asterisk users - May 2013 - asterisk-users Digest, Vol 105, Issue 39

*I'm trying to build an application that provides statistics of
calls*>* and call recording. Someone told me this could be done out of
band*>* with a SPAN (?) port that would replicate SIP and media
packets to a*>* separate NIC without having to actually pass the
real-calls thru*>* asterisk. It was explained that this SPAN port
would in the SBC*>* would replicate data received.*>* *>* *>* If
this
is done, is there a way I can utilize asterisk to interpret*>* these
packets without actually having any control of the calls? If so*>*
how? Sorry, I'm new to telco, so hopefully this post makes sense to*>*
someone.*



On Tue, Apr 30, 2013 at 10:30 PM,
<asterisk-users-request at lists.digium.com>wrote:
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> Today's Topics:
>
>    1. Re: Asterisk 11.3.0 - Mask for new file not correct (David M. Lee)
>    2. Gateway? (James Wystead)
>    3. Re: Gateway? (jg)
>    4. Re: Asterisk 11.3.0 - Mask for new file not correct (Ludovic Bou?)
>    5. Re: Gateway? (A J Stiles)
>    6. Re: Can't register to Asterisk 1.6 with old Aastra        phones
>       (Bob Kyeyune)
>    7. Re: Gateway? (Eric Wieling)
>    8. hello! (Rahul Pachauri)
>    9. Asterisk QSIG doesnt send the calling name to     Nortel CS1000
>       (Danilo Dionisi)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 29 Apr 2013 12:51:42 -0500
> From: "David M. Lee" <dlee at digium.com>
> Subject: Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not
>         correct
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Message-ID: <38652031-864B-45A9-A779-B90862F1AE97 at digium.com>
> Content-Type: text/plain; charset=iso-8859-1
>
>
> On Apr 29, 2013, at 10:51 AM, Ludovic Bou? wrote:
>
> > The fact is we want to use the RECORDED_FILE function from
> Application_Record module and create a file with 666 permissions. But when
> I check the created file, rights are not what I expected.
> >
> > [root at STD1-SRVASTSVI-03 pseudos]$ ll
> > -rw-r--r-- 1 asterisk asterisk 51244 mars 29 16:04 Pseudo_2_1111.wav
> >
> > I checked the doc on
> https://wiki.asterisk.org/wiki/display/AST/Application_Record but I
> didn't find anything about umask permissions. I checked Doxygen, I can
see
> file creation permissions is set to 666
> > #define AST_FILE_MODE 0666
> >
>
http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42
> >
> > What can I do to fix that or debug?
>
> The AST_FILE_MODE works by the same rules as mode parameter in open(2):
> "The effective permissions are modified by the process's umask in
the usual
> way: The permissions of the created file are (mode & ~umask)."[1]
>
> My guess is that the umask of your asterisk process is 022, which is very
> typical. You'll have to play around with your umask settings and file
> permissions to get things the way you want them.
>
>  [1]: http://linux.die.net/man/2/open
>
> > Ludovic BOU?
>
> --
> David M. Lee
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 29 Apr 2013 15:56:24 -0400
> From: James Wystead <szilverthorne at gmail.com>
> Subject: [asterisk-users] Gateway?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Message-ID:
>         <CAMoLvkyLF_U5N_8aAOvz40JqQuFOpU> A+gZaW0+ziTGaRZXE2g at
mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> This is going to sound like a dumb-ass question:
>
> The device that allows you to bridge Asterisk (or any other PBX) into the
> pstn.. What is that called? So, I guess, not a SIP trunk, but the device
> that actually IS the SIP trunk.
>
> Am I making sense?
>
> Thanks
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> ------------------------------
>
> Message: 3
> Date: Mon, 29 Apr 2013 22:35:08 +0200
> From: jg <webaccounts at jgoettgens.de>
> Subject: Re: [asterisk-users] Gateway?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Message-ID: <517ED97C.8090606 at jgoettgens.de>
> Content-Type: text/plain; charset=UTF-8; format=flowed
>
> Here are your answers:
>
> 1st question: Anything that makes sense.
> 2nd question: Maybe
>
> Please, explain your setup.
