*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, <[email protected]>wrote: > Send asterisk-users mailing list submissions to > [email protected] > > 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 > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > 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" <[email protected]> > Subject: Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not > correct > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > 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@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 <[email protected]> > Subject: [asterisk-users] Gateway? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: > <CAMoLvkyLF_U5N_8aAOvz40JqQuFOpU= > [email protected]> > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20130429/5eba090f/attachment-0001.htm > > > > ------------------------------ > > Message: 3 > Date: Mon, 29 Apr 2013 22:35:08 +0200 > From: jg <[email protected]> > Subject: Re: [asterisk-users] Gateway? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > 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? <[email protected]> > Subject: Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not > correct > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > Content-Type: text/plain; charset=utf-8 > > ----- Mail original ----- > De: "David M. Lee" <[email protected]> > ?: "Asterisk Users Mailing List - Non-Commercial Discussion" < > [email protected]> > 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@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 <[email protected]> > Subject: Re: [asterisk-users] Gateway? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Message-ID: <[email protected]> > 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 <[email protected]> > Subject: Re: [asterisk-users] Can't register to Asterisk 1.6 with old > Aastra phones > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: > <CAPd1dq_ktetBpLHhXDm=0wr4QTRj2MHmp4-9ufCjG= > [email protected]> > 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 > [email protected] > > Integrated IT services from > Plot 57B Luthuli Avenue Bugolobi, Kampala > > > > > > > On Sun, Apr 28, 2013 at 11:56 PM, Carlos Alvarez <[email protected] > >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@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 > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/f7f1bc6b/attachment-0001.htm > > > > ------------------------------ > > Message: 7 > Date: Tue, 30 Apr 2013 09:11:48 -0400 > From: Eric Wieling <[email protected]> > Subject: Re: [asterisk-users] Gateway? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: > > <616B4ECE1290D441AD56124FEBB03D081713F8F0C2@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 <[email protected]> > Subject: [asterisk-users] hello! > To: hr ccsgroups <[email protected]>, coolguyrocks > <[email protected]>, simbus hr <[email protected]>, > gowdanar > <[email protected]>, asterisk users > <[email protected]>, hr <[email protected]>, > rcnoida > <[email protected]> > Message-ID: > <[email protected]> > Content-Type: text/plain; charset="utf-8" > > > http://seed4life.org/wp-content/themes/twentytwelve/basesball.php?hfazq792vlxjd > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ________________ > You're going to find that many of the truths we cling to depend entirely > upon one's point of view. -- Obi-Wan Kenobi > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/fbb4af3e/attachment-0001.htm > > > > ------------------------------ > > Message: 9 > Date: Tue, 30 Apr 2013 18:30:27 +0200 > From: Danilo Dionisi <[email protected]> > Subject: [asterisk-users] Asterisk QSIG doesnt send the calling name > to Nortel CS1000 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: > < > caabtbuys04zw8u5jan+zcjybofxrvks_wigy5c60ytjrz-j...@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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/b1c54de0/attachment-0001.htm > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2010 - October 26-28 Washington, DC > Register Now: http://www.astricon.net/ > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 105, Issue 39 > *********************************************** > -- BIPIN RAGHUVANSHI OPERATION HEAD ASTERISK (DEVELOPMENT AND RESEARCH) WWW.EHORIZONS.IN [email protected] [email protected]
-- _____________________________________________________________________ -- 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
