[asterisk-users] asterisk-users Digest, Vol 73, Issue 58
Jonathan Leong
jonathan at e-numx.com
Fri Aug 27 09:05:51 CDT 2010
On 8/27/10, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
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> Today's Topics:
>
> 1. CDR on Transfer... (Carlos Chavez)
> 2. Re: Asterisk 1.6.1.17 ACK/BYE question (Trevor Benson)
> 3. Re: Use of AGISIGHUP (Danny Nicholas)
> 4. double DTMF digits (M S)
> 5. Re: double DTMF digits (Andres)
> 6. Re: Use of AGISIGHUP (Steve Edwards)
> 7. Re: Use of AGISIGHUP (Danny Nicholas)
> 8. Re: Use of AGISIGHUP (Steve Edwards)
> 9. Re: double DTMF digits (Matt Desbiens)
> 10. Asterisk 1.6 Displaying in Debug that it is playing a ulaw
> file using BackGround() but no audio is heard from the phone
> (Joe Wood)
> 11. Re: double DTMF digits (M S)
> 12. Re: Use of AGISIGHUP (Lee Archer)
> 13. dynamic MeetMe, min. digits (Xavier)
> 14. Re: dynamic MeetMe, min. digits (Doug Lytle)
> 15. Re: dynamic MeetMe, min. digits (Xavier D.)
> 16. music on hold in blind transfer (Tino)
> 17. queue agent and blind transfer (Tino)
> 18. Call Forwarding (Dan Journo)
> 19. Re: music on hold in blind transfer (Paul Belanger)
> 20. Re: Call Forwarding (Stefan Schmidt)
> 21. Duplicate channel variables after transfer (Alex Hermann)
> 22. Re: CDR on Transfer... (Andra?)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 26 Aug 2010 12:25:07 -0500
> From: Carlos Chavez <cursor at telecomabmex.com>
> Subject: [asterisk-users] CDR on Transfer...
> To: Asterisk <asterisk-users at lists.digium.com>
> Message-ID: <1282843507.2830.13.camel at cursor.telecomabmex.com>
> Content-Type: text/plain; charset="utf-8"
>
> I have searched for some time but I have not found an asnwer on how to
> fix the CDR when a call is transferred. The problem is that if someone
> dials a cell phone and then transfers the call to another extensi?n the
> CDR for the cell call stops and there is no way to track that the call
> was transferred so we can bill correctly. Many people have asked this
> question but there is no answer, only a mention that it should be fixed
> in 1.6 which it is not (at least on 1.6.2.11).
>
> Any tips oh how to correct this problem? A lot of customers give me
> grief about this because they cannot properly bill people for their cell
> calls.
>
> --
> Telecomunicaciones Abiertas de M?xico S.A. de C.V.
> Carlos Ch?vez Prats
> Director de Tecnolog?a
> +52-55-91169161 ext 2001
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> ------------------------------
>
> Message: 2
> Date: Thu, 26 Aug 2010 10:30:16 -0700
> From: Trevor Benson <tbenson at a-1networks.com>
> Subject: Re: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <65F20266-3E46-4DCC-A17D-D181F8E4A6AE at a-1networks.com>
> Content-Type: text/plain; charset="windows-1252"
>
> We have a box running 1.6.2.11 on CentOS 5 using the RPM's from the Digium
> CentOS repository. We just left a 60 second voicemail on the system and had
> the full audio as well in the inbox. Not sure how your SIP configuration
> ties your SBC in, but native "users" created via users.conf and sip.conf
> appears to be working for me. Wouldnt be able to test more without knowing
> what settings you had between Asterisk and the SBC.
>
>
> --
> Trevor Benson
> dCAP, LPIC-1, CLA, Network+, MCP, CNA
> A1 Networks - Network Engineer
> DID (707)703-1041
> FAX (707)703-1983
>
>
>
>
>
>
> On Aug 26, 2010, at 8:47 AM, Steven C. Blair wrote:
>
>>
>> As a test we built Asterisk v1.6.2.11 on a new server. This version of
>> Asterisk exhibits the same behavior. From ngrep?s perspective we see an
>> ACK followed immediately by a BYE message. The user hears the recording
>> being played, begins to leave a message and is disconnected about 10
>> seconds into the call.
