[Asterisk-Users] Problems with PRI with T410 messages
CW_ASN
cw_asn at fibertel.com.ar
Thu Jun 17 03:54:00 MST 2004
Send traces.
----- Original Message -----
From: "Aimable" <aimable at terracom.rw>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, June 17, 2004 6:28 AM
Subject: [Asterisk-Users] Problems with PRI with T410 messages
> Hi all,
> I have a box running asterisk with T410 connected to a Nortel DMS 100
switch
> and another box running SER with grandstream phones on it
> So if there is a call from the pstn it goes from the Nortel to the
asterisk
> and then to the SER box and finally to the phones.if the phone is busy or
> the number is invalid the * box will first send an ALERT message to the
> Nortel and say the call is going on and the phone is ringing (which is not
> the case )and after it will send a RELEASE message saying that the line
is
> busy or the # is invalid .is there any way * can send a progress message
> instead of the alerting message until it gets the correct message from
SER?
>
>
> Thanks
> Habiyakare Aimable
> Phone Services
> TERRACOM Broadband
> aimable at terracom.rw
>
>
>
>
> -----Original Message-----
> From: asterisk-users-request at lists.digium.com
> [mailto:asterisk-users-request at lists.digium.com]
> Sent: Thursday, June 17, 2004 10:56 AM
> To: asterisk-users at lists.digium.com
> Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs
>
> Send Asterisk-Users mailing list submissions to
> asterisk-users at 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 at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-admin at 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. RE: Soekris Engineering net4801 (Senad Jordanovic)
> 2. Accepting SIP calls from unregistered gateways (Axel)
> 3. Re: pri with TE410P not working (Austria) (Peter Svensson)
> 4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig)
> 5. Calling the firefly network? (Martijn van Oosterhout)
> 6. RE: IAX2 no compatible codecs (Jason Penton)
> 7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler)
> 8. Re: embedded Asterisk (Klaus-Peter Junghanns)
> 9. Re: pri with TE410P not working (Austria) (Michael Bielicki)
> 10. RE: Cost of IP Phones, or Isn't It Just
> Software? (Andy Powell)
> 11. Re: pri with TE410P not working (Austria) (Peter Svensson)
>
> --__--__--
>
> Message: 1
> From: "Senad Jordanovic" <senad at boltblue.com>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] Soekris Engineering net4801
> Date: Thu, 17 Jun 2004 08:34:01 +0100
> Reply-To: asterisk-users at lists.digium.com
>
> John Bittner wrote:
> > Hi,
> >
> > I have it working great. I have debian running on it with music on
> > hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with
> > calls on all 10 phones at the same time through voicepulse with no
> > issues. I ran top with all the phones running and I was only up to
> > 45% cpu. Seems to run ok but I am still in the testing phase.
>
> Great...
> Have you tried to connect a X100P or TDM400P to it?
>
>
> --__--__--
>
> Message: 2
> From: "Axel" <asterisk at avenue500.com>
> To: <asterisk-users at lists.digium.com>
> Date: Thu, 17 Jun 2004 03:43:12 -0400
> Subject: [Asterisk-Users] Accepting SIP calls from unregistered gateways
> Reply-To: asterisk-users at lists.digium.com
>
> This is a multi-part message in MIME format.
>
> ------=_NextPart_000_0351_01C4541D.36B45830
> Content-Type: text/plain;
> charset="iso-8859-1"
> Content-Transfer-Encoding: quoted-printable
>
> Hi,
> Is there a way to accept SIP calls from unregistered gateways?
> autocreatpeer=3Dyes seems to disable checking credentials but the =
> originating gateway is still required to register itself with a username =
> and password (which can be anything since it won't check it).
> I like to be able to receive the call from any gateway without them =
> having to register even, just like a Cisco gateway that you can =
> terminate a call from clients who are not registered. Is such thing =
> possible with Asterisk?
