[Voiceglue] Voiceglue Digest, Vol 33, Issue 6
Chris Matthieu
chris at getvocal.com
Thu Apr 8 22:01:21 UTC 2010
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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> 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>
> <sip%3A17476491417 at proxy01.sipphone.com<sip%253A17476491417 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<sip%3A17476491417 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>
> <sip%3A17476491417 at proxy01.sipphone.com<sip%253A17476491417 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>
> <sip%3A17476491417 at proxy01.sipphone.com<sip%253A17476491417 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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