>
> jg
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 30 Apr 2013 10:35:58 +0200 (CEST)
> From: Ludovic Bou? <lboue at afone.com>
> Subject: Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not
>         correct
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Message-ID: <1379623362.9592526.1367310958910.JavaMail.root at
afone.com>
> Content-Type: text/plain; charset=utf-8
>
> ----- Mail original -----
> De: "David M. Lee" <dlee at digium.com>
> ?: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users at lists.digium.com>
> Envoy?: Lundi 29 Avril 2013 19:51:42
> Objet: Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not correct
>
>
> On Apr 29, 2013, at 10:51 AM, Ludovic Bou? wrote:
>
> > The fact is we want to use the RECORDED_FILE function from
> Application_Record module and create a file with 666 permissions. But when
> I check the created file, rights are not what I expected.
> >
> > [root at STD1-SRVASTSVI-03 pseudos]$ ll
> > -rw-r--r-- 1 asterisk asterisk 51244 mars 29 16:04 Pseudo_2_1111.wav
> >
> > I checked the doc on
> https://wiki.asterisk.org/wiki/display/AST/Application_Record but I
> didn't find anything about umask permissions. I checked Doxygen, I can
see
> file creation permissions is set to 666
> > #define AST_FILE_MODE 0666
> >
>
http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42
> >
> > What can I do to fix that or debug?
>
> The AST_FILE_MODE works by the same rules as mode parameter in open(2):
> "The effective permissions are modified by the process's umask in
the usual
> way: The permissions of the created file are (mode & ~umask)."[1]
>
> My guess is that the umask of your asterisk process is 022, which is very
> typical. You'll have to play around with your umask settings and file
> permissions to get things the way you want them.
>
>  [1]: http://linux.die.net/man/2/open
>
>
> You were right, it was necessary to change asterisk process umask. I put
> the following in /etc/init.d/asterisk init script and it works:
>         # umask 002 to create files with 0664 and folders with 0775
>         umask 002
>
> Thanks a lot
>
>
>
> ------------------------------
>
> Message: 5
> Date: Tue, 30 Apr 2013 10:57:32 +0100
> From: A J Stiles <asterisk_list at earthshod.co.uk>
> Subject: Re: [asterisk-users] Gateway?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>         <asterisk-users at lists.digium.com>
> Message-ID: <201304301057.32260.asterisk_list at earthshod.co.uk>
> Content-Type: Text/Plain;  charset="iso-8859-6"
>
> On Monday 29 April 2013, James Wystead wrote:
> > This is going to sound like a dumb-ass question:
> >
> > The device that allows you to bridge Asterisk (or any other PBX) into
the
> > pstn.. What is that called?
>
> Usually it is an expansion card that plugs into a PCI or PCI express slot
> on
> the motherboard; so most people would just call it an analogue telephony
> card
> (such as a TDM410P, for instance)  or an ISDN card  (such as a TE410P).
>  One
> that connects to the mobile networks would be called a GSM card.
>
> Analogue telephony cards are further subdivided into two flavours; FXO
>  (which
> connects to an exchange line)  and FXS  (which connects to a telephone, and
> provides the necessary line bias and ringing voltages).  Usually a single
> card
> will provide for multiple lines, by fitting either FXO or FXS modules as
> required.
>
> --
> AJS
>
> Answers come *after* questions.
>
>
>
> ------------------------------
>
> Message: 6
> Date: Tue, 30 Apr 2013 14:21:49 +0300
> From: Bob Kyeyune <bkyeyune at gmail.com>
> Subject: Re: [asterisk-users] Can't register to Asterisk 1.6 with old
>         Aastra  phones
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Message-ID:
>         <CAPd1dq_ktetBpLHhXDm=0wr4QTRj2MHmp4-9ufCjG> QmvzWuug at
mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> am also stuck with Alcatel lucent IP Touch 4018
> any one connected them to Asterisk
>
> thanks
>
> Regards.
> Kyeyune Bob
> Network & IT Engineer
> +256 774 702 258
> bob.kyeyune at onesolutions.ug
>
> Integrated IT services from
>  Plot 57B Luthuli Avenue Bugolobi, Kampala
>
>
>
>
>
>
> On Sun, Apr 28, 2013 at 11:56 PM, Carlos Alvarez <carlos at
televolve.com
> >wrote:
>
> > We have a new customer with a lot of old phones like the 9133i.  They
> > won't register, and we see some very strange behavior with them. 
If
> > the SIP peer exists, they simply fail silently, with no error in the
> > CLI or the messages log.  Nothing works, but no errors.
> >
> > If the peer does not exist, it's clear that it's registering
improperly:
> >
> > [2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: Registration from
> > 'abc123 <sip:abc123@>' failed for '68.2.x.x' -
No matching peer found
> >
> > Typically of course we'd expect to see:  <sip:abc123 at
server>
> >
> > We're running the latest available firmware, but it's from
2009.  Any
> > ideas on this before we just trash all the older phones?