>
> Best regards,
>
> Axel
>
> ------=_NextPart_000_0351_01C4541D.36B45830
> Content-Type: text/html;
> charset="iso-8859-1"
> Content-Transfer-Encoding: quoted-printable
>
> <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
> <HTML><HEAD>
> <META http-equiv=3DContent-Type content=3D"text/html; =
> charset=3Diso-8859-1">
> <META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR>
> <STYLE></STYLE>
> </HEAD>
> <BODY bgColor=3D#ffffff>
> <DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>Is there a way to accept SIP calls from =
>
> unregistered gateways?</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>autocreatpeer=3Dyes seems to disable =
> checking=20
> credentials but the originating gateway is still required to register =
> itself=20
> with a username and password (which can be anything since it won't check =
>
> it).</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>I like to be able to receive the call =
> from any=20
> gateway without them having to register even, just like a Cisco gateway =
> that you=20
> can terminate a call from clients who are not registered. Is such =
> thing=20
> possible with Asterisk?</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2></FONT> </DIV>
> <DIV><FONT face=3DArial size=3D2>Best regards,</FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial size=3D2>Axel<BR></FONT></DIV></BODY></HTML>
>
> ------=_NextPart_000_0351_01C4541D.36B45830--
>
>
>
> --__--__--
>
> Message: 3
> Date: Thu, 17 Jun 2004 09:43:37 +0200 (CEST)
> From: Peter Svensson <psvasterisk at psv.nu>
> To: Asterisk-Users Mailinglist <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
> Reply-To: asterisk-users at lists.digium.com
>
> On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
>
> > ... on the card i can see the two leds pulsing red (i think thats the
> > yellow alaram - or i am wrong) ?
>
> Are you sure it is not a red alarm? That would indicate a loss of link.
> I think you can check with the command zttool.
>
> Are you sure the cables are correct?
> Have you set the jumpers on the card to E1 and not left them on T1?
>
> I think the leds should turn green when the card senses a correct carrier
> and framing on the lines.
>
> Peter
> --
> Peter Svensson ! Pgp key available by finger, fingerprint:
> <petersv at psv.nu> ! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF
> ------------------------------------------------------------------------
> Remember, Luke, your source will be with you... always...
>
>
>
> --__--__--
>
> Message: 4
> From: Holger Schurig <hs4233 at mail.mn-solutions.de>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] ZAPHFC - only for * 0.7.2?
> Date: Thu, 17 Jun 2004 09:59:33 +0200
> Reply-To: asterisk-users at lists.digium.com
>
> > I've got Zaphfc working running Asterisk v. 0.7.2
> >
> > Then I have tried with Asterisk V. 1.0 and the latest from CVS - with
> > no succes. Has anybody got zaphfc working with newer version than 0.7.2
>
> zaphfc is in bri-stuff from www.junghanns.net --- or in a patched version
> at http://capi4linux.thepenguin.de/download/asterisk/. I downloaded the
> latter and let the ./download.sh and ./compile.sh scripts run normally.
>
> Then I install zaptel.o and zaphfc.o to /lib/modules/<kernelversion>/misc
> and do the usual mambo in /etc/modules to run ztcfg after loading zaphfc
> and to load zaptel before zaphfc:
>
> pre-install zaphfc /sbin/modprobe zaptel
> post-install zaphfc /sbin/ztcfg -v
>
> Now I go to a different directory and do a CVS checkout of Asterisk head.
> Just before compiling, I replace channels/chan_zap.c with
> bri-stuff-0.0.2a-pp/asterisk/channels/chan.zap.c.
>
> I then change the lines of the form
>
> static ast_mutex_t usecnt_lock = AST_MUTEX_INITIALIZER;
>
> into
>
> AST_MUTEX_DEFINE_STATIC(usecnt_lock);
>
> and compile & install. And voila, now I have an Asterisk from (almost) CVS
> HEAD working with zaphfc.
>
>
>
>
> The real solution would have been to apply all the patches from
> bri-stuff*/libpri.patch to libpri in CVS. After looking at how much has
> been changed and considering that I don't have a clue about q.921 and
> q.931 I decided to not doing it that way :-)
>
> Also, I'd thing it would be better if KaPeJot put's his software into some
> CVS so that more than one person can add changes and keep things
> up-to-date.
>
> Greetings, Holger
>
>
> --__--__--
>
> Message: 5
> Date: Thu, 17 Jun 2004 18:12:10 +1000
> From: Martijn van Oosterhout <martijn at ecomtel.com.au>
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Calling the firefly network?
> Reply-To: asterisk-users at lists.digium.com
>
> Is there a way to register with or call the firefly network from an
Asterisk
> server. It would be pretty cool if you could gateway calls onto it.
>
> Have a nice day,
> --
> Martijn van Oosterhout
>
> --__--__--
>
> Message: 6
> From: "Jason Penton" <j.penton at ru.ac.za>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] IAX2 no compatible codecs
> Date: Thu, 17 Jun 2004 10:22:10 +0200
> Reply-To: asterisk-users at lists.digium.com
>
> Hi Adam
>
> Thanks - Here are the two attempts:
>
> This is the first one where * dials firefly via the dialplan (which works
> fine):
>
> Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
> Timestamp: 00001ms SCall: 00004 DCall: 00000 [146.231.125.65:4569]
> VERSION : 2
> CALLED NUMBER : s
> CALLING NUMBER : 7001
> CALLING NAME : Alfredo+Terzoli
> LANGUAGE : en
> FORMAT : 4
> CAPABILITY : 2147483647
> ADSICPE : 2
> DATE TIME : 147935435
>
> Now the following output is when I use the manager ORIGINATE command:
>
> Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
> Timestamp: 00001ms SCall: 00004 DCall: 00000 [146.231.125.65:4569]
> VERSION : 2
> CALLED NUMBER : s
> LANGUAGE : en
> FORMAT : 64
> CAPABILITY : 2147483647
> ADSICPE : 0
> DATE TIME : 147935484
>
>
> Jun 17 10:07:57 WARNING[180236]: chan_iax2.c:4534 socket_read: Call
rejected
> by 146.231.125.65: No compatible Codecs
>
>
> I can see the inconsitency with the FORMAT header of the two setup
messages.
> According to the IAX protocol spec. The FORMAT (0x4) represents G.711
U-LAW,
> which is exactly what the resulting call uses. However, the funny thing is
> that the protocol spec has no entry for FORMAT(0x64) in the second
message -
> an undefined format. The quesiton is how the * manager API causes * to
> inititiate an IAX call with this FORMAT type (0x64)??????? An how we can
fix
> it ???????.
>
> Any ideas, anyone
> Thanks again Adam for the help
> Cheers
> Jason
>
>
>
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Adam Hart
> > Sent: 17 June 2004 09:19 AM
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] IAX2 no compatible codecs
> >
> > iax2 debug is your friend, looks at the capibilities asterisk
> > is sending
> > in it's NEW message
> >
> > Jason Penton wrote:
> >
> > > Hi Adam
> > >
> > > Done all that but still the same problem.
> > >
> > > Do you have any other ideas?
> > >
> > > Cheers
> > > Jason
> > >
> > >
> > >>-----Original Message-----
> > >>From: asterisk-users-admin at lists.digium.com
> > >>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> > Adam Hart
> > >>Sent: 17 June 2004 08:29 AM
> > >>To: asterisk-users at lists.digium.com
> > >>Subject: Re: [Asterisk-Users] IAX2 no compatible codecs
> > >>
> > >>check under your network settings that you have all the
> > >>codecs selected
> > >>and obviously type IAX
> > >>
> > >>Jason Penton wrote:
> > >>
> > >>>Hi All
> > >>>
> > >>>I have a strange problem using IAX2. When placing a call to
> > >>
> > >>my IAX clients
> > >>
> > >>>(firefly) via the Asterisk dialplan all works great.
> > >>
> > >>However trying to
> > >>
> > >>>initiate a call via the manager interface to the IAX client
> > >>
> > >>using the
> > >>
> > >>>following command results in an error:
> > >>>
> > >>>Action: Originate
> > >>>Channel: IAX2/7000
> > >>>Extension: 7000
> > >>>Context: local
> > >>>Priority: 1
> > >>>ActionID: 1
> > >>>
> > >>>The error I get in the CLI is "Jun 17 08:18:36 WARNING[180236]:
> > >>>chan_iax2.c:4534 socket_read: Call rejected by #IP: No
> > >>
> > >>compatible Codecs"
> > >>
> > >>>Does anyone have any ideas.
> > >>>
> > >>>Thanks in advance
> > >>>Jason
> > >>>
> > >>>_______________________________________________
> > >>>Asterisk-Users mailing list
> > >>>Asterisk-Users at lists.digium.com
> > >>>http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>>To UNSUBSCRIBE or update options visit:
> > >>> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>>
> > >>
> > >>_______________________________________________
> > >>Asterisk-Users mailing list
> > >>Asterisk-Users at lists.digium.com
> > >>http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>To UNSUBSCRIBE or update options visit:
> > >> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>
> > >>
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> --__--__--
>
> Message: 7
> Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
> From: Wolfgang Pichler <madmin at dialog-telekom.at>
> To: Asterisk-Users Mailinglist <Asterisk-Users at lists.digium.com>
> Date: Thu, 17 Jun 2004 10:28:09 +0200
> Reply-To: asterisk-users at lists.digium.com
>
> Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43:
> > On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
> >
> > > ... on the card i can see the two leds pulsing red (i think thats the
> > > yellow alaram - or i am wrong) ?
> >
> > Are you sure it is not a red alarm? That would indicate a loss of link.
> > I think you can check with the command zttool.
