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@terracom.rw
-----Original Message-----
From: asterisk-users-request@lists.digium.com
[mailto:asterisk-users-request@lists.digium.com]
Sent: Thursday, June 17, 2004 10:56 AM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs
Send Asterisk-Users mailing list submissions to
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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@boltblue.com>
To: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] Soekris Engineering net4801
Date: Thu, 17 Jun 2004 08:34:01 +0100
Reply-To: asterisk-users@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@avenue500.com>
To: <asterisk-users@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@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@psv.nu>
To: Asterisk-Users Mailinglist <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
Reply-To: asterisk-users@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@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@mail.mn-solutions.de>
To: asterisk-users@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@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@ecomtel.com.au>
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Calling the firefly network?
Reply-To: asterisk-users@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@ru.ac.za>
To: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] IAX2 no compatible codecs
Date: Thu, 17 Jun 2004 10:22:10 +0200
Reply-To: asterisk-users@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@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Hart
> Sent: 17 June 2004 09:19 AM
> To: asterisk-users@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@lists.digium.com
> >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
> Adam Hart
> >>Sent: 17 June 2004 08:29 AM
> >>To: asterisk-users@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@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@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@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@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@dialog-telekom.at>
To: Asterisk-Users Mailinglist <Asterisk-Users@lists.digium.com>
Date: Thu, 17 Jun 2004 10:28:09 +0200
Reply-To: asterisk-users@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@junghanns.net>
To: asterisk-users@lists.digium.com
Organization: Junghanns.NET GmbH
Date: Thu, 17 Jun 2004 10:11:11 +0200
Reply-To: asterisk-users@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@Global-Gateway.net>
To: asterisk-users@lists.digium.com
Organization: TAAN Consultants Ltd.
Date: Thu, 17 Jun 2004 10:32:41 +0200
Reply-To: asterisk-users@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@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@beagles-den.demon.co.uk>
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just
Software?
Reply-To: asterisk-users@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@psv.nu>
To: Asterisk-Users Mailinglist <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
Reply-To: asterisk-users@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@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@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
End of Asterisk-Users Digest
Robinson Tim-W10277
2004-Jun-17 03:37 UTC
[Asterisk-Users] Problems with PRI with T410 messages
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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aimable
Sent: 17 June 2004 10:29
To: asterisk-users@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
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-----Original Message-----
From: asterisk-users-request@lists.digium.com
[mailto:asterisk-users-request@lists.digium.com]
Sent: Thursday, June 17, 2004 10:56 AM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs
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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@boltblue.com>
To: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] Soekris Engineering net4801
Date: Thu, 17 Jun 2004 08:34:01 +0100
Reply-To: asterisk-users@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@avenue500.com>
To: <asterisk-users@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@lists.digium.com
This is a multi-part message in MIME format.
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charset="iso-8859-1"
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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
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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>
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--__--__--
Message: 3
Date: Thu, 17 Jun 2004 09:43:37 +0200 (CEST)
From: Peter Svensson <psvasterisk@psv.nu>
To: Asterisk-Users Mailinglist <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
Reply-To: asterisk-users@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@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@mail.mn-solutions.de>
To: asterisk-users@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@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@ecomtel.com.au>
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Calling the firefly network?
Reply-To: asterisk-users@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@ru.ac.za>
To: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] IAX2 no compatible codecs
Date: Thu, 17 Jun 2004 10:22:10 +0200
Reply-To: asterisk-users@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@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Hart
> Sent: 17 June 2004 09:19 AM
> To: asterisk-users@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@lists.digium.com
> >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
> Adam Hart
> >>Sent: 17 June 2004 08:29 AM
> >>To: asterisk-users@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@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@lists.digium.com
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
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> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >
> >
> > _______________________________________________
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> >
>
> _______________________________________________
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>
--__--__--
Message: 7
Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
From: Wolfgang Pichler <madmin@dialog-telekom.at>
To: Asterisk-Users Mailinglist <Asterisk-Users@lists.digium.com>
Date: Thu, 17 Jun 2004 10:28:09 +0200
Reply-To: asterisk-users@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@junghanns.net>
To: asterisk-users@lists.digium.com
Organization: Junghanns.NET GmbH
Date: Thu, 17 Jun 2004 10:11:11 +0200
Reply-To: asterisk-users@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@Global-Gateway.net>
To: asterisk-users@lists.digium.com
Organization: TAAN Consultants Ltd.