>>
>>
>>
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven C.
>> Blair
>> Sent: Wednesday, August 25, 2010 2:08 PM
>> To: asterisk-users at lists.digium.com
>> Subject: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question
>>
>>
>> We?re running Asterisk 1.6.1.17 for our campus voicemail server and
>> Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are
>> diverted to voicemail using a 302 redirect when the called party doesn?t
>> answer. In this case the caller is able to hear the greetings and begin to
>> leave a message only to have Asterisk terminate the call mid-recording.
>>
>> We?re uncertain why this is happening and this is where we are hoping you
>> can help. In our environment the caller is any set on the PSTN. They call
>> one of our IP phones which no one answers. Our proxy, SER, responds to the
>> SBC with a 302 redirect and the call is diverted to Asterisk. The caller
>> hears the unavailable greeting for 6-4050, begins to leave a message and
>> is cut-off after about 10 seconds. In an ngrep trace we see Asterisk
>> receive an ACK from the SBC and it immediately responds with a BYE message
>> for that call.
>>
>> Has anyone else experienced this type of issue?
>>
>>
>> ---
>>
>> ISC Networking & Telecommunications
>> 3401 Walnut Street, Suite 221A
>> Philadelphia, PA 19104
>> 215-573-8396
>> 215-898-9348 (fax)
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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> ------------------------------
>
> Message: 3
> Date: Thu, 26 Aug 2010 12:58:46 -0500
> From: "Danny Nicholas" <danny at debsinc.com>
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <201008261730.o7QHULt4029526 at mail.debsinc.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Can you post the CLI output showing the hangup/script failure?
>
>
>
> _____
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lee Archer
> Sent: Thursday, August 26, 2010 11:39 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Use of AGISIGHUP
>
>
>
> Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but it
> doesn't seem to be doing anything as the script is still exiting on a hangup
> and not completing properly. I am using 1.4.35 and have tried various
> combinations. Can anyone shed any light on this?
>
> Regards
>
> Lee
>
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> ------------------------------
>
> Message: 4
> Date: Thu, 26 Aug 2010 14:55:50 -0400
> From: M S <101mcs at gmail.com>
> Subject: [asterisk-users] double DTMF digits
> To: asterisk-users at lists.digium.com
> Message-ID:
> <AANLkTinoT4+HrPBHdKAx6WOU-vtyUPq3KcH9vRAE=E53 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> I've been getting complaints lately that callers to my IVR are pressing a
> digit once but the system is responding as if they pressed it twice (once
> for each of two consecutive menus).
> I'm using an AGI script and logging all DTMF entries - and to the script, at
> least, it looks like the digit is being pressed twice. The TN being called
> is a VOIP number (provided by Flowroute) and being forwarded via SIP to my
> asterisk 1.6.2.4 server. The dtmfmode is set to rfc28333 in sip.conf.
>
> The first time this happened, I figured the caller pressed the number twice
> without realizing it. It's happening to too many people for that to be
> plausible anymore. I also experienced it once myself, months ago, when I
> entered my tn as 1234567890 and had it read back to me as 1122334455.
>
> Can anyone give me some pointers where to start troubleshooting? Can
> overloading a system cause such an error?
>
> Thanks,
> Mira
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> ------------------------------
>
> Message: 5
> Date: Thu, 26 Aug 2010 15:23:44 -0400
> From: Andres <andres at telesip.net>
> Subject: Re: [asterisk-users] double DTMF digits
> To: asterisk-users at lists.digium.com
> Message-ID: <4C76BF40.20204 at telesip.net>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 8/26/2010 2:55 PM, M S wrote:
>> Hi,
>>
>> I've been getting complaints lately that callers to my IVR are
>> pressing a digit once but the system is responding as if they pressed
>> it twice (once for each of two consecutive menus).
>> I'm using an AGI script and logging all DTMF entries - and to the
>> script, at least, it looks like the digit is being pressed twice. The
>> TN being called is a VOIP number (provided by Flowroute) and being
>> forwarded via SIP to my asterisk 1.6.2.4 server. The dtmfmode is set
>> to rfc28333 in sip.conf.