> you are right - its a red alarm - zttool says "Red Alarm/Not Open"
> >
> > Are you sure the cables are correct?
> > Have you set the jumpers on the card to E1 and not left them on T1?
> The jumpers are on E1 - the cables should be ok (they are working with
> other hardware) - and the card is directly connected to a simens ULAF+
> STU Desktop (can't really find much information about this device on the
> net) - which turns off a red led when i load the driver and do a ztcfg.
> >
> > I think the leds should turn green when the card senses a correct
carrier
> > and framing on the lines.
> green is always a wounderful color ;-)
>
> so, what else could cause this ?
>
> wolfgang
>
>
> --__--__--
>
> Message: 8
> Subject: Re: [Asterisk-Users] embedded Asterisk
> From: Klaus-Peter Junghanns <kpj at junghanns.net>
> To: asterisk-users at lists.digium.com
> Organization: Junghanns.NET GmbH
> Date: Thu, 17 Jun 2004 10:11:11 +0200
> Reply-To: asterisk-users at lists.digium.com
>
> Hi,
>
> > Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at
> > 233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which
> > is a downstripped Debian (< 64 MB) on a readonly ext2 filesystem, you
> > should be grand. Installing asterisk + some extra stuff will probably
> > require, that you have at least a 128MB or 256MB flash or so.
>
> Dont go for "stripped down but complete" distributions which include a
> lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like
> i used the SuSE rescue system (14 mb), then you can add what you need
> (sshd,...) and compile asterisk on another box and then just copy it.
> My compressed ramdisk image is 32 mb, including all voice prompts and
> some mp3s for MOH.
>
> >
> > There are actually quite some board around on that CPU, like Soekris,
> > pcengines and i think also Mikrotik at prices from 120EUR and up.
> >
> I just put together the demo system for Linuxtag:
> - Via EPIA 5000 (C3-533), EUR 80,-
> - Morex case with external power supply, EUR 80,-
> - some old 256 mb SDRAMM
> - 128 MB USB memory stick, EUR 30,-
> - 1 quadBRI (could also easily handle an octoBRI, or a PRI card,
> with the dual riser pci card you can use 2 cards)
>
> The C3-533 is an i586 CPU. According to "show translation" it needs
> 30 ms for transcoding 1 channel from g711 to gsm (and vice versa).
> So, neglecting any overhead caused by channel handling it could
> transcode 30 channels to gsm.
>
> Linux BIOS has support for the EPIA boards, so you can speed up booting
> very much and also disable the VGA port (very useful for production
> deployments....).
>
> > I'm running pebble on a pcengines board, just needed to customize the
> > kernel a bit, haven't been testing asterisk on that yet, but i definatly
> > will in the sooner future.
> >
> > Kind regards,
> > Martin List-Petersen
> > martin (at) list (dash) petersen (dot) net
>
> best regards
>
> Klaus
> --
> Klaus-Peter Junghanns
>
> CEO, CTO
> Junghanns.NET GmbH
> Breite Strasse 13a - 12167 Berlin - Germany
> fon: (de) +49 30 79705390
> fon: (uk) +44 870 1244692
> fax: (de) +49 30 79705391
> iaxtel: 1-700-157-8753
> http://www.Junghanns.NET/asterisk/
>
>
>
> --__--__--
>
> Message: 9
> Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
> From: Michael Bielicki <Michael.Bielicki at Global-Gateway.net>
> To: asterisk-users at lists.digium.com
> Organization: TAAN Consultants Ltd.
> Date: Thu, 17 Jun 2004 10:32:41 +0200
> Reply-To: asterisk-users at lists.digium.com
>
> What is in your config file ?
> zaptel.conf ?
> also, check the crc4 settings
> and
> maybe the wire you are using is wrong since some equippment needs
> crossed wires, some needs straight wires. Crossed would be 1-4 2-5
>
> cheers
>
> Michael
>
> On Thu, 2004-06-17 at 10:28, Wolfgang Pichler wrote:
> > Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43:
> > > On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
> > >
> > > > ... on the card i can see the two leds pulsing red (i think thats
the
> > > > yellow alaram - or i am wrong) ?
> > >
> > > Are you sure it is not a red alarm? That would indicate a loss of
link.
> > > I think you can check with the command zttool.
> > you are right - its a red alarm - zttool says "Red Alarm/Not Open"
> > >
> > > Are you sure the cables are correct?
> > > Have you set the jumpers on the card to E1 and not left them on T1?
> > The jumpers are on E1 - the cables should be ok (they are working with
> > other hardware) - and the card is directly connected to a simens ULAF+
> > STU Desktop (can't really find much information about this device on the
> > net) - which turns off a red led when i load the driver and do a ztcfg.
