[Voiceglue] Voiceglue Digest, Vol 33, Issue 6
Willian S. Domingues
willian at fiscalweb.com.br
Fri Apr 9 11:08:28 UTC 2010
You have FreePBX installed? Try to set Allow Anonymous Inbound SIP Calls
to Yes.
Willian
Em Qui, 2010-04-08 às 15:01 -0700, Chris Matthieu escreveu:
> SIP Transfer Follow-up: Problem solved - kind of...
>
> The hangup in my extensions.conf was causing the transfer to terminate
> before the connection was made. Commenting this line out, allows
> transfers to complete. The only problem now is that transfers are
> occurring before any of the VoiceXML in the script has a chance to
> process.
>
> One step closer :)
>
> Thanks,
> Chris
>
>
> On Thu, Apr 8, 2010 at 12:58 PM, <voiceglue-request at voiceglue.org>
> wrote:
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> Today's Topics:
>
> 1. Re: Spidermonkey issue..? (James Green)
> 2. VXML Transfers (Chris Matthieu)
> 3. Re: VXML Transfers (Chris Matthieu)
> 4. Re: VXML Transfers (Chris Matthieu)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 8 Apr 2010 16:25:40 +0100
> From: "James Green" <james.green at mjog.com>
> To: "General discussion about voiceglue"
> <voiceglue at voiceglue.org>
> Subject: Re: [Voiceglue] Spidermonkey issue..?
> Message-ID:
>
> <5144EF5B0ABC6745BEE64C74E43ABD8F01EDB0A0 at moneypenny.softoption.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Yet another follow-up.
>
> Remove the line:
> -DJSI_MUST_DECLARE_VARS \
> from openvxi-3.4+vglue/src/jsi/Makefile to make Spidermonkey
> operate in
> non-strict mode. It fixed the problem below, but did not
> enable the JSON
> native object.
>
> Naturally you must rebuild voiceglue having made this change.
>
> Is there a particular argument to having this in the current
> release, or
> might it be removed safely in future releases by default?
>
> ________________________________
>
> From: voiceglue-bounces at voiceglue.org
> [mailto:voiceglue-bounces at voiceglue.org] On Behalf Of James
> Green
> Sent: 08 April 2010 12:44
> To: General discussion about voiceglue
> Subject: Re: [Voiceglue] Spidermonkey issue..?
>
>
> Well I've switched from one 3rd party lib to another 3rd party
> lib with
> the same error:
>
> 12:33:27:289 EROR OPEN_VXI scaraman callid=[8005]
> |139901345446224|8005|SEVERE|swi:SBjsi|501|SBjsi: ECMAScript
> engine
> exception|errmsg=TypeError: function $ does not always return
> a
> value|line=139899969732909|linetxt=function
> $(c,t){t=c[m];delete
> c[m];try{e(c)}catch(z){c[m]=t;return 1}};|tokentxt=};
>
> ...Where '$' was 'str' in the first library.
>
> Something, somewhere, is not happy. I'm now concerned as to
> just what I
> can and cannot expect from using Javascript within voicexml
> using
> voiceglue.
>
> All I want to do is record a history of key=value pairs and
> submit them
> at the end as a list parsable within PHP. I hoped JSON might
> provide an
> easy path.
>
>
> ________________________________
>
> From: voiceglue-bounces at voiceglue.org
> [mailto:voiceglue-bounces at voiceglue.org] On Behalf Of James
> Green
> Sent: 08 April 2010 11:54
> To: General discussion about voiceglue
> Subject: [Voiceglue] Spidermonkey issue..?
>
>
> Hi,
>
> Apparently Voiceglue uses Spidermonkey to provide JavaScript.
> In our
> case we have 1.8.1 [1] which provides a JSON.stringify()
> method [2].
>
> Using this within voiceglue doesn't appear to work however:
>
> 11:40:58:191 EROR OPEN_VXI scaraman callid=[7995]
> |139901345446224|7995|SEVERE|swi:SBjsi|501|SBjsi: ECMAScript
> engine
> exception|errmsg=ReferenceError: JSON is not
> defined|line=139899969732609|linetxt=|tokentxt=
>
> Snippet from the script:
> <script>var packet =
> JSON.stringify(document.actions);</script>
>
> I was rather hoping not to have to import a 3rd party script,
> so if I'm
> doing something wrong could someone kindly point out the
> problem?
>
> Thanks,
>
> James
>
> [1] $ dpkg -l | grep libmoz
> ii libmozjs-dev
> 1.8.1.16+nobinonly-0ubuntu1 Development files for the
> Mozilla
> SpiderMonk
> ii libmozjs0d
> 1.8.1.16+nobinonly-0ubuntu1 The Mozilla SpiderMonkey
> JavaScript
> library
> [2] https://developer.mozilla.org/en/Using_native_JSON
>
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> Date: 04/08/10
> 07:32:00
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> ------------------------------
>
> Message: 2
> Date: Thu, 8 Apr 2010 10:43:30 -0700
> From: Chris Matthieu <chris at getvocal.com>
> To: voiceglue at voiceglue.org
> Subject: [Voiceglue] VXML Transfers
> Message-ID:
>
> <i2m5afd756d1004081043n60e60079yaef8e5a1fc5f838b at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Does Asterisk/VoiceGlue support SIP transfers? I am trying to
> transfer a
> SIP call to another SIP address and have not had any success
> with the
> following two examples:
>
> <?xml version="1.0" encoding="UTF-8"?>
> <vxml xmlns="http://www.w3.org/2001/vxml" xmlns:xsi="
> http://www.w3.org/2001/XMLSchema-instance"
> xsi:schemaLocation="
> http://www.w3.org/2001/vxml
> http://www.w3.org/TR/voicexml20/vxml.xsd"
> version="2.0" application="/root.vxml">
> <form id="form1">
> <block><prompt>Transferring call</prompt></block>
> <transfer dest="sip:17476491417 at proxy01.sipphone.com<sip%
> 3A17476491417 at proxy01.sipphone.com>"
> />
> </form>
> </vxml>
>
> OR
>
> <transfer dest="17476491417 at proxy01.sipphone.com" />
>
> Please advise.
>
> Thanks,
> Chris
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>
> ------------------------------
>
> Message: 3
> Date: Thu, 8 Apr 2010 12:25:03 -0700
> From: Chris Matthieu <chris at getvocal.com>
> To: voiceglue at voiceglue.org
> Subject: Re: [Voiceglue] VXML Transfers
> Message-ID:
>
> <x2j5afd756d1004081225k7ca4b675qa64183b101dabb89 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I ran across this article (
> http://www.voiceglue.org/pipermail/voiceglue/2009-September/000596.html) and
> added the following line to my voiceglue.conf file:
>
> blind_xfer_method = dial
>
> I have also experimented with the following transfer tag as
> referenced in
> the above link as well as with and without the leading '1' on
> the SIP
> address:
>
> <transfer name="blindTransfer" bridge="false"
> dest="sip:17476491417 at proxy01.sipphone.com|20|t" />
>
> No luck so far... Any help would be greatly appreciated.
>
> Thanks,
> Chris
>
>
> On Thu, Apr 8, 2010 at 10:43 AM, Chris Matthieu
> <chris at getvocal.com> wrote:
>
> > Does Asterisk/VoiceGlue support SIP transfers? I am trying
> to transfer a
> > SIP call to another SIP address and have not had any success
> with the
> > following two examples:
> >
> > <?xml version="1.0" encoding="UTF-8"?>
> > <vxml xmlns="http://www.w3.org/2001/vxml" xmlns:xsi="
> > http://www.w3.org/2001/XMLSchema-instance"
> xsi:schemaLocation="
> > http://www.w3.org/2001/vxml
> http://www.w3.org/TR/voicexml20/vxml.xsd"
> > version="2.0" application="/root.vxml">
> > <form id="form1">
> > <block><prompt>Transferring call</prompt></block>
> > <transfer dest="sip:17476491417 at proxy01.sipphone.com<sip%
> 3A17476491417 at proxy01.sipphone.com>"
> > />
> > </form>
> > </vxml>
> >
> > OR
> >
> > <transfer dest="17476491417 at proxy01.sipphone.com" />
> >
> > Please advise.