Date: Thu, 17 Jun 2004 10:32:41 +0200
Reply-To: asterisk-users@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@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 10
Date: Thu, 17 Jun 2004 10:33:52 +0200
From: "Andy Powell" <andy@beagles-den.demon.co.uk>
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just
Software?
Reply-To: asterisk-users@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@psv.nu>
To: Asterisk-Users Mailinglist <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria)
Reply-To: asterisk-users@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@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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Send traces. ----- Original Message ----- From: "Aimable" <aimable@terracom.rw> To: <asterisk-users@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 100switch> 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 theasterisk> 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 lineis> 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 fromSER?> > > Thanks > Habiyakare Aimable > Phone Services > TERRACOM Broadband > aimable@terracom.rw > > > > > -----Original Message----- > From: asterisk-users-request@lists.digium.com > [mailto:asterisk-users-request@lists.digium.com] > Sent: Thursday, June 17, 2004 10:56 AM > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs > > Send Asterisk-Users mailing list submissions to > asterisk-users@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@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@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@boltblue.com> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] Soekris Engineering net4801 > Date: Thu, 17 Jun 2004 08:34:01 +0100 > Reply-To: asterisk-users@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@avenue500.com> > To: <asterisk-users@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@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@psv.nu> > To: Asterisk-Users Mailinglist <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria) > Reply-To: asterisk-users@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@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@mail.mn-solutions.de> > To: asterisk-users@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@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@ecomtel.com.au> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Calling the firefly network? > Reply-To: asterisk-users@lists.digium.com > > Is there a way to register with or call the firefly network from anAsterisk> 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@ru.ac.za> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] IAX2 no compatible codecs > Date: Thu, 17 Jun 2004 10:22:10 +0200 > Reply-To: asterisk-users@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: Callrejected> by 146.231.125.65: No compatible Codecs > > > I can see the inconsitency with the FORMAT header of the two setupmessages.> According to the IAX protocol spec. The FORMAT (0x4) represents G.711U-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 secondmessage -> an undefined format. The quesiton is how the * manager API causes * to > inititiate an IAX call with this FORMAT type (0x64)??????? An how we canfix> it ???????. > > Any ideas, anyone > Thanks again Adam for the help > Cheers > Jason > > > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Hart > > Sent: 17 June 2004 09:19 AM > > To: asterisk-users@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@lists.digium.com > > >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > > Adam Hart > > >>Sent: 17 June 2004 08:29 AM > > >>To: asterisk-users@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@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@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@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@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@dialog-telekom.at> > To: Asterisk-Users Mailinglist <Asterisk-Users@lists.digium.com> > Date: Thu, 17 Jun 2004 10:28:09 +0200 > Reply-To: asterisk-users@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 correctcarrier> > 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@junghanns.net> > To: asterisk-users@lists.digium.com > Organization: Junghanns.NET GmbH > Date: Thu, 17 Jun 2004 10:11:11 +0200 > Reply-To: asterisk-users@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@Global-Gateway.net> > To: asterisk-users@lists.digium.com > Organization: TAAN Consultants Ltd. > Date: Thu, 17 Jun 2004 10:32:41 +0200 > Reply-To: asterisk-users@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 thatsthe> > > > yellow alaram - or i am wrong) ? > > > > > > Are you sure it is not a red alarm? That would indicate a loss oflink.> > > 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@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@beagles-den.demon.co.uk> > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just > Software? > Reply-To: asterisk-users@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 softwareyou> 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 manypies> 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 sellsit> 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 etcand> 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 recentlysome> 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@psv.nu> > To: Asterisk-Users Mailinglist <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria) > Reply-To: asterisk-users@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@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@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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
Now what is the normal behavior and how can I set it so that * behaves
normally?
-----Original Message-----
From: asterisk-users-request@lists.digium.com
[mailto:asterisk-users-request@lists.digium.com]
Sent: Thursday, June 17, 2004 2:06 PM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users digest, Vol 1 #4186 - 11 msgs
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Today's Topics:
1. Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n
et? (Alessio Focardi)
2. RE: LDAP synchronization script (Stefan de Konink)
3. Re: Problems with PRI with T410 messages (CW_ASN)
4. RE: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from jung
hanns.n et? (Robinson Tim-W10277)
5. RE: LDAP synchronization script (David Hajek)
6. Zapata.conf & Signaling for Bulgaria (PSTN: Siemens PABX) (Miroslav
Nachev)
7. Re: embedded Asterisk (listas iPfone)
8. Re[4]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n
et? (Alessio Focardi)
9. SFTP (Dean Collins)
10. Re: embedded Asterisk (Stefan de Konink)
--__--__--
Message: 1
Date: Thu, 17 Jun 2004 13:18:51 +0200
From: Alessio Focardi <afoc@interconnessioni.it>
To: Robinson Tim-W10277 <Tim.Robinson@motorola.com>,
asterisk-users@lists.digium.com
Subject: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from
junghanns.n et?
Reply-To: asterisk-users@lists.digium.com
Hello Robinson,
Thursday, June 17, 2004, 12:42:21 PM, you wrote:
RTW> Please can you explain in more details as to what your
RTW> problem is? I have 2 cards working in one PC, but have had
RTW> problems with Dell motherboards.
voice is out of sync, it syncs for some second if I run something over
another console, like, for instance a "find /" then slips away again.
I suspect an Irq problem, what do you think ? What kind of problems
have you found with dell's ?
Tnx for the help !
--
Best regards,
Alessio mailto:afoc@interconnessioni.it
--__--__--
Message: 2
Date: Thu, 17 Jun 2004 13:12:25 +0200 (CEST)
From: Stefan de Konink <skinkie@xs4all.nl>
To: David Hajek <david.hajek@systinet.com>
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] LDAP synchronization script
Reply-To: asterisk-users@lists.digium.com
I'm planning to incorporate this (native and dynamic) LDAP for my own
system on short term. Do you have any LDAP design in mind?
Stefan
On Thu, 17 Jun 2004, Jeremy Jones wrote:
>
> > David Hajek
> > Sent: Thursday, June 17, 2004 2:41 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] LDAP synchronization script
> >
> > Hello,
> >
> > I understand there's no possibility to have asterisk configuration
> > (sipusers, extensions, voicemail) in LDAP right now. I'm thinking
> > about put the (sipusers, extensions, voicemail) info in LDAP
> > and then run
> > some synchronization script on the asterisk server which will build up
> > appropriate configuration files and reload asterisk.
> >
> > I'm sure this script is already around. Can some share one with
me/us?
> >
>
> Not aware of any scripts like that, but...
> you could use the odbc support in asterisk in conjunction with some
> slick odbc-ldap connectivity.
>
> Jeremy Jones
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@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: 3
From: "CW_ASN" <cw_asn@fibertel.com.ar>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] Problems with PRI with T410 messages
Date: Thu, 17 Jun 2004 08:13:03 -0300
Reply-To: asterisk-users@lists.digium.com
>
> 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
>
> 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
Call Proceeding can be sent only by transit network, not by the local switch
or pbx. AFAIK, * behavior for this scenario is like as local switch.
Certainly, this is not a normal behavior.
Regards,
Gus
--__--__--
Message: 4
From: Robinson Tim-W10277 <Tim.Robinson@motorola.com>
To: asterisk-users@lists.digium.com
Cc: "'Alessio Focardi'" <afoc@interconnessioni.it>
Subject: RE: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from jung
hanns.n et?
Date: Thu, 17 Jun 2004 12:19:12 +0100
Reply-To: asterisk-users@lists.digium.com
Hi Alessio
Yes, the problems you report do seem similar to the issues I had. I found
on the Dells that the audio prompts were very choppy and played slower than
normal. Occasionally there would be 'bursts' oav a second or so of
'good'
audio.
I also suspected IRQ issues but the Dell Mobos had no way of adjusting them.
Best thing is to try and get the card on its own unshared IRQ. If this
fails, you either have to try a different pc, or collect 600 euros together
and send them to Junghanns.net, and they will send you a quadBRI card that
does not have this problem.
Rgds
Tim
-----Original Message-----
From: Alessio Focardi [mailto:afoc@interconnessioni.it]
Sent: 17 June 2004 12:19
To: Robinson Tim-W10277; asterisk-users@lists.digium.com
Subject: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from
junghanns.n et?
Hello Robinson,
Thursday, June 17, 2004, 12:42:21 PM, you wrote:
RTW> Please can you explain in more details as to what your problem is?