>>
>> The first time this happened, I figured the caller pressed the number
>> twice without realizing it. It's happening to too many people for
>> that to be plausible anymore. I also experienced it once myself,
>> months ago, when I entered my tn as 1234567890 and had it read back to
>> me as 1122334455.
>>
>> Can anyone give me some pointers where to start troubleshooting? Can
>> overloading a system cause such an error?
>>
>> Thanks,
> I have seen this before. Upon careful analisys we saw that the far end
> was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't
> remember). Thus Asterisk detected double digits. The solution was to
> ask the remote end to only send RFC2833.
>
> Andres
> http://www.telesip.net
>
>
>
> ------------------------------
>
> Message: 6
> Date: Thu, 26 Aug 2010 12:41:25 -0700 (PDT)
> From: Steve Edwards <asterisk.org at sedwards.com>
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <alpine.DEB.2.00.1008261233160.25774 at localhost.localdomain>
> Content-Type: text/plain; charset="iso-8859-7"
>
> On Thu, 26 Aug 2010, Lee Archer wrote:
>
>> Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
>> it doesn?t seem to be doing anything as the script is still exiting on a
>> hangup and not completing properly.? I am using 1.4.35 and have tried
>> various combinations.? Can anyone shed any light on this?
>
> I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a
> bad idea to protect lazy programmers :)
>
> Program defensively!
>
> Trap the HUP and do what is appropriate for your script -- even if that is
> to ignore it.
>
> If the successful execution of your system depends on a setting, how long
> will it take the next guy to figure out when the setting disappears
> unexpectedly?
>
> --
> Thanks in advance,
> -------------------------------------------------------------------------
> Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> ------------------------------
>
> Message: 7
> Date: Thu, 26 Aug 2010 14:52:53 -0500
> From: "Danny Nicholas" <danny at debsinc.com>
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <201008261924.o7QJOREn030652 at mail.debsinc.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
>>From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Edwards
>>Subject: Re: [asterisk-users] Use of AGISIGHUP
>
>>On Thu, 26 Aug 2010, Lee Archer wrote:
>
>>> Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
>>> it doesn?t seem to be doing anything as the script is still exiting on a
>>> hangup and not completing properly.? I am using 1.4.35 and have tried
>>> various combinations.? Can anyone shed any light on this?
>
>>I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a
> bad idea to protect lazy programmers :)
>
>>Program defensively!
>
>>Trap the HUP and do what is appropriate for your script -- even if that is
> to ignore it.
>
>>If the successful execution of your system depends on a setting, how long
> will it take the next guy to figure out when the setting disappears
> unexpectedly?
>
> As usual, Steve is right. Here's a one-liner that should "fix" the problem
>
> local $SIG{HUP} = 'IGNORE';
>
> Does that make me lazy?
>
> TIA.
>
>
>
>
> ------------------------------
>
> Message: 8
> Date: Thu, 26 Aug 2010 13:02:20 -0700 (PDT)
> From: Steve Edwards <asterisk.org at sedwards.com>
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <alpine.DEB.2.00.1008261257210.15301 at localhost.localdomain>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
>>> On Thu, 26 Aug 2010, Lee Archer wrote:
>>
>>>> I am setting AGISIGHUP=no before running a Perl script via AGI but it
>>>> doesn?t seem to be doing anything as the script is still exiting on a
>>>> hangup and not completing properly.
>
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
>> Edwards
>>
>>> I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a
>>> bad idea to protect lazy programmers :)
>
> On Thu, 26 Aug 2010, Danny Nicholas wrote:
>
>> Here's a one-liner that should "fix" the problem
>>
>> local $SIG{HUP} = 'IGNORE';
>>
>> Does that make me lazy?
>
> Not at all. If that is the correct "response" for your program, it's
> perfect:
>
> 1) The program will execute correctly on your system, my system, any
> system regardless of the configuration.
>
> 2) It tells the next guy explicitly what you intended to happen upon
> receiving the signal.