> >
> > Thanks,
> > Chris
> >
> >
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> ------------------------------
>
> Message: 4
> Date: Thu, 8 Apr 2010 12:58:31 -0700
> From: Chris Matthieu <chris at getvocal.com>
> To: voiceglue at voiceglue.org
> Subject: Re: [Voiceglue] VXML Transfers
> Message-ID:
>
> <j2y5afd756d1004081258h39429d11m3c55220da946a61 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I raised the Dynlog level to 7 and received the following
> output results on
> the transfer portion of the log. Notice the Network Out of
> Order message at
> the end of the call. Does something need to be configured in
> Asterisk to
> support SIP call transfers?
>
> 19:36:44:000 DBUG PHONGLUE domU-12- callid=[31] snd played
> callid=31
> status=0 msg="" reason=end-of-data to CT client on
> fh="::CTCLIENT1" at
> host=127.0.0.1 proto=SATC
> 19:36:44:000 DBUG PHONGLUE domU-12- snd "played 31 0 \"\" 1\n"
> to
> ::CTCLIENT1
> 19:36:44:001 DBUG VOICEGLU domU-12- rcv ctsrv: "played 31 0
> \"\" 1\n"
> 19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] parsed played
> callid=31
> status=0 msg="" reason=end-of-data
> 19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] dpal():
> do_prompt_and_listen() called
> 19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] dpal():
> Checking for prompts
> 19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] dpal(): Found
> prompts to
> play, playing
> 19:36:44:001 DBUG VOICEGLU domU-12- callid=[31] snd play
> callid=31
> files=("voiceglue/tts/Transferring_call")
> stopkeys="0123456789*#" to CT
> server on fh="::CTSRV" at host=localhost
> 19:36:44:001 DBUG VOICEGLU domU-12- snd "play 31
> \"voiceglue/tts/Transferring_call\" 0123456789*#\n" to ::CTSRV
> 19:36:44:001 DBUG PHONGLUE domU-12- rcv ct(::CTCLIENT1): "play
> 31
> \"voiceglue/tts/Transferring_call\" 0123456789*#\n"
> 19:36:44:002 DBUG PHONGLUE domU-12- callid=[31] parsed play
> callid=31
> files=("voiceglue/tts/Transferring_call")
> stopkeys="0123456789*#"
> 19:36:44:002 DBUG PHONGLUE domU-12- callid=[31] snd PLAYFILE
> file="voiceglue/tts/Transferring_call" stopkeys="0123456789*#"
> to AGI client
> on fh="::FASTAGI31" at host=127.0.0.1 callid=[31]
> 19:36:44:002 DBUG PHONGLUE domU-12- snd "STREAM FILE
> voiceglue/tts/Transferring_call 0123456789*#\n" to ::FASTAGI31
> 19:36:45:567 DBUG PHONGLUE domU-12- rcv agi(::FASTAGI31): "200
> result=0
> endpos=12537\n"
> 19:36:45:567 DBUG PHONGLUE domU-12- callid=[31] AGI result=0
> values:
> endpos=12537
> 19:36:45:567 DBUG PHONGLUE domU-12- callid=[31] snd played
> callid=31
> status=0 msg="" reason=end-of-data to CT client on
> fh="::CTCLIENT1" at
> host=127.0.0.1 proto=SATC
> 19:36:45:568 DBUG PHONGLUE domU-12- snd "played 31 0 \"\" 1\n"
> to
> ::CTCLIENT1
> 19:36:45:568 DBUG VOICEGLU domU-12- rcv ctsrv: "played 31 0
> \"\" 1\n"
> 19:36:45:568 DBUG VOICEGLU domU-12- callid=[31] parsed played
> callid=31
> status=0 msg="" reason=end-of-data
> 19:36:45:568 DBUG VOICEGLU domU-12- callid=[31] dpal():
> do_prompt_and_listen() called
> 19:36:45:568 DBUG VOICEGLU domU-12- callid=[31] dpal():
> Sending response to
> wait
> 19:36:45:569 DBUG VOICEGLU domU-12- callid=[31] dpal() Data
> Cleared
> 19:36:45:569 DBUG VOICEGLU domU-12- callid=[31] snd Waited to
> VXML
> interpreter on fh="::PERL_VXML_31" at host=localhost
> callid=[31]
> 19:36:45:569 DBUG VOICEGLU domU-12- snd "Waited\n"
> to ::PERL_VXML_31
