Hello:
I have successfully install spandsp and patch asterisk with it. But when
I received a Fax is garble or shrink. Does any one know why???... Am using a
PRI T100P card to receive the fax and save it to a tiff file... Any help
will be greatly appreciated. Here are the versions.
Latest csv from asterisk,
spandsp-0.0.1k.tar.gz
redhat 7.3
T100P has its own IRQ.
Any help will be greatly appreciated...
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
asterisk-users-request@lists.digium.com
Sent: Friday, January 14, 2005 2:28 AM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users Digest, Vol 6, Issue 199
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-owner@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: Problem patching asterisk CVS with SpanDSP (Matt Riddell)
2. DIAX 0.9.9g more features and higher stability (Dan)
3. R2/MFC Mexico FREE calls to test chan_unicall (Gonzalo Gasca Meza)
4. Re: Updated kphone 4.0.5, asterisk v1.0.3 (Howard Lowndes)
5. RE: [Asterisk-biz] SS7 and Asterisk solution (Rob Lith)
6. RE: TE410P card in an HP-Compaq DL380 G4 server (Joshua McAdam)
7. Polycom Shared Call Appearance (John Bittner)
8. Re: SER vs Asterisk for SIP (Julio Tejera)
9. