ASterisk CDR file Master.csv
How can we set the CDR Master file to rollover at say 30 Meg and create a new one
* You are viewing the archive for August, 2010
Protect yourself
Tags: address, asterisk, firewall, Hey, intrusion attempts, linux, switches, today
Hey all
We are seeing intrusion attempts coming from address 201.47.236.122 today
They were hitting our switches trying to get in. So we blocked them at our
firewall.
Just wanted to put the word out so you all can protect your self.
Bryant
Asterisk DTMF RFC2833 issues
Tags: asterisk, audiocodes, broadvox, dev board, linux, network operations center
Hi all
I have posted a question on the asterisk dev board about this issue but I
want to see if any users have run up against this.
This issue is that when calls are run through Broadvox and Level 3 the
in-call rfc2833 dtmf is not reliable. This occured for me on asterisk
version 1.6.1.18, 1.6.1.20 it appears to have been fixed when I went to
1.6.2.11 but broken again in 1.6.2.12-rc1.
I have tested with Grandstream and SNOM phones and both fail 90% of the
time Unidata phones fail 10% of the time Audiocodes and Grandstream ATA’s
appear to not suffer from the issue on any version of asterisk.
What happens is when a caller trys to enter DTMF keys durring a call the
far end routed through these carriers do not detect all of the digits. We
did captures with broadvox and here is what they have said.
Hello,
Per our phone conversation I have attached our signaling capture. The issue
is that after we receive a RTP packet, the RTP event that follows needs to
be sent within 100 ms. Anything greater than 100 ms will not be received.
Thank you,
Broadvox
Network Operations Center
Any one else seen this? Any ideas?
Please note you must be being proxied directly to the carrier so your RTP
flows direct other wise you will not see the issue.
Thanks
Bryant
asterisk-users Digest, Vol 73, Issue 58
WARNING: this is an automatic post retrieved from the Asterisk-Users Mailing List, not an authored post
Mailing-list Collector
August 27, 2010
Asterisk Users
Comments Off
Tags: asterisk, carlos chavez, dtmf digits, lee archer, linux, steve edwards
On 8/27/10,
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body ‘help’ to
> asterisk-users-request@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-owner@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than “Re: Contents of asterisk-users digest…”
>
>
> 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
> Subject: [asterisk-users] CDR on Transfer…
> To: Asterisk
> Message-ID: <1282843507.2830.13.camel@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
> Subject: Re: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID: <65F20266-3E46-4DCC-A17D-D181F8E4A6AE@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:
>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steven C.
>> Blair
>> Sent: Wednesday, August 25, 2010 2:08 PM
>> To:
>> 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”
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: “‘Asterisk Users Mailing List – Non-Commercial Discussion’”
>
> Message-ID: <201008261730.o7QHULt4029526@mail.debsinc.com>
> Content-Type: text/plain; charset=”us-ascii”
>
> Can you post the CLI output showing the hangup/script failure?
>
>
>
> _____
>
> From:
> [mailto:asterisk-users-bounces@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
>
> ————– next part ————–
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>
> ——————————
>
> Message: 4
> Date: Thu, 26 Aug 2010 14:55:50 -0400
> From: M S <101mcs@gmail.com>
> Subject: [asterisk-users] double DTMF digits
> To:
> Message-ID:
>
> 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
> Subject: Re: [asterisk-users] double DTMF digits
> To:
> Message-ID: <4C76BF40.20204@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
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID:
>
> 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
> Newline Fax: +1-760-731-3000
>
> ——————————
>
> Message: 7
> Date: Thu, 26 Aug 2010 14:52:53 -0500
> From: “Danny Nicholas”
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: “‘Asterisk Users Mailing List – Non-Commercial Discussion’”
>
> Message-ID: <201008261924.o7QJOREn030652@mail.debsinc.com>
> Content-Type: text/plain; charset=”iso-8859-1″
>
>>From:
> [mailto:asterisk-users-bounces@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
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID:
>
> 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@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
> Newline Fax: +1-760-731-3000
>
>
>
> ——————————
>
> Message: 9
> Date: Thu, 26 Aug 2010 16:51:04 -0400
> From: Matt Desbiens
> Subject: Re: [asterisk-users] double DTMF digits
> To:
> Discussion
> Message-ID:
>
> 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
>
>> 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
>>
> ————– next part ————–
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>
> ——————————
>
> Message: 10
> Date: Thu, 26 Aug 2010 18:58:31 -0700
> From: Joe Wood
> 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
>
> Message-ID:
>
> 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@conference-calls:1]
> Answer(“SIP/2063161626-00000001″, “”) in new stack
> == Using SIP RTP CoS mark 5
> — Executing [s@conference-calls:1]
> Answer(“SIP/2063161626-00000002″, “”) in new stack
> — Executing [s@conference-calls:2]
> BackGround(“SIP/2063161626-00000001″, “welcome”) in new stack
> —
> — Executing [s@conference-calls:2]
> BackGround(“SIP/2063161626-00000002″, “welcome”) in new stack
> —
> — Executing [s@conference-calls:3]
> BackGround(“SIP/2063161626-00000001″, “and”) in new stack
> —
> — Executing [s@conference-calls:3]
> BackGround(“SIP/2063161626-00000002″, “and”) in new stack
> —
> — Executing [s@conference-calls:4]
> BackGround(“SIP/2063161626-00000001″, “thank-you-for-calling”) in new
> stack
> —
> (language ‘en’)
> — Executing [s@conference-calls:4]
> BackGround(“SIP/2063161626-00000002″, “thank-you-for-calling”) in new
> stack
> —
> (language ‘en’)
> — Executing [s@conference-calls:5]
> BackGround(“SIP/2063161626-00000001″, “conference-reservations”) in
> new stack
> —
> ‘conference-reservations.ulaw’ (language ‘en’)
> — Executing [s@conference-calls:5]
> BackGround(“SIP/2063161626-00000002″, “conference-reservations”) in
> new stack
> —
> ‘conference-reservations.ulaw’ (language ‘en’)
> — Executing [s@conference-calls:6]
> Wait(“SIP/2063161626-00000001″, “2″) in new stack
> — Executing [s@conference-calls:6]
> Wait(“SIP/2063161626-00000002″, “2″) in new stack
> — Executing [s@conference-calls:7]
> BackGround(“SIP/2063161626-00000001″, “enter-conf-pin-number”) in new
> stack
> —
> (language ‘en’)
> — Executing [s@conference-calls:7]
> BackGround(“SIP/2063161626-00000002″, “enter-conf-pin-number”) in new
> stack
> —
> (language ‘en’)
> — Executing [s@conference-calls:8]
> WaitExten(“SIP/2063161626-00000001″, “10″) in new stack
> — Executing [s@conference-calls:8]
> WaitExten(“SIP/2063161626-00000002″, “10″) in new stack
> — Timeout on SIP/2063161626-00000001, going to ‘t’
> — Executing [t@conference-calls:1]
> Playback(“SIP/2063161626-00000001″, “vm-goodbye”) in new stack
> —
> — Timeout on SIP/2063161626-00000002, going to ‘t’
> — Executing [t@conference-calls:1]
> Playback(“SIP/2063161626-00000002″, “vm-goodbye”) in new stack
> —
> — Executing [t@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@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@gmail.com>
> Subject: Re: [asterisk-users] double DTMF digits
> To:
> Message-ID:
>
> 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
>
>>
>> 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”
> Subject: Re: [asterisk-users] Use of AGISIGHUP
> To: “Asterisk Users Mailing List – Non-Commercial Discussion”
>
> Message-ID:
>
> 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:
> [mailto:asterisk-users-bounces@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@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
> 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
> Subject: [asterisk-users] dynamic MeetMe, min. digits
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID: <4C77851D.4090802@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 !
> ————– next part ————–
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>
> ——————————
>
> Message: 14
> Date: Fri, 27 Aug 2010 05:58:57 -0400
> From: Doug Lytle
> Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID: <4C778C61.4080104@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.”
> Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID: <4C779356.1070405@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
>>
> ————– next part ————–
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>
> ——————————
>
> Message: 16
> Date: Fri, 27 Aug 2010 17:09:33 +0530
> From: Tino
> Subject: [asterisk-users] music on hold in blind transfer
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID:
>
> Content-Type: text/plain; charset=”iso-8859-1″
>
> Hello,
>
> Is it possible to avoid playing music on hold during a blind transfer ?
>
> Thanks
> ————– next part ————–
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>
> ——————————
>
> Message: 17
> Date: Fri, 27 Aug 2010 17:35:26 +0530
> From: Tino
> Subject: [asterisk-users] queue agent and blind transfer
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID:
>
> 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 ?
> ————– next part ————–
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>
> ——————————
>
> Message: 18
> Date: Fri, 27 Aug 2010 08:51:04 -0400
> From: Dan Journo
> Subject: [asterisk-users] Call Forwarding
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID:
> <31C6BA8C3525D840B022617ACBB7BC036FE20831FF@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
>
> ————– next part ————–
> An HTML attachment was scrubbed…
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>
> ——————————
>
> Message: 19
> Date: Fri, 27 Aug 2010 08:52:58 -0400
> From: Paul Belanger
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID:
>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On Fri, Aug 27, 2010 at 7:39 AM, Tino
>> 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:
> blog.polybeacon.com
>
>
>
> ——————————
>
> Message: 20
> Date: Fri, 27 Aug 2010 15:13:22 +0200
> From: Stefan Schmidt
> Subject: Re: [asterisk-users] Call Forwarding
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID: <4C77B9F2.5030900@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
> 059944 – 2440 zur Verf?gung.
>
> Mit freundlichen Gr?ssen
> –
> Stefan Schmidt
> Sysadmin/VOIP //
> ————————————————-
> 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
> Subject: [asterisk-users] Duplicate channel variables after transfer
> To:
> Message-ID: <201008271513.54789.alex@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
> i
> see that the channel variables of A have been merged into B2
> 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
> ================================================================================
> Info:
> Name= SIP/sipout-00000055
> Type= SIP
> UniqueID= 1282913436.108
> ….
> Variables:
> …
> SIPCALLID=eae94252-ebf238ff@172.28.4.112
> …
> SIPCALLID=lyvkqtybsgrtsnh@172.28.4.113
> …
> ================================================================================
>
>
> I want to get
>
>
> –
> Greetings,
>
> Alex Hermann
>
>
>
>
> ——————————
>
> Message: 22
> Date: Fri, 27 Aug 2010 15:46:44 +0200
> From: Andra?
> Subject: Re: [asterisk-users] CDR on Transfer…
> To: Asterisk Users Mailing List – Non-Commercial Discussion
>
> Message-ID:
>
> Content-Type: text/plain; charset=”utf-8″
>
> Did you find the solution?
>
> On Thu, Aug 26, 2010 at 7:25 PM, Carlos Chavez
>
>
>> 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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>
> ——————————
>
> _______________________________________________
> –Bandwidth and Colocation Provided by http://www.api-digital.com–
>
> AstriCon 2010 – October 26-28 Washington, DC
> Register Now: http://www.astricon.net/
>
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> End of asterisk-users Digest, Vol 73, Issue 58
> **********************************************
>
Duplicate channel variables after transfer
WARNING: this is an automatic post retrieved from the Asterisk-Users Mailing List, not an authored post
Mailing-list Collector
August 27, 2010
Asterisk Users
Comments Off
Tags: asterisk, b1 and b2, Duplicate, extension, linux, lyvkqtybsgrtsnh, sip, transfer, variables
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
see that the channel variables of A have been merged into B2
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
================================================================================
Info:
Name= SIP/sipout-00000055
Type= SIP
UniqueID= 1282913436.108
….
Variables:
…
SIPCALLID=eae94252-ebf238ff@172.28.4.112
…
SIPCALLID=lyvkqtybsgrtsnh@172.28.4.113
…
================================================================================
I want to get