>
> --
> Thanks in advance,
> -------------------------------------------------------------------------
> Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
>
>
> ------------------------------
>
> Message: 9
> Date: Thu, 26 Aug 2010 16:51:04 -0400
> From: Matt Desbiens <desbiensm at gmail.com>
> Subject: Re: [asterisk-users] double DTMF digits
> To: andres at telesip.net, Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTinfR_xKtGAnOUn4TMfPm6Afhq=GWLAyFf5ZA-_d at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> We've actually had issues with Flowroute in the past where DTMF was a
> constant issue. My best suggestion for course of action is find another
> provider. NexVortex is pretty solid all around. They also had the quickest
> recourse for when GNAPS went bottoms up last month and sent pretty much all
> VoIP traffic in New England into a tailspin.
>
> --Matt
>
> On Thu, Aug 26, 2010 at 3:23 PM, Andres <andres at telesip.net> wrote:
>
>> On 8/26/2010 2:55 PM, M S wrote:
>> > Hi,
>> >
>> > I've been getting complaints lately that callers to my IVR are
>> > pressing a digit once but the system is responding as if they pressed
>> > it twice (once for each of two consecutive menus).
>> > I'm using an AGI script and logging all DTMF entries - and to the
>> > script, at least, it looks like the digit is being pressed twice. The
>> > TN being called is a VOIP number (provided by Flowroute) and being
>> > forwarded via SIP to my asterisk 1.6.2.4 server. The dtmfmode is set
>> > to rfc28333 in sip.conf.
>> >
>> > The first time this happened, I figured the caller pressed the number
>> > twice without realizing it. It's happening to too many people for
>> > that to be plausible anymore. I also experienced it once myself,
>> > months ago, when I entered my tn as 1234567890 and had it read back to
>> > me as 1122334455.
>> >
>> > Can anyone give me some pointers where to start troubleshooting? Can
>> > overloading a system cause such an error?
>> >
>> > Thanks,
>> I have seen this before. Upon careful analisys we saw that the far end
>> was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't
>> remember). Thus Asterisk detected double digits. The solution was to
>> ask the remote end to only send RFC2833.
>>
>> Andres
>> http://www.telesip.net
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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> ------------------------------
>
> Message: 10
> Date: Thu, 26 Aug 2010 18:58:31 -0700
> From: Joe Wood <schmoe at gmail.com>
> Subject: [asterisk-users] Asterisk 1.6 Displaying in Debug that it is
> playing a ulaw file using BackGround() but no audio is heard from the
> phone
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTin5Axv=J7MOmCGoxeLQPbdNnouKXKyyJT=zdZcN at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> First off, let me first say that this is not a one-way audio problem.
> Sometimes I can get 'her' to speak to me, other times I can't for a
> long time.
>
> I'm just using a very simple system to dump people into MeetMe().
> Nothing fancy.
>
> I do have the following in my modules.conf:
>
> preload => format_mp3.so
> preload => codec_ulaw.so
> preload => format_pcm.so
>
> My extensions.conf looks like:
>
> [general]
> autofallthrough=yes
> static=no
> writeprotect=no
> extenpatternmatchnew=yes
> clearglobalvars=no
>
>
> [conference-calls]
> exten => s,1,Answer()
> exten => s,n,Background(welcome)
> exten => s,n,Background(and)
> exten => s,n,Background(thank-you-for-calling)
> exten => s,n,Background(conference-reservations)
> exten => s,n,Wait(2)
> exten => s,n,Background(enter-conf-pin-number)
> exten => s,n,WaitExten(10)
> exten => i,1,Playback(pbx-invalid)
> exten => i,n,Goto(conference-calls,9000,1)
> exten => t,1,Playback(vm-goodbye)
> exten => t,n,Hangup()
>
> exten => ${EXTEN},1,Meetme(${EXTEN})
>
>
> == Using SIP RTP CoS mark 5
> -- Executing [s at conference-calls:1]