RTW> I have 2 cards working in one PC, but have had problems with Dell
RTW> motherboards.
voice is out of sync, it syncs for some second if I run something over
another console, like, for instance a "find /" then slips away again.
I suspect an Irq problem, what do you think ? What kind of problems have you
found with dell's ?
Tnx for the help !
--
Best regards,
Alessio mailto:afoc@interconnessioni.it
--__--__--
Message: 5
From: "David Hajek" <david.hajek@systinet.com>
To: "'Stefan de Konink'" <skinkie@xs4all.nl>
Cc: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] LDAP synchronization script
Date: Thu, 17 Jun 2004 13:22:21 +0200
Organization: Systinet
Reply-To: asterisk-users@lists.digium.com
I think I'll use something from this article -
http://www.marko.net/asterisk/archives/0205/0006.html
-David
> -----Original Message-----
> From: Stefan de Konink [mailto:skinkie@xs4all.nl]
> Sent: Thursday, June 17, 2004 1:12 PM
> To: David Hajek
> Cc: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] LDAP synchronization script
>
> I'm planning to incorporate this (native and dynamic) LDAP
> for my own system on short term. Do you have any LDAP design in mind?
>
> Stefan
>
> On Thu, 17 Jun 2004, Jeremy Jones wrote:
>
> >
> > > David Hajek
> > > Sent: Thursday, June 17, 2004 2:41 AM
> > > To: asterisk-users@lists.digium.com
> > > Subject: [Asterisk-Users] LDAP synchronization script
> > >
> > > Hello,
> > >
> > > I understand there's no possibility to have asterisk
> configuration
> > > (sipusers, extensions, voicemail) in LDAP right now. I'm
thinking
> > > about put the (sipusers, extensions, voicemail) info in LDAP and
> > > then run some synchronization script on the asterisk server which
> > > will build up appropriate configuration files and reload
asterisk.
> > >
> > > I'm sure this script is already around. Can some share
> one with me/us?
> > >
> >
> > Not aware of any scripts like that, but...
> > you could use the odbc support in asterisk in conjunction with some
> > slick odbc-ldap connectivity.
> >
> > Jeremy Jones
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@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: 6
Date: Thu, 17 Jun 2004 14:23:17 +0200
From: Miroslav Nachev <miro@space-comm.com>
Organization: COSMOS Software Enterprises, Ltd.
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Zapata.conf & Signaling for Bulgaria (PSTN:
Siemens PABX)
Reply-To: asterisk-users@lists.digium.com
Hi,
How to configure our ZAPATA.CONF in case that the PSTN in Bulgaria
is based on Siemens equipment?
Now my configuration is:
[channels]
language=en
busydetect=no
when is "yes" I have problems with answering of FXO when FXS line is
open
callprogress=no
when is "yes" I have problems with answering of FXO when FXS line is
open
; interfaces for internal analog phones
signalling=fxo_ks
threewaycalling=yes
; interfaces for external PSTN line
signalling=fxs_ks
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel: (+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: m_natchev@yahoo.com
miro@space-comm.com
http://www.space-comm.com
Post address:
P. O. Box 941,
1000 Sofia,
Bulgaria
Office address:
ap. 9, fl. 4,
"11 August" str., No. 43,
1202 Sofia,
Bulgaria
--__--__--
Message: 7
From: "listas iPfone" <listas@ipfone.com.br>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] embedded Asterisk
Date: Thu, 17 Jun 2004 08:24:14 -0300
Reply-To: asterisk-users@lists.digium.com
Hi
That rescue disk sugestion seems to be very good...
Let?s see if i undestood:
1. burn the rescue iso
1. copy the rescue disk to a hard drive
2. compile asterisk
3. copy all to the flash disk
It is that simple?
Miklos
----- Original Message -----
From: "Klaus-Peter Junghanns" <kpj@junghanns.net>
To: <asterisk-users@lists.digium.com>
Sent: Thursday, June 17, 2004 5:11 AM
Subject: Re: [Asterisk-Users] embedded Asterisk
> 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/
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@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: 8
Date: Thu, 17 Jun 2004 13:33:13 +0200
From: Alessio Focardi <afoc@interconnessioni.it>
To: Robinson Tim-W10277 <Tim.Robinson@motorola.com>
Cc: asterisk-users@lists.digium.com
Subject: Re[4]: [Asterisk-Users] HFC ISDN card with bristuff from jung
hanns.n et?
Reply-To: asterisk-users@lists.digium.com
Hello Robinson,
Thursday, June 17, 2004, 1:19:12 PM, you wrote:
RTW> Hi Alessio
RTW> Yes, the problems you report do seem similar to the issues
RTW> I had. I found on the Dells that the audio prompts were very
RTW> choppy and played slower than normal. Occasionally there would
RTW> be 'bursts' oav a second or so of 'good' audio.
RTW> I also suspected IRQ issues but the Dell Mobos had no way
RTW> of adjusting them. Best thing is to try and get the card on its
RTW> own unshared IRQ. If this fails, you either have to try a
RTW> different pc, or collect 600 euros together and send them to
RTW> Junghanns.net, and they will send you a quadBRI card that does
RTW> not have this problem.
Well card has his own irq, I will try to tweak bios parameters to see if
something gets better.
Meanwhile since I orderer 2 dell's yesterday hoping to solve the problem
I'm
going to bang my head
against the wall until they arrive ....
Tnx for now !
--
Best regards,
Alessio mailto:afoc@interconnessioni.it
--__--__--
Message: 9
Date: Thu, 17 Jun 2004 21:35:40 +1000
From: "Dean Collins" <dean@collins.net.pr>
To: <asterisk-users@lists.digium.com>
Subject: [Asterisk-Users] SFTP
Reply-To: asterisk-users@lists.digium.com
This is a multi-part message in MIME format.
------_=_NextPart_001_01C4545F.37D595B0
Content-Type: text/plain;
charset="us-ascii"
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I'm having problems with a new install of Asterisk (I had to reinstall
because hard drive failed). I've used debian net install this time and
for some reason WS FTP will not connect using SFTP (it keeps coming back
with username and password fail) but when I use Putty to connect with
the same password and username it works no problems.
=20
Any thoughts?
=20
Any other programs I can use for SFTP?
=20
=20
Cheers,
Dean
=20
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<div class=3DSection1>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'>I’m having problems with a new install of
Asterisk (I
had to reinstall because hard drive failed). I’ve used debian net
install
this time and for some reason WS FTP will not connect using SFTP (it keeps
coming back with username and password fail) but when I use Putty to connect
with the same password and username it works no
problems.<o:p></o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'>Any
thoughts?<o:p></o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'>Any other programs I can use for
SFTP?<o:p></o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'>Cheers,<o:p></o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'>Dean<o:p></o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
</div>
</body>
</html>
------_=_NextPart_001_01C4545F.37D595B0--
--__--__--
Message: 10
Date: Thu, 17 Jun 2004 13:48:34 +0200 (CEST)
From: Stefan de Konink <skinkie@xs4all.nl>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] embedded Asterisk
Reply-To: asterisk-users@lists.digium.com
On Thu, 17 Jun 2004, listas iPfone wrote:
> 1. burn the rescue iso
mount -o loop -t iso9660 /file /mnt/loop
> 1. copy the rescue disk to a hard drive
cp -dpR /mnt/loop/* /new/location
> 2. compile asterisk
make PREFIX=/new/location install (check if asterisk don't copy all
development non-sence)
> 3. copy all to the flash disk
fdisk /dev/hdX[0-9]
make partitions
mkfs.ext2 /dev/hdX[0-9]
mount -t ext2 /dev/hdX[0-9] /mnt/flash
cp -dpR /new/location /mnt/flash
> It is that simple?
Probably you want something that actually boots the system too. I don't
know if the ISOLINUX pakage supports a LILO kind of thing, but I guess it
does. That should be in the MBR of your flash disk and you could probably
boot it. I wrote the instructions by mind, so probably something is
missing :)
Stefan
>
> Miklos
>
> ----- Original Message -----
> From: "Klaus-Peter Junghanns" <kpj@junghanns.net>
> To: <asterisk-users@lists.digium.com>
> Sent: Thursday, June 17, 2004 5:11 AM
> Subject: Re: [Asterisk-Users] embedded Asterisk
>
>
> > 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/
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@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@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@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
End of Asterisk-Users Digest
Robinson Tim-W10277
2004-Jun-17 05:29 UTC
[Asterisk-Users] Problems with PRI with T410 messages
I do not believe you are correct. We see CALL PROCEEDING in both directions as
part of the normal ISDN call setup process. See trace below.
