[asterisk-users] PRI TE110P Configuration (Solved)
Shyju K
shyjuk at gmail.com
Fri Sep 26 09:28:27 CDT 2008
Hi,
The problem solved
After installing new zaptel drivers, we ran the "genzaptel" command to
generate /etc/zaptel.conf file,checked with "zttool" command and the card
status was "Yellow alarm/Blue alarm/Recovering" and the card LED was
blinking red and green.
The problem was with the generated zaptel configuration., but not with
the pin configuration.When we generate zaptel.conf file with genzaptel
command it created one extra "crc4" field in span setting line.We have to
remove this field to make it work.
Generated zaptel.conf file
---------------
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
loadzone=us
defaultzone=us
---------------------
zaptel.conf file After editing
---------------
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
loadzone=in
defaultzone=in
------------------------------
Normal straight through LAN cable worked for this solution.
Note: genzaptel command in new zaptel driver created the problem and note
that the PRI card TE110P is discontinued by Digium.
On 9/26/08, asterisk-users-request at lists.digium.com <
asterisk-users-request at lists.digium.com> wrote:
>
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> Today's Topics:
>
> 1. Re: Astricon people please post the announcement (Tim Panton)
> 2. Re: Astricon people please post the announcement (Tim Panton)
> 3. PRI TE110P Configuration (Shyju K)
> 4. Re: OT: Do You Know What the Problem With CDMA is? (Drew Gibson)
> 5. Re: Astricon people please post the announcement (Ming Yong)
> 6. Re: Astricon people please post the announcement (Tzafrir Cohen)
> 7. Re: Astricon people please post the announcement
> (Brian J. Murrell)
> 8. Re: Asterisk 1.4 is asking me for Mailbox # (Joseph)
> 9. Re: PRI TE110P Configuration (Steven Howes)
> 10. Re: Astricon people please post the announcement (Steve Anness)
> 11. asterisk-app_nv_faxdetect - Gentoo ebuild for *-1.4 was:
> NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6
> (Joseph)
> 12. sip forking needed for ekiga 3.0 (Brian J. Murrell)
> 13. Re: sip forking needed for ekiga 3.0 (SIP)
> 14. Re: sip forking needed for ekiga 3.0 (Brian J. Murrell)
> 15. Server Dimensioning (Jon Weisman)
> 16. Re: sip forking needed for ekiga 3.0 (SIP)
> 17. Re: Server Dimensioning (Sam Tam)
> 18. Re: sip forking needed for ekiga 3.0 (Brian J. Murrell)
> 19. Create virtual extension (Manolet Gmail)
> 20. Re: Server Dimensioning (Philipp Kempgen)
> 21. Re: Server Dimensioning (Philipp Kempgen)
> 22. Skype + Asterisk Interview at Astricon (Ming Yong)
> 23. Re: Create virtual extension (Brent Davidson)
> 24. Mysql Command and number rows returned (David Murphy)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 25 Sep 2008 18:00:00 +0100
> From: Tim Panton <thp at westhawk.co.uk>
> Subject: Re: [asterisk-users] Astricon people please post the
> announcement
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <405E2CDC-2083-458D-9EFC-28A62D8366C9 at westhawk.co.uk>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>
> It's essentially a channel driver.
> Licensed per channel in the same way that the g729 codec is.
>
> Limited private beta opening soon.
>
> Tim.
>
>
> On 25 Sep 2008, at 17:47, Steve Anness wrote:
>
> > So does this mean that my users who currently have skype running on
> > their
> > systems won't have to install anything new once I get things rolling
> > on the
> > Asterisk server?
> >
> > Steve
> >
> >
> > On 9/25/08 11:38 AM, "randulo" <spamsucks2005 at gmail.com> wrote:
> >
> >> So Skype finally will talk to Asterisk!!!! Excellent news!