> >
> > --
> > Carlos Alvarez
> > TelEvolve
> > 602-889-3003
> >
> > --
> > _____________________________________________________________________
> > -- 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: 7
> Date: Tue, 30 Apr 2013 09:11:48 -0400
> From: Eric Wieling <EWieling at nyigc.com>
> Subject: Re: [asterisk-users] Gateway?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Message-ID:
>
> <616B4ECE1290D441AD56124FEBB03D081713F8F0C2 at
mailserver2007.nyigc.globe>
>
> Content-Type: text/plain; charset="us-ascii"
>
> On Monday 29 April 2013, James Wystead wrote:
> > This is going to sound like a dumb-ass question:
> >
> > The device that allows you to bridge Asterisk (or any other PBX) into
> > the pstn.. What is that called?
>
> For 1 - 2 ports they are usually called an ATA (Analog Terminal Adapter).
>  For more than 2 ports they are usually called Media Gateways.
>
>
>
> ------------------------------
>
> Message: 8
> Date: Tue, 30 Apr 2013 21:42:43 +0800 (SGT)
> From: Rahul Pachauri <rahul.pachauri at ymail.com>
> Subject: [asterisk-users] hello!
> To: hr ccsgroups <hr.ccsgroups at gmail.com>, coolguyrocks
>         <coolguyrocks at in.com>,  simbus hr <simbus.hr at
simbustech.com>,
> gowdanar
>         <gowdanar at ChetanaSforum.com>,   asterisk users
>         <asterisk-users at lists.digium.com>, hr <hr at
metaoption.com>,
>  rcnoida
>         <rcnoida at ignou.ac.in>
> Message-ID:
>         <1367329363.26314.YahooMailNeo at
web194905.mail.sg3.yahoo.com>
> Content-Type: text/plain; charset="utf-8"
>
>
>
http://seed4life.org/wp-content/themes/twentytwelve/basesball.php?hfazq792vlxjd
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> ________________
> You're going to find that many of the truths we cling to depend
entirely
> upon one's point of view. -- Obi-Wan Kenobi
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> ------------------------------
>
> Message: 9
> Date: Tue, 30 Apr 2013 18:30:27 +0200
> From: Danilo Dionisi <dionisi.danilo at gmail.com>
> Subject: [asterisk-users] Asterisk QSIG doesnt send the calling name
>         to      Nortel CS1000
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Message-ID:
>         <
> CAAbtBUys04Zw8u5jan+zcjybOfxrVks_wiGy5C60YTjrZ-JQbg at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello to all,
>
> I have a problem with an asterisk qsig.
>
> I have three machines:
>
> Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP
Trunk--->
> Asterisk
>
> I use Snom phones on Asterisk.
> If I call from Asterisk to Nortel, Nortel reminds me of the name of the
> person
> i'm calling and I visualize on the display of Snom phone, but if I call
> from
> Nortel to Asterisk, the QSIG does not send Nortel on the display of the
> name of the person i'm calling ... why?
>
> example:
> Snom phone = "Danilo <1001>"
> Nortel phone = "Marco <2002>"
>
> If I call from Nortel to Asterisk, I have the display of the Snom
"Marco <
> 2002>" and the display of Nortel "Danilo <1001>";
If I call from Nortel to
> Asterisk, I have the display of the Snom "Marco <2002>" and
the display of
> Nortel "<1001>"
>
> This is my / etc / asterisk / chan_dahdi.conf
>
> [channels]
> cc_offer_timer=20
> ccbs_available_timer=4800
> ccnr_available_timer=7200
> cc_recall_timer=20
> cc_agent_policy=native
> cc_monitor_policy=native
> pridialplan=private
> prilocaldialplan=private
>
> context=default
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> facilityenable=yes
> callerid=asreceived
>
>
>
> ;Sangoma A104 port 1 [slot:4 bus:17 span:1] <wanpipe1>
> switchtype=qsig
> context=from_nortel
> group=0
> echocancel=yes
> faxdetect=incoming
> signalling=pri_cpe
> channel =>1-15,17-31
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> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
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> End of asterisk-users Digest, Vol 105, Issue 39
> ***********************************************
>


-- 
BIPIN RAGHUVANSHI
OPERATION HEAD
ASTERISK (DEVELOPMENT AND RESEARCH)
WWW.EHORIZONS.IN
bipinraghuvanshi at gmail.com
bipin.singh at ehorizons.in
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