Asterisk sends 'CALL PROCEEDING' followed immediately by
'ALERTING'. CALL PROCEEDING is normally an acknowledgement to a SETUP.
See Q931 below:
3.1.2 CALL PROCEEDING
This message is sent by the called user to the network or by the network to the
calling user to indicate that requested call establishment has been initiated
and no more call establishment information will be accepted. See Table 3-3.
ALERTING has a very specific meaning:
3.2.1 ALERTING
This message is sent by the called user to the network to indicate that called
user alerting has been initiated. See Table 3 23.
i.e. the channel to the called party has been established, and the phone at the
other end is physically ringing or making some other indication that an incoming
call is there to be answered.
It is 'ALERTING' that is being sent in the wrong place, as Asterisk
sends 'ALERTING' before the remote party (be it a SIP or IAX channel) is
actually 'ringing'. Receipt of 'ALERTING' from the called party
is the trigger for the calling party to be presented with 'ringback
tone'. So to send a 'RELEASE' message with 'busy' after the
caller has been told the phone is ringing is not a logical thing to do, and
causes a lot of problems here.
It needs fixing!!!!
Rgds
Tim
Connected to Asterisk CVS-D2004.05.25.23.00.00-06/14/04-12:46:31 currently
running on localhost (pid = 4875)
mote UNIX connection
< Protocol Discriminator: Q.931 (8) len=40
< Call Ref: len= 1 (reference 1/0x1) (Originator)
< Message type: SETUP (5)
< Sending Complete (len= 4)
< Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: 3.1kHz audio (16)
< Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
< Ext: 1 User information layer 1: A-Law (35)
< Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive
Dchan: 0
< ChanSel: B1 channel
]
< Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Network beyond the interworking point (10)
< Ext: 1 Progress Description: Call is not
end-to-end ISDN; further call progress information may be available inband. (1)
]
< Calling Number (len=18) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
< Called Number (len= 5) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown
Number Plan (0) '14' ]
-- Making new call for cr 1
-- Processing Q.931 Call Setup
-- Processing IE 33 (Sending Complete)
-- Processing IE 4 (Bearer Capability)
-- Processing IE 24 (Channel Identification)
-- Processing IE 30 (Progress Indicator)
-- Processing IE 108 (Calling Party Number)
-- Processing IE 112 (Called Party Number)> Protocol Discriminator: Q.931 (8) len=7
> Call Ref: len= 1 (reference 129/0x81) (Terminator)
> Message type: CALL PROCEEDING (2)
> Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive
Dchan: 0
> ChanSel: B1 channel
]> Protocol Discriminator: Q.931 (8) len=7
> Call Ref: len= 1 (reference 129/0x81) (Terminator)
> Message type: ALERTING (1)
> Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive
Dchan: 0
> ChanSel: B1 channel
]
-- Executing Wait("Zap/1-1", "2") in new stack
-- Accepting call from '00441256790000' to '14' on channel
1, span 1
-- Executing Goto("Zap/1-1", "default|8714|1") in new
stack
-- Goto (default,8714,1)
-- Executing SetMusicOnHold("Zap/1-1", "default") in new
stack
-- Executing Answer("Zap/1-1", "") in new
stack> Protocol Discriminator: Q.931 (8) len=11
> Call Ref: len= 1 (reference 129/0x81) (Terminator)
> Message type: CONNECT (7)
> Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive
Dchan: 0
> ChanSel: B1 channel
]> Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
> Ext: 1 Progress Description: Called
equipment is non-ISDN. (2) ]
-- Executing SayDigits("Zap/1-1", "00441256790000") in
new stack
-- Playing 'digits/0' (language 'en')
< Protocol Discriminator: Q.931 (8) len=4
< Call Ref: len= 1 (reference 1/0x1) (Originator)
< Message type: CONNECT ACKNOWLEDGE (15)
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of CW_ASN
Sent: 17 June 2004 12:13
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Problems with PRI with T410 messages
>
> 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
>
> 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
Call Proceeding can be sent only by transit network, not by the local switch or
pbx. AFAIK, * behavior for this scenario is like as local switch. Certainly,
this is not a normal behavior.
Regards,
Gus
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