> >>
> >> On Thu, Sep 25, 2008 at 6:17 PM, randulo <spamsucks2005 at gmail.com>
> >> wrote:
> >>> Digium is making a "big announcement" today at Astricon. So who's
> >>> gonna post this and where? I must know before I go to sleep. It may
> >>> change my life!
> >>>
> >>> r
> >>>
> >>
> >> _______________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> >> Register Now: http://www.astricon.net
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 25 Sep 2008 18:01:30 +0100
> From: Tim Panton <thp at westhawk.co.uk>
> Subject: Re: [asterisk-users] Astricon people please post the
> announcement
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <B71922F3-1F0C-49F8-8F89-57E5BDC76410 at westhawk.co.uk>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>
> They demoed it - everyone seems pretty confident it works
> as advertized.
> No wide-band codec (yet)
>
> Tim.
>
> On 25 Sep 2008, at 17:55, randulo wrote:
>
> > I know a lot of linux and open source people think it's superfluous,
> > but a pseudo chan_skype is huge (assuming it works as advertised). It
> > means anyone with Skype can connect to your server presence. And
> > presumably you can call people via Skype. And use Skype out, etc.
> >
> >
> >
> > On Thu, Sep 25, 2008 at 6:47 PM, Steve Anness
> > <steve.anness at gmail.com> wrote:
> >> So does this mean that my users who currently have skype running on
> >> their
> >> systems won't have to install anything new once I get things
> >> rolling on the
> >> Asterisk server?
> >>
> >> Steve
> >>
> >>
> >> On 9/25/08 11:38 AM, "randulo" <spamsucks2005 at gmail.com> wrote:
> >>
> >>> So Skype finally will talk to Asterisk!!!! Excellent news!
> >>>
> >>> On Thu, Sep 25, 2008 at 6:17 PM, randulo <spamsucks2005 at gmail.com>
> >>> wrote:
> >>>> Digium is making a "big announcement" today at Astricon. So who's
> >>>> gonna post this and where? I must know before I go to sleep. It may
> >>>> change my life!
> >>>>
> >>>> r
> >>>>
> >>>
> >>> _______________________________________________
> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
> >>> --
> >>>
> >>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> >>> Register Now: http://www.astricon.net
> >>>
> >>> asterisk-users mailing list
> >>> To UNSUBSCRIBE or update options visit:
> >>> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >>
> >> _______________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> >> Register Now: http://www.astricon.net
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 25 Sep 2008 23:08:52 +0530
> From: "Shyju K" <shyjuk at gmail.com>
> Subject: [asterisk-users] PRI TE110P Configuration
> To: asterisk-users at lists.digium.com
> Message-ID:
> <d2bccc2d0809251038y15444d30u5f97e93b298aa940 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I was configuring asterisk with TE110P Card.When run zttool
> It is showing "Blue Alarm/Yellow Alarm/Recovering" and the
> card's LED is blinking RED and GREEN.
> I have connected 1&2,4&5 Lines from ISDN modem(RAD ASMi-52)
> to 1&2,4&5 of the PRI card respectively.
>
> I am using Airtel's(India) ISDN connection
>
> Plese Help me to sort it out..........
> --
> Regards,
> Shyju
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>
> ------------------------------
>
> Message: 4
> Date: Thu, 25 Sep 2008 13:39:41 -0400
> From: Drew Gibson <drew at oanda.com>
> Subject: Re: [asterisk-users] OT: Do You Know What the Problem With
> CDMA is?
> To: akohlsmith-lists at mixdown.ca, Asterisk Users Mailing List -
> Non-Commercial Discussion <asterisk-users at lists.digium.com>
> Message-ID: <48DBCCDD.2080407 at oanda.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Andrew Kohlsmith (lists) wrote:
> > On September 25, 2008 10:41:45 am Drew Gibson wrote:
> >
> >> Once CDMA has gone the way of the dodo in North America, I really will
> >> miss one of my favourite scenes:-
> >>
> >> Visiting Brit steps off plane and checks phone for messages...
