT=callback)
>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>>
>> And here the script for callback.
>> [callback]
>> exten => 5,1,agi(str_concat.sh)
>> exten => 5,n,Hangup
>>
>> If I call someone and press 5, nothing happens. What could be a problem?
>> DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
>> I can enter
>> the voicmail menue.
>>
>> I'm using Asterisk 1.4.21.1.
>>
>> Any successions are very appreciated.
>>
>> Chris...
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
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> To UNSUBSCRIBE or update options visit:
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>
> - --
> commpany dialog solutions gmbh
>
> Dipl.-Ing.(FH) Christoph Fürstaller
> IP-Communications
>
> Ischlerbahnstraße 14, 5301 Eugendorf
> Tel: +43 662 879512 Fax: +43 662 875960
> IP-Tel: +43 780 commpany (26667269)
> Email: c.fuerstal...@commpany.at
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (MingW32)
>
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> 5dcAoIVK3L7pxTBLZrDi+kJGpOCPVa47
> =hEGE
> -END PGP SIGNATURE-
>
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>
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial, but
I wanna do it during the dial, when user A is waiting for user B, answering the
phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
> I'd change callback to this
> [callback]
> Exten => s,1,Playback(press5msg)
> Exten => s,n,Waitexten(5)
> Exten => s,n,Hangup
> exten => 5,1,agi(str_concat.sh)
> exten => 5,n,Hangup
>
> This will play a message, wait 5 seconds for user to press 5, then hangup if
> they don't.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
> Fuerstaller
> Sent: Tuesday, April 14, 2009 5:04 AM
> To: Asterisk Users Mailing List
> Subject: [asterisk-users] Exit Dial Application
>
> Hi,
>
> I' try to implement an automatic callback mechanism, just for local SIP
> calls.. Callback
> on busy and on no answer. If the other party doen't answer, it should be
> possible to press
> 5 to place an callback.
>
> Here is my dial:
> exten => _X.,1,Set(EXITCONTEXT=callback)
> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>
> And here the script for callback.
> [callback]
> exten => 5,1,agi(str_concat.sh)
> exten => 5,n,Hangup
>
> If I call someone and press 5, nothing happens. What could be a problem?
> DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
> I can enter
> the voicmail menue.
>
> I'm using Asterisk 1.4.21.1.
>
> Any successions are very appreciated.
>
> Chris...
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commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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=hEGE
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Re: [asterisk-users] Exit Dial Application
I'd change callback to this
[callback]
Exten => s,1,Playback(press5msg)
Exten => s,n,Waitexten(5)
Exten => s,n,Hangup
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten => _X.,1,Set(EXITCONTEXT=callback)
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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=K4sI
-END PGP SIGNATURE-
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[asterisk-users] Exit Dial Application
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I' try to implement an automatic callback mechanism, just for local SIP calls..
Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten => _X.,1,Set(EXITCONTEXT=callback)
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly, I
can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
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Version: GnuPG v2.0.11 (GNU/Linux)
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YTMAn0jDBdNOxsd5jjxBZ1yJ2J9HcCR5
=K4sI
-END PGP SIGNATURE-
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