> > >
> > > I think the leds should turn green when the card senses a correct
> carrier
> > > and framing on the lines.
> > green is always a wounderful color ;-)
> >
> > so, what else could cause this ?
> >
> > wolfgang
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --__--__--
>
> Message: 10
> Date: Thu, 17 Jun 2004 10:33:52 +0200
> From: "Andy Powell" <andy at beagles-den.demon.co.uk>
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just
> Software?
> Reply-To: asterisk-users at lists.digium.com
>
>
> On 16/06/2004 at 22:53 Jay Milk wrote:
>
> >You're correct -- I believe I pointed out in my original post that there
> >is a $200+ difference between a cordless Cisco with/without software.
> >And that's plain ridiculous. Plus, the phone alone isn't worth $500 in
> >hardware -- so we're obviously dealing with GREED here.
> >
> >My knee-jerk response to such business tactics always has been to do it
> >better and cheaper. Six years ago, I was talking to IT personel in
> >industry "X". There were two established mainframe solutions in that
> >industry serving 80% of the market, costing $50K-$75K start-up cost per
> >location, plus $1K+ per seat. Never mind the $10K-$15K monthly
> >"maintenance" cost. Never mind that everyone had to be able to work a
> >terminal with a lovely amber on black, text-based "GUI".
> >
> <snip for brevity>
>
> I think you're missing the point. When you develop hardware or software
you
> need to recoup the cost of development (the period in which you aren't=
> selling
> anything, so not making any money). Now Cisco has it's fingers in many
pies
> so they aren't going to suffer to much from that now, but they do have to=
> fund
> development.
>
> Secondly, Cisco don't really care if their phones are out of your price=
> range,
> they are typically sold as part of a solution costing 10's of 1000's or=
> 100's of
> 1000's of USD/GBP/EUR and (most probably) with big discounts.
>
> Thirdly, If I make a device at a cost of $5 and sell it for $500, some=
> people will
> buy it, up to the point where someone builds a similar device and sells
it=
> for
> $150 ...You have a choice. companies are not charities, they do this to=
> make
> money. This is what we call capitalism.
>
> I don't want to dig at your business, and this isn't intended to but..
what=
> you did
> is look at what was already on offer and it's costs, how it worked etc
and=
> built a
> cheaper solution. The reason you could do this is because you had the=
> exposure
> to the 'system' as was.. i.e. You looked at it and said 'I can do that=
> cheaper' but
> without that original system you probably wouldn't have.
>
> One final point... There are some companies that have this weird feeling=
> that anything
> under a certain amount must be cheap and nasty and not work properly.
These=
> people
> are fools imho, but they do exist...and they wont buy an cheap phone,=
> they'll buy an
> expensive phone, regardless of it's ability... as we've seen recently
some=
> governments
> will even buy helicopters that can't fly in fog or where it's sandy for=
> silly money...
>
> Now I feel dirty...
>
>
> Andy
>
>
>
> --__--__--
>
> Message: 11
> Date: Thu, 17 Jun 2004 10:38:30 +0200 (CEST)
> From: Peter Svensson <psvasterisk at psv.nu>
> To: Asterisk-Users Mailinglist <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
> Reply-To: asterisk-users at lists.digium.com
>
> On Thu, 17 Jun 2004, Wolfgang Pichler wrote:
>
> > > Are you sure the cables are correct?
> > > Have you set the jumpers on the card to E1 and not left them on T1?
> > The jumpers are on E1 - the cables should be ok (they are working with
> > other hardware) - and the card is directly connected to a simens ULAF+
> > STU Desktop (can't really find much information about this device on the
> > net) - which turns off a red led when i load the driver and do a ztcfg.
>
> Then the tx (from TE410P to the Siemens equipment) circuit is ok but the
> rx may not be.
>
> > > I think the leds should turn green when the card senses a correct
> carrier
> > > and framing on the lines.
> > green is always a wounderful color ;-)
> >
> > so, what else could cause this ?
>
> I'd try to find out if the cable is wired the way the TE410P expects it to
> be. Do you know the pinout of both ends of the cables? RX (from the TE410P
> point of view) should be on the pins 1-2 at the TE410P end and TX on 4-5.
>
> Peter
> --
> Peter Svensson ! Pgp key available by finger, fingerprint:
> <petersv at psv.nu> ! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF
> ------------------------------------------------------------------------
> Remember, Luke, your source will be with you... always...
>
>
>
>
>
> --__--__--
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list