> >>
> >> Puzzled look appears as they ask "Why doesn't my phone work? It worked
> >> fine in France/Italy/Germany/Timbuktu."
> >>
> >> You start to explain about CDMA and their eyes open wide as they realize
> >> they have just stepped back into the cellular stone age...
> >>
> >
> > You don't have AT&T towers near airports?
> >
> > -A
>
> Nope, no AT&T north of Buffalo.
>
> To be honest, it happened a few years ago (~2002).
>
> We now have Rogers' towers near airports (and 3G iPhones in stores).
>
> Bell Canada and Telus are moving to GSM 3G (side-stepping standard GSM
> so they don't have to admit their mistakes)
>
>
> regards,
>
> Drew
>
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> ------------------------------
>
> Message: 5
> Date: Thu, 25 Sep 2008 11:45:19 -0600
> From: "Ming Yong" <ming at voiceroute.net>
> Subject: Re: [asterisk-users] Astricon people please post the
> announcement
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <4e96fe9a0809251045tffeda03raf4d180194a501ec at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi all,
> Voiceroute is twittering abt it
> http://twitter.com/voiceroute
> Video with mark on announcement will be uploaded in 1 hour.
> http://youtube.com/voiceroute
> Ming
>
>
>
> On 9/25/08, randulo <spamsucks2005 at gmail.com> wrote:
> > Digium is making a "big announcement" today at Astricon. So who's
> > gonna post this and where? I must know before I go to sleep. It may
> > change my life!
> >
> > r
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --
> Sent from Gmail for mobile | mobile.google.com
>
> Ming Yong
> CEO, www.voiceroute.org
> Druid - Open Source Unified Communications
> DID: +1-877-242-3704
> Office: +1-866-915-2407 ext 301
> SIP/email: ming at voiceroute.net
> ------------------------------------------------------------------
> Attend Vikram's talk at ASTRICON 2008, 23-25 Sept 08, Glendale Arizona
> http://www.astricon.net/2008/glendale/web/confTracks.php#t193
>
> Meet us at WEB 2.0 EXPO, 17-18 Sept 08, Booth #17 in Long Tail
> Pavilion, Javits Center, NYC
> http://druidweb20.eventbrite.com
>
> DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform)
> http://www.voiceroute.org/druidcon
>
> VoiceCON 08 San Francisco
> 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco,
> CA
> http://druidvoicecon.eventbrite.com
>
> UC 2.0 Video - Mozilla Ubiquity + Druid
> http://www.youtube.com/watch?v=f-5rDBPuGRc
>
>
>
> ------------------------------
>
> Message: 6
> Date: Thu, 25 Sep 2008 20:49:13 +0300
> From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
> Subject: Re: [asterisk-users] Astricon people please post the
> announcement
> To: asterisk-users at lists.digium.com
> Message-ID: <20080925174913.GH4244 at xorcom.com>
> Content-Type: text/plain; charset=us-ascii
>
> On Thu, Sep 25, 2008 at 06:38:24PM +0200, randulo wrote:
> > So Skype finally will talk to Asterisk!!!! Excellent news!
>
> Great news! You mean that there is finally a free implementation of the
> skype protocol so I can start using it?
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.cohen at xorcom.com<jabber%3Atzafrir.cohen at xorcom.com>
> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>
>
>
> ------------------------------
>
> Message: 7
> Date: Thu, 25 Sep 2008 13:53:43 -0400
> From: "Brian J. Murrell" <brian at interlinx.bc.ca>
> Subject: Re: [asterisk-users] Astricon people please post the
> announcement
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1222365223.25585.222.camel at pc.ilinx>
> Content-Type: text/plain; charset="us-ascii"
>
> On Thu, 2008-09-25 at 20:49 +0300, Tzafrir Cohen wrote:
> >
> > Great news! You mean that there is finally a free implementation of the
> > skype protocol so I can start using it?
>
> Free? AFAICT, not. Neither free as in beer nor speech. Move along,
> nothing to see here.