> Answer("SIP/2063161626-00000001", "") in new stack
> == Using SIP RTP CoS mark 5
> -- Executing [s at conference-calls:1]
> Answer("SIP/2063161626-00000002", "") in new stack
> -- Executing [s at conference-calls:2]
> BackGround("SIP/2063161626-00000001", "welcome") in new stack
> -- <SIP/2063161626-00000001> Playing 'welcome.ulaw' (language 'en')
> -- Executing [s at conference-calls:2]
> BackGround("SIP/2063161626-00000002", "welcome") in new stack
> -- <SIP/2063161626-00000002> Playing 'welcome.ulaw' (language 'en')
> -- Executing [s at conference-calls:3]
> BackGround("SIP/2063161626-00000001", "and") in new stack
> -- <SIP/2063161626-00000001> Playing 'and.ulaw' (language 'en')
> -- Executing [s at conference-calls:3]
> BackGround("SIP/2063161626-00000002", "and") in new stack
> -- <SIP/2063161626-00000002> Playing 'and.ulaw' (language 'en')
> -- Executing [s at conference-calls:4]
> BackGround("SIP/2063161626-00000001", "thank-you-for-calling") in new
> stack
> -- <SIP/2063161626-00000001> Playing 'thank-you-for-calling.ulaw'
> (language 'en')
> -- Executing [s at conference-calls:4]
> BackGround("SIP/2063161626-00000002", "thank-you-for-calling") in new
> stack
> -- <SIP/2063161626-00000002> Playing 'thank-you-for-calling.ulaw'
> (language 'en')
> -- Executing [s at conference-calls:5]
> BackGround("SIP/2063161626-00000001", "conference-reservations") in
> new stack
> -- <SIP/2063161626-00000001> Playing
> 'conference-reservations.ulaw' (language 'en')
> -- Executing [s at conference-calls:5]
> BackGround("SIP/2063161626-00000002", "conference-reservations") in
> new stack
> -- <SIP/2063161626-00000002> Playing
> 'conference-reservations.ulaw' (language 'en')
> -- Executing [s at conference-calls:6]
> Wait("SIP/2063161626-00000001", "2") in new stack
> -- Executing [s at conference-calls:6]
> Wait("SIP/2063161626-00000002", "2") in new stack
> -- Executing [s at conference-calls:7]
> BackGround("SIP/2063161626-00000001", "enter-conf-pin-number") in new
> stack
> -- <SIP/2063161626-00000001> Playing 'enter-conf-pin-number.ulaw'
> (language 'en')
> -- Executing [s at conference-calls:7]
> BackGround("SIP/2063161626-00000002", "enter-conf-pin-number") in new
> stack
> -- <SIP/2063161626-00000002> Playing 'enter-conf-pin-number.ulaw'
> (language 'en')
> -- Executing [s at conference-calls:8]
> WaitExten("SIP/2063161626-00000001", "10") in new stack
> -- Executing [s at conference-calls:8]
> WaitExten("SIP/2063161626-00000002", "10") in new stack
> -- Timeout on SIP/2063161626-00000001, going to 't'
> -- Executing [t at conference-calls:1]
> Playback("SIP/2063161626-00000001", "vm-goodbye") in new stack
> -- <SIP/2063161626-00000001> Playing 'vm-goodbye.ulaw' (language 'en')
> -- Timeout on SIP/2063161626-00000002, going to 't'
> -- Executing [t at conference-calls:1]
> Playback("SIP/2063161626-00000002", "vm-goodbye") in new stack
> -- <SIP/2063161626-00000002> Playing 'vm-goodbye.ulaw' (language 'en')
> -- Executing [t at conference-calls:2]
> Hangup("SIP/2063161626-00000001", "") in new stack
> == Spawn extension (conference-calls, t, 2) exited non-zero on
> 'SIP/2063161626-00000001'
> -- Executing [t at conference-calls:2]
> Hangup("SIP/2063161626-00000002", "") in new stack
> == Spawn extension (conference-calls, t, 2) exited non-zero on
> 'SIP/2063161626-00000002'
>
> Has anyone else encountered this problem before? I saw one posting on
> the listserv, but it said to add in the pcm lib and I did that and no
> change.
>
> Help.
>
> Thanks a bunch,
>
> Joe
>
>
>
> ------------------------------
>
> Message: 11
> Date: Thu, 26 Aug 2010 22:25:37 -0400
> From: M S <101mcs at gmail.com>
> Subject: Re: [asterisk-users] double DTMF digits
> To: asterisk-users at lists.digium.com
> Message-ID:
> <AANLkTimn9i+Sxf_qKLTWFurohU7R6u-tx58iahgjDfns at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> How were you able to determine that the far end was sending the digits in
> RFC2833 plus SIP INFO?
