[Asterisk-Users] Problems with PRI with T410 messages
Robinson Tim-W10277
Tim.Robinson at motorola.com
Thu Jun 17 03:37:30 MST 2004
This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture.
Rgds
Tim
-----Original Message-----
From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Aimable
Sent: 17 June 2004 10:29
To: asterisk-users at lists.digium.com
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-----
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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.
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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:
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> >
>
> _______________________________________________
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>
--__--__--
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...
--__--__--
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