>
> b.
>
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> ------------------------------
>
> Message: 8
> Date: Thu, 25 Sep 2008 12:00:31 -0600
> From: Joseph <syscon780 at gmail.com>
> Subject: Re: [asterisk-users] Asterisk 1.4 is asking me for Mailbox #
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <20080925180031.GB6327 at syscon2.inet>
> Content-Type: text/plain; charset=iso-8859-1; format=flowed
>
> On 09/25/08 11:34, Mark Michelson wrote:
> >Joseph wrote:
> >> I just installed *-1.4 and when I enter mail extension it keep asking me
> for Mailbox #
> >>
> >> I have in sip.conf under my extension mailbox=11 type=friend
> >> *-1.2 was jumping straight to messages.
> >>
> >> What did change?
> >>
> >
> >When you call VoiceMailMain, you need to provide the mailbox number as an
> >argument to the application if you don't want to be prompted for a mailbox
> number.
> >
> >Mark Michelson
>
> Thanks I got this one:
> in *-1.2 it I had {CALLERIDNUM}
> in *-1.4 it changed to {CALLERID(num)}
>
> --
> #Joseph
>
>
>
> ------------------------------
>
> Message: 9
> Date: Thu, 25 Sep 2008 19:03:28 +0100
> From: Steven Howes <steve at geekinter.net>
> Subject: Re: [asterisk-users] PRI TE110P Configuration
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <3ADAC20F-306A-434F-8C55-9DA0B6BA7276 at geekinter.net>
> Content-Type: text/plain; charset=US-ASCII; format=flowed
>
>
> On 25 Sep 2008, at 18:38, Shyju K wrote:
>
> > I was configuring asterisk with TE110P Card.When run zttool
> > It is showing "Blue Alarm/Yellow Alarm/Recovering" and the
> > card's LED is blinking RED and GREEN.
> > I have connected 1&2,4&5 Lines from ISDN modem(RAD ASMi-52)
> > to 1&2,4&5 of the PRI card respectively.
> >
> > I am using Airtel's(India) ISDN connection
> >
> > Plese Help me to sort it out..........
>
> Config?
>
>
>
> ------------------------------
>
> Message: 10
> Date: Thu, 25 Sep 2008 13:06:33 -0500
> From: Steve Anness <steve.anness at gmail.com>
> Subject: Re: [asterisk-users] Astricon people please post the
> announcement
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <C5013D59.6C1B%steve.anness at gmail.com<C5013D59.6C1B%25steve.anness at gmail.com>
> >
> Content-Type: text/plain; charset="US-ASCII"
>
> So what a minute. They will charge us to use Skype with our Asterisk
> servers? Yes, I think I shall move along.
>
> Steve
>
>
> On 9/25/08 12:53 PM, "Brian J. Murrell" <brian at interlinx.bc.ca> wrote:
>
> > On Thu, 2008-09-25 at 20:49 +0300, Tzafrir Cohen wrote:
> >>
> >> Great news! You mean that there is finally a free implementation of the
> >> skype protocol so I can start using it?
> >
> > Free? AFAICT, not. Neither free as in beer nor speech. Move along,
> > nothing to see here.
> >
> > b.
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>
> ------------------------------
>
> Message: 11
> Date: Thu, 25 Sep 2008 12:29:50 -0600
> From: Joseph <syscon780 at gmail.com>
> Subject: [asterisk-users] asterisk-app_nv_faxdetect - Gentoo ebuild
> for *-1.4 was: NVFaxDetect (1.0.6), NVBackgroundDetect
> was: Asterisk
> 1.4 or 1.6
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <20080925182950.GC6327 at syscon2.inet>
> Content-Type: text/plain; charset=iso-8859-1; format=flowed
>
> On 09/25/08 08:11, Tzafrir Cohen wrote:
> >On Wed, Sep 24, 2008 at 10:25:45PM -0600, Joseph wrote:
> >
> >> My problme is that few lines in a source code needs to be modified
> >> before compiling it. Changing the source code is a simple thing but
> >> now the ebuild needs to be modified as well to point to the source code;
> >> too many problems.