>
> On Thu, Aug 26, 2010 at 3:23 PM, Andres <andres at telesip.net> wrote:
>
>>
>> I have seen this before. Upon careful analisys we saw that the far end
>> was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't
>> remember). Thus Asterisk detected double digits. The solution was to
>> ask the remote end to only send RFC2833.
>>
>> Andres
>> http://www.telesip.net
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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>
> ------------------------------
>
> Message: 12
> Date: Fri, 27 Aug 2010 09:36:48 +0100
> From: "Lee Archer" <Lee.Archer at thebigword.com>
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <B916037C74E10442982E494BA3F108F60D2E301D at MAIL1.thebigword.com>
> Content-Type: text/plain; charset="US-ASCII"
>
> Thanks for the replies. I am already ignoring the signal but it doesn't
> seem to be making much difference on this system with this script. It's
> an old legacy script I should hopefully be dropping and writing properly
> within the dial plan.
>
> I will keep trying!
>
> Thanks
>
> Lee
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
> Edwards
> Sent: 26 August 2010 21:02
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Use of AGISIGHUP
>
>>> On Thu, 26 Aug 2010, Lee Archer wrote:
>>
>>>> I am setting AGISIGHUP=no before running a Perl script via AGI but
>>>> it doesn?t seem to be doing anything as the script is still exiting
>>>> on a hangup and not completing properly.
>
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
>> Edwards
>>
>>> I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's
>
>>> a bad idea to protect lazy programmers :)
>
> On Thu, 26 Aug 2010, Danny Nicholas wrote:
>
>> Here's a one-liner that should "fix" the problem
>>
>> local $SIG{HUP} = 'IGNORE';
>>
>> Does that make me lazy?
>
> Not at all. If that is the correct "response" for your program, it's
> perfect:
>
> 1) The program will execute correctly on your system, my system, any
> system regardless of the configuration.
>
> 2) It tells the next guy explicitly what you intended to happen upon
> receiving the signal.
>
> --
> Thanks in advance,
> ------------------------------------------------------------------------
> -
> Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867
> PST
> Newline Fax:
> +1-760-731-3000
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ------------------------------
>
> Message: 13
> Date: Fri, 27 Aug 2010 11:27:57 +0200
> From: Xavier <magicrhesus at ouranos.be>
> Subject: [asterisk-users] dynamic MeetMe, min. digits
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4C77851D.4090802 at ouranos.be>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi All,
>
> Is there a way to use the dynamic feature of the meetme application (D)
> and to set an option to configure the minimum length of the numbers for
> the conference and the associated pin.
> In my case, I'd like them to be at least four digits.
>
> Thanks in advance !
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> ------------------------------
>
> Message: 14
> Date: Fri, 27 Aug 2010 05:58:57 -0400
> From: Doug Lytle <support at drdos.info>
> Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4C778C61.4080104 at drdos.info>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Xavier wrote:
>> Hi All,
>>
>> Is there a way to use the dynamic feature of the meetme application
>> (D) and to set an option to configure the minimum length of the
>> numbers for the conference and the associated pin.
>
> You can use the read application to get the password and then check the
> length, before going onto the conference setup.
>
>
>
> Doug
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
>
>
> ------------------------------
>
> Message: 15
> Date: Fri, 27 Aug 2010 12:28:38 +0200
> From: "Xavier D." <magicrhesus at ouranos.be>
> Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4C779356.1070405 at ouranos.be>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Yes but what about the conference number ?
>
> On 08/27/2010 11:58 AM, Doug Lytle wrote:
>> Xavier wrote:
>>> Hi All,
>>>
>>> Is there a way to use the dynamic feature of the meetme application
>>> (D) and to set an option to configure the minimum length of the
>>> numbers for the conference and the associated pin.
>> You can use the read application to get the password and then check the
>> length, before going onto the conference setup.