> 19:36:45:571 DBUG VOICEGLU domU-12- callid=[31] rcv ovxi: "\n"
> 19:36:45:571 INFO VOICEGLU domU-12- callid=[31] had its VXML
> interpreter
> thread exit
> 19:36:45:571 DBUG VOICEGLU domU-12- callid=[31] deallocating
> VXML thread
> 19:36:45:571 NOTI VOICEGLU domU-12- callid=[31] hanging up
> because VXML
> interpreter exited
> 19:36:45:571 DBUG VOICEGLU domU-12- callid=[31] snd hangup to
> CT server on
> fh="::CTSRV" at host=localhost
> 19:36:45:571 DBUG VOICEGLU domU-12- snd "hangup 31\n"
> to ::CTSRV
> 19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31] rcv vg: Waited
> 19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31]
> |-1229898864|31|60001|testClient::ChannelThread|NULL result
> 19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31]
> |-1229898864|31|60001|testClient::ChannelThread|Call
> Terminated
> 19:36:45:572 NOTI OPEN_VXI domU-12- callid=[31] Channel 31:
> Call Terminated
> 19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31]
> |-1229898864|-1|3000|SBinetDestroyResource|entering:
> 0x0x8ba4e90
> (0x0x8afb5d0)
> 19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31]
> |-1229898864|-1|3000|SBinetDestroyResource|exiting, returned 0
> 19:36:45:572 DBUG OPEN_VXI domU-12- callid=[31] snd vg:
> 19:36:45:574 DBUG PHONGLUE domU-12- rcv ct(::CTCLIENT1):
> "hangup 31\n"
> 19:36:45:574 DBUG PHONGLUE domU-12- callid=[31] parsed hangup
> 19:36:45:574 INFO PHONGLUE domU-12- AGI client filehandle
> "::FASTAGI31"
> stopped: hangup requested
> 19:36:45:574 DBUG PHONGLUE domU-12- callid=[31] snd hungup
> callid=31 to CT
> client on fh="::CTCLIENT1" at host=127.0.0.1 proto=SATC
> 19:36:45:574 DBUG PHONGLUE domU-12- snd "hungup 31\n"
> to ::CTCLIENT1
> 19:36:45:574 DBUG PHONGLUE domU-12- rcv mgr(::ASTMGR): "Event:
> Newexten\r\nPrivilege: call,all\r\nChannel:
> SIP/sip2sip.info-086ee748\r\nContext: phoneglue\r\nExtension:
> 183\r\nPriority: 3\r\nApplication: Hangup\r\nAppData: \r
> \nUniqueid:
> asterisk-1270755393.30\r\n\r\n"
> 19:36:45:574 DBUG PHONGLUE domU-12- rcv mgr(::ASTMGR): "Event:
> Hangup\r\nPrivilege: call,all\r\nChannel:
> SIP/sip2sip.info-086ee748\r\nUniqueid: asterisk-1270755393.30
> \r\nCause:
> 38\r\nCause-txt: Network out of order\r\n\r\n"
> 19:36:45:574 DBUG VOICEGLU domU-12- rcv ctsrv: "hungup 31\n"
> 19:36:45:574 DBUG VOICEGLU domU-12- callid=[31] parsed hungup
> callid=31
>
> Thanks,
> Chris
>
> On Thu, Apr 8, 2010 at 10:43 AM, Chris Matthieu
> <chris at getvocal.com> wrote:
>
> > Does Asterisk/VoiceGlue support SIP transfers? I am trying
> to transfer a
> > SIP call to another SIP address and have not had any success
> with the
> > following two examples:
> >
> > <?xml version="1.0" encoding="UTF-8"?>
> > <vxml xmlns="http://www.w3.org/2001/vxml" xmlns:xsi="
> > http://www.w3.org/2001/XMLSchema-instance"
> xsi:schemaLocation="
> > http://www.w3.org/2001/vxml
> http://www.w3.org/TR/voicexml20/vxml.xsd"
> > version="2.0" application="/root.vxml">
> > <form id="form1">
> > <block><prompt>Transferring call</prompt></block>
> > <transfer dest="sip:17476491417 at proxy01.sipphone.com<sip%
> 3A17476491417 at proxy01.sipphone.com>"
> > />
> > </form>
> > </vxml>
> >
> > OR
> >
> > <transfer dest="17476491417 at proxy01.sipphone.com" />
> >
> > Please advise.
> >
> > Thanks,
> > Chris
> >
> >
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>
> End of Voiceglue Digest, Vol 33, Issue 6
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