> >
> >Asterisk 1.2 -> 1.4 is a change in the build system. Most of it (except
> >menuselect) is for the better). Adjusting your build scripts for that
> >(and a packaging system is essentially a glorified build script) only
> >takes some work.
> >
> >I would appreciate it if you hadn't kept your patches for yourselves.
> >This would have also saved you some time on the next release (there are
> >already RCs of 1.6.0 for yor test-building pleassure).
> >
> >BTW: maybe you need a newer version of nvfaxdetect? There has been one
> >released, IIRC. If not, there should be such a version on agx's modules
> >addons collection. Again, keeping your changes to yourself is bad.
> >
> >Also recall that for 1.4 and above you must define AST_MODULE. If you
> >don't do so, you get very strange errors.
>
> I didn't intent to keep it for myself. I would be willing to work on it
> but I might needs some help as I'm not a pro programmer.
> If anybody is willing to help write and ebuild for
> asterisk-app_nv_faxdetect.
>
> We can post it on portage/overlay.
>
> Please drop me private email.
>
> --
> #Joseph
>
>
>
> ------------------------------
>
> Message: 12
> Date: Thu, 25 Sep 2008 14:29:26 -0400
> From: "Brian J. Murrell" <brian at interlinx.bc.ca>
> Subject: [asterisk-users] sip forking needed for ekiga 3.0
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1222367366.25585.230.camel at pc.ilinx>
> Content-Type: text/plain; charset="us-ascii"
>
> So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't
> work. I am told by the ekiga devs in
> http://bugzilla.gnome.org/show_bug.cgi?id=553595 and
> http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is
> that Asterisk does not support SIP forking.
>
> The issue is that I have multiple addresses on my workstation:
>
> 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast qlen
> 1000
> link/ether xx:xx:xx:xx:xx:xx brd ff:ff:ff:ff:ff:ff
> inet 10.75.22.1/24 brd 10.75.22.255 scope global eth0
> inet 10.75.22.101/24 brd 10.75.22.255 scope global secondary eth0:1
>
> So when ekiga (3.0) tries to place a call through Asterisk it in fact
> does parallel requests from all addresses. This is what appears to
> confuse Asterisk. Please see the above tickets for more details.
>
> Thots?
>
> b.
>
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> ------------------------------
>
> Message: 13
> Date: Thu, 25 Sep 2008 14:56:04 -0400
> From: SIP <sip at arcdiv.com>
> Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <48DBDEC4.1000203 at arcdiv.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> My thoughts are that to do parallel requests from every IP address on
> the machine is extremely weird behaviour.
>
> How would any server know which to respond to?
>
> SIP forking is supposed to send requests to multiple different
> destinations (or fork mid-stream to send to different destinations).
> Sending from multiple different points of origin doesn't make any sense
> at all in either a logical or rational fashion. What's it supposed to
> accomplish?
>
> N.
>
> Brian J. Murrell wrote:
> > So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't
> > work. I am told by the ekiga devs in
> > http://bugzilla.gnome.org/show_bug.cgi?id=553595 and
> > http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is
> > that Asterisk does not support SIP forking.
> >
> > The issue is that I have multiple addresses on my workstation:
> >
> > 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast qlen
> 1000
> > link/ether xx:xx:xx:xx:xx:xx brd ff:ff:ff:ff:ff:ff
> > inet 10.75.22.1/24 brd 10.75.22.255 scope global eth0
> > inet 10.75.22.101/24 brd 10.75.22.255 scope global secondary eth0:1
> >
> > So when ekiga (3.0) tries to place a call through Asterisk it in fact
> > does parallel requests from all addresses. This is what appears to
> > confuse Asterisk. Please see the above tickets for more details.
> >
> > Thots?
> >
> > b.