>>
>>
>>
>> Doug
>>
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>
> ------------------------------
>
> Message: 16
> Date: Fri, 27 Aug 2010 17:09:33 +0530
> From: Tino <tino at sparksupport.com>
> Subject: [asterisk-users] music on hold in blind transfer
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTikFKs7JCW-vKObVO6cBW0Fc+-KoC9ndHSO1pC1t at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello,
>
> Is it possible to avoid playing music on hold during a blind transfer ?
>
> Thanks
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>
> ------------------------------
>
> Message: 17
> Date: Fri, 27 Aug 2010 17:35:26 +0530
> From: Tino <tino at sparksupport.com>
> Subject: [asterisk-users] queue agent and blind transfer
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTinmCcg5f5EbjGO4moeNu7Cd4o6tz5f-fuMb+0S5 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello,
>
> When an agent does a blind transfer the call hangups for him but shows as
> "In use" in queue in my CRM (used for auto dialing). As a result the agent
> have to wait until the transfered call completes. Is there any way to change
> this behaviour ?
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>
> ------------------------------
>
> Message: 18
> Date: Fri, 27 Aug 2010 08:51:04 -0400
> From: Dan Journo <dan at keshercommunications.com>
> Subject: [asterisk-users] Call Forwarding
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <31C6BA8C3525D840B022617ACBB7BC036FE20831FF at VMBX123.ihostexchange.net>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi,
>
> I'm currently programming an interface for my Asterisk service.
>
> I've noticed an issue if someone sets up call forwarding on their sip phone.
> Asterisk receives a 302 "Moved Temporarily" message, and forwards the call
> successfully.
>
> However, the CDR is not correct.
>
> If I set up call forwarding to a mobile on extension 201, and then place a
> call from extension 202, the call gets diverted.
> I answer the call and talk for 30 seconds, then I hang up.
>
> The CDR shows two calls:-
>
> 2010-08-27 13:38:24 - 202 -> 201 - Answered - Billsec is 30
> 2010-08-27 13:38:24 - 202 -> 5551234 - Answered - Billsec is 0
>
> 5551234 is the mobile number.
> The second CDR entry should read 30 seconds, and the first should read 0 (or
> 30)
>
> Since it isn't behaving like I want, is there any way to disable the feature
> that allows a SIP phone to perform call forwarding?
>
> Thanks
> Dan
>
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>
> ------------------------------
>
> Message: 19
> Date: Fri, 27 Aug 2010 08:52:58 -0400
> From: Paul Belanger <paul.belanger at polybeacon.com>
> Subject: Re: [asterisk-users] music on hold in blind transfer
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTimnsLn2Y8T1venZrnbXQ1A8JGTvDUN7XiO0oNQn at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On Fri, Aug 27, 2010 at 7:39 AM, Tino <tino at sparksupport.com> wrote:
>> Is it possible to avoid playing music on hold during a blind transfer ?
>>
> Please do not cross-post the same message to multiple lists.
>
> Yes, configure an empty MoH class or not loading MoH are some options, also:
>
> *CLI> core show application Dial
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
>
>
> ------------------------------
>
> Message: 20
> Date: Fri, 27 Aug 2010 15:13:22 +0200
> From: Stefan Schmidt <sst at sil.at>
> Subject: Re: [asterisk-users] Call Forwarding
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4C77B9F2.5030900 at sil.at>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Dan Journo schrieb:
>>
>>
>>
>> Since it isn't behaving like I want, is there any way to disable the
>> feature that allows a SIP phone to perform call forwarding?
>>
>>
>>
>> Thanks
>>
>> Dan
>>
>>
>>
> Hello,
>
> in asterisk 1.6.x there is a Dial option i which suppress a 302 redirect
> which is very nice when dialing more than one phone at once, but you can
> use it also if you just dial one channel.
>
> see output of core show application dial:
>
> i - Asterisk will ignore any forwarding requests it may receive on
> this
> dial attempt.
>
>
> best regards
>
> steve
>
> --
> F?r weitere Fragen stehen wir gerne unter voip at sil.at oder
> 059944 - 2440 zur Verf?gung.