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
>
> Message: 14
> Date: Thu, 25 Sep 2008 15:03:59 -0400
> From: "Brian J. Murrell" <brian at interlinx.bc.ca>
> Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1222369439.25585.238.camel at pc.ilinx>
> Content-Type: text/plain; charset="us-ascii"
>
> On Thu, 2008-09-25 at 14:56 -0400, SIP wrote:
> > Sending from multiple different points of origin doesn't make any sense
> > at all in either a logical or rational fashion. What's it supposed to
> > accomplish?
>
> It seems to be a "shot-gun" approach to making a SIP connection. The
> assumption being I suppose that one or more of the IP aliases will fail
> for whatever reason (policy routing, filtering, etc.), so just try them
> all, and use the first one to make a completion and drop the others.
>
> b.
>
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>
> ------------------------------
>
> Message: 15
> Date: Thu, 25 Sep 2008 15:08:19 -0400
> From: "Jon Weisman" <jweisman at ibell.net>
> Subject: [asterisk-users] Server Dimensioning
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <6C86A52999604699998A419BD4F8289C at VaioMatic>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
> reply-type=original
>
> All,
>
> I'm planning on getting a Dell PowerEdge 1950. We want to use our Digium
> TE410P card, calls will come in TDM and go out VoIP, we will require to
> compress them using G729. What specs do I need to support for 4 E-1's with
> cdr logging to mysql? We're thinking about getting two servers 4 E-1's
> each,
> is it possible to fit both cards in one machine?
>
> Thanks,
> Jon
>
>
>
>
>
> ------------------------------
>
> Message: 16
> Date: Thu, 25 Sep 2008 15:31:49 -0400
> From: SIP <sip at arcdiv.com>
> Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <48DBE725.1070800 at arcdiv.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> That strikes me as being careless and unreliable. Call me a purist, but
> I'm of the opinion that you should KNOW which interface to use based on
> which interface is registered and choose ONE interface based on the
> rules you've established during registration. What happens if you want
> to ensure that data goes across a VPN (in order to encrypt your VoIP
> communications) instead of the public internet? Or if you want to ensure
> a particular route based on why you created your multiple interfaces in
> the first place?
>
> That takes all the logic out of the equation and just says, "Here's a
> bunch of packets. Figure out what to do with them. I'll be waiting for
> your response."
>
> There's a reason routing rules exist and mature services allow you to
> control the interface from which it originates.
>
> N.
>
>
> Brian J. Murrell wrote:
> > On Thu, 2008-09-25 at 14:56 -0400, SIP wrote:
> >
> >> Sending from multiple different points of origin doesn't make any sense
> >> at all in either a logical or rational fashion. What's it supposed to
> >> accomplish?
> >>
> >
> > It seems to be a "shot-gun" approach to making a SIP connection. The
> > assumption being I suppose that one or more of the IP aliases will fail
> > for whatever reason (policy routing, filtering, etc.), so just try them
> > all, and use the first one to make a completion and drop the others.
> >
> > b.
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
>
> Message: 17
> Date: Fri, 26 Sep 2008 03:33:23 +0800
> From: "Sam Tam" <samtam888 at gmail.com>
> Subject: Re: [asterisk-users] Server Dimensioning
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <106d01c91f45$93689940$6601a8c0 at samavww62a3fb7>
> Content-Type: text/plain; charset="us-ascii"
>
> If I am right I think you will find that you will not have enough power to
> run 4e1 with g729 codec on little 1950..
> Sam
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jon Weisman
> Sent: Friday, September 26, 2008 3:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Server Dimensioning
>
> All,
>
> I'm planning on getting a Dell PowerEdge 1950. We want to use our Digium
> TE410P card, calls will come in TDM and go out VoIP, we will require to
> compress them using G729. What specs do I need to support for 4 E-1's with
> cdr logging to mysql? We're thinking about getting two servers 4 E-1's
> each,
>
> is it possible to fit both cards in one machine?
>
> Thanks,
> Jon
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
>
> Message: 18
> Date: Thu, 25 Sep 2008 15:56:30 -0400
> From: "Brian J. Murrell" <brian at interlinx.bc.ca>
> Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1222372590.25585.251.camel at pc.ilinx>
> Content-Type: text/plain; charset="us-ascii"
>
> On Thu, 2008-09-25 at 15:31 -0400, SIP wrote:
> > That strikes me as being careless and unreliable.
>
> That's one argument. I can also see the ekiga developers' argument
> though and that's to strive for the most automatic functionality
> possible. The less things you have to ask users, the more likely you
> are to "just work".
>
> > Call me a purist, but
> > I'm of the opinion that you should KNOW which interface to use based on
> > which interface is registered
>
> We are talking about IP aliases here, not real interfaces.
>
> > and choose ONE interface based on the
> > rules you've established during registration.
>
> What rules would you establish during registration?
>
> > What happens if you want
> > to ensure that data goes across a VPN (in order to encrypt your VoIP
> > communications) instead of the public internet?
>
> Presumably you have some [policy] routing that ensures that. But on the
> other hand, if you did have two addresses on an interface, one for VPN
> and one for "everything else", unless you "shotgun" out you need to
> either know which address to use or ask the user. Either case may fail.
>
> > That takes all the logic out of the equation and just says, "Here's a
> > bunch of packets. Figure out what to do with them. I'll be waiting for
> > your response."
>
> I don't think it's quite that bad. It's more like here's a bunch of
> session requests, please complete them [you don't know it yet, but I'm
> going to tear down all but the first one you complete]. But the glitch
> is that even though I send you 3 of them, due to [policy] routing and
> firewalling, you might only get one.
>
> > There's a reason routing rules exist and mature services allow you to
> > control the interface from which it originates.
>
> Really, I'm just the messenger here. I doubt the ekiga team and the
> asterisk team would be willing to sit down and discuss who is right
> here, so I'm trying to be the conduit.
>
> b.
>
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>
> ------------------------------
>
> Message: 19
> Date: Thu, 25 Sep 2008 15:29:00 -0500
> From: "Manolet Gmail" <manolet at gmail.com>
> Subject: [asterisk-users] Create virtual extension
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <353f56170809251329s2282a8e7v38bfd5dfda28570b at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Have, i want to create a sip extension to a context in my dialplan.
> how i can do that?
>
>
>
> ------------------------------
>
> Message: 20
> Date: Thu, 25 Sep 2008 23:15:05 +0200
> From: Philipp Kempgen <philipp.kempgen at amooma.de>
> Subject: Re: [asterisk-users] Server Dimensioning
> To: Asterisk Users <asterisk-users at lists.digium.com>
> Message-ID: <48DBFF59.6060903 at amooma.de>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Jon Weisman schrieb:
>
> > I'm planning on getting a Dell PowerEdge 1950.
>
> All I can tell is that I have bad experiences with those Dell
> PowerEdges. A standard Debian Etch install (2.6.18 kernel I think)
> didn't even have the driver to run the network interface.
> At least Dell doesn't seem to play nice with Debian.
>
>
> Philipp Kempgen
>
> --
> http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
> Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
>
>
> ------------------------------
>
> Message: 21
> Date: Thu, 25 Sep 2008 23:19:41 +0200
> From: Philipp Kempgen <philipp.kempgen at amooma.de>
> Subject: Re: [asterisk-users] Server Dimensioning
> To: Asterisk Users <asterisk-users at lists.digium.com>
> Message-ID: <48DC006D.4040206 at amooma.de>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Philipp Kempgen schrieb:
> > Jon Weisman schrieb:
> >
> >> I'm planning on getting a Dell PowerEdge 1950.
> >
> > All I can tell is that I have bad experiences with those Dell
> > PowerEdges. A standard Debian Etch install (2.6.18 kernel I think)
> > didn't even have the driver to run the network interface.
>
> But afaicr that was a PowerEdge 2950 or something.