>
> Mit freundlichen Gr?ssen
> --
> Stefan Schmidt
> Sysadmin/VOIP // voip at sil.at // Tel 059944-2440//
> -------------------------------------------------
> SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 //
> A-1160 Wien // Fax 059944-9000 // www.sil.at //
> -------------------------------------------------
>
>
>
>
> ------------------------------
>
> Message: 21
> Date: Fri, 27 Aug 2010 15:13:54 +0200
> From: Alex Hermann <alex at speakup.nl>
> Subject: [asterisk-users] Duplicate channel variables after transfer
> To: asterisk-users at lists.digium.com
> Message-ID: <201008271513.54789.alex at speakup.nl>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi all,
>
>
> with an (attended) transfer i see the following happening:
>
> 1) A calls B1
> 2) B2 calls C
> 3) B2 transfers call to A
> 4) A talks to C
>
>
> At step 3, the channel A is connected to channel C and B1 and B2 are hung
> up.
> In the h extension for channel B2, the channel is renamed to B2<ZOMBIE> and
> i
> see that the channel variables of A have been merged into B2<ZOMBIE>. If
> there
> were duplicate names for variables, the channel now has those variables
> doubled. The DumpChan() application called from the h extension confirms
> this.
>
> In my case the channels are all SIP channels and in the h extension I want
> to
> access the SIPCALLID variable of the A channel. Every access to it gives me
> the wrong value namely that of channel B1. How do i access the _second_
> variable named SIPCALLID in the channel?
>
> Extract from DumpChan() as an example:
>
> Dumping Info For Channel: SIP/sipout-00000055<ZOMBIE>:
> ================================================================================
> Info:
> Name= SIP/sipout-00000055<ZOMBIE>
> Type= SIP
> UniqueID= 1282913436.108
> ....
> Variables:
> ...
> SIPCALLID=eae94252-ebf238ff at 172.28.4.112
> ...
> SIPCALLID=lyvkqtybsgrtsnh at 172.28.4.113
> ...
> ================================================================================
>
>
> I want to get lyvkqtybsgrtsnh at 172.28.4.113 instead of eae94252-
> ebf238ff at 172.28.4.112 as a result.
>
> --
> Greetings,
>
> Alex Hermann
>
>
>
>
> ------------------------------
>
> Message: 22
> Date: Fri, 27 Aug 2010 15:46:44 +0200
> From: Andra? <atletek at gmail.com>
> Subject: Re: [asterisk-users] CDR on Transfer...
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTim2m+fkmnS5UpGrQwQTAKpbB-N_wa29XyRK9-Qm at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Did you find the solution?
>
> On Thu, Aug 26, 2010 at 7:25 PM, Carlos Chavez
> <cursor at telecomabmex.com>wrote:
>
>> I have searched for some time but I have not found an asnwer on how
>> to
>> fix the CDR when a call is transferred. The problem is that if someone
>> dials a cell phone and then transfers the call to another extensi?n the
>> CDR for the cell call stops and there is no way to track that the call
>> was transferred so we can bill correctly. Many people have asked this
>> question but there is no answer, only a mention that it should be fixed
>> in 1.6 which it is not (at least on 1.6.2.11).
>>
>> Any tips oh how to correct this problem? A lot of customers give
>> me
>> grief about this because they cannot properly bill people for their cell
>> calls.
>>
>> --
>> Telecomunicaciones Abiertas de M?xico S.A. de C.V.
>> Carlos Ch?vez Prats
>> Director de Tecnolog?a
>> +52-55-91169161 ext 2001
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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>
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>
> AstriCon 2010 - October 26-28 Washington, DC
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> End of asterisk-users Digest, Vol 73, Issue 58
> **********************************************
>
--
Sent from my mobile device
Your kind advice is highly appreciated.
Warmest regards,
Jonathan Leong
Chief Executive Officer
eSky Multimedia Sdn. Bhd.
Address: 51-01-02 Jalan Austin Heights 3, Taman Mount Austin 81100
Johor Bahru, Johor, Malaysia
R&D Lab: Esky Multimedia Resources Lab, AR0008 (Faculty of
Engineering, MMU), Persiaran Multimedia 63100 Cyberjaya, Selangor,
Malaysia
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web : www.e-numX.com
e-numX : 8881000-2288
e-mail: jonathan at e-numx.com
Tel : +6.07.352.7777
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