>
> Philipp Kempgen
>
> --
> http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
> Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
>
>
> ------------------------------
>
> Message: 22
> Date: Thu, 25 Sep 2008 16:08:30 -0600
> From: "Ming Yong" <ming at voiceroute.net>
> Subject: [asterisk-users] Skype + Asterisk Interview at Astricon
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>, "Commercial and
> Business-Oriented
> Asterisk Discussion" <asterisk-biz at lists.digium.com>
> Message-ID:
> <4e96fe9a0809251508q5105ed1eie851e3ce281d2a56 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi all
>
> Interview with Mark Spencer & Wilhelm Lundborg (Manager, Skype for
> Business)
> on Skype + Asterisk announcement at Astricon 2008
> See video here
> http://www.youtube.com/watch?v=ABYkNUuShpY
> Follow up on coverage of Astricon & Druid
> http://twitter.com/voiceroute
>
> Ming
>
> --
> Ming Yong
> CEO, www.voiceroute.org
> Druid - Open Source Unified Communications
> DID: +1-877-242-3704
> Office: +1-866-915-2407 ext 301
> SIP/email: ming at voiceroute.net
> ------------------------------------------------------------------
> DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform)
> http://www.voiceroute.org/druidcon
>
> VoiceCON 08 San Francisco
> 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco,
> CA
> http://druidvoicecon.eventbrite.com
>
> Voiceroute videos on Druid, Open Source Unified Communications & Asterisk
> http://youtube.com/voiceroute
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>
> ------------------------------
>
> Message: 23
> Date: Thu, 25 Sep 2008 17:16:35 -0500
> From: Brent Davidson <brent at texascountrytitle.com>
> Subject: Re: [asterisk-users] Create virtual extension
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <48DC0DC3.5050708 at texascountrytitle.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Manolet Gmail wrote:
> > Have, i want to create a sip extension to a context in my dialplan.
> > how i can do that?
> >
> > _______________________________________________
> >
> Simple. Use a Goto:
>
> [context1]
> exten => 123,1,Goto (context2,456,1)
>
> [context2]
> exten => 456,1,Background(tt-monkeys)
>
>
>
>
>
> ------------------------------
>
> Message: 24
> Date: Thu, 25 Sep 2008 17:17:03 -0500
> From: "David Murphy" <dmurphy at leadgeniuses.com>
> Subject: [asterisk-users] Mysql Command and number rows returned
> To: <asterisk-users at lists.digium.com>
> Message-ID: <035701c91f5c$70454550$50cfcff0$@com>
> Content-Type: text/plain; charset="us-ascii"
>
> Without issuing a separate loop thru a result set. Is there any way
> anyone knows of to output the number of rows a mysql query returned.
>
>
>
> Aka
>
> ..
>
> exten => 1,n,MYSQL(Query resultid ${connid} SELECT\ `State`\ FROM\
> `AreaCodes`\ WHERE\ `AreaCode`=\'${CIDArea}\')
>
> exten => 1,n(fetchrow),MYSQL(Fetch foundRow ${resultid} number) ; fetch row
>
> exten => 1,n,GotoIf($["${foundRow}" = "1"]?done) ; leave loop if no row
> found
>
> exten => 1,n,Set(State=${State})
>
> exten => 1,n,Goto(fetchrow) ; continue loop if row found
>
> exten => 1,n,Set(RowsReturned=MYSQL(Fetch rowcount ${resultid})
>
> exten => 1,n(done),MYSQL(Clear ${resultid})
>
> ..
>
>
>
>
>
>
>
>
>
> David Murphy Systems Adminsitrator
>
> myLogo
>
> Email: AIM:
>
> <mailto:omar at icewatermedia.com> david at icewatermedia.com
> lgdavidmurphy
>
>
>
>
>
>
>
>
>
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> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> End of asterisk-users Digest, Vol 50, Issue 74
> **********************************************
>
--
Regards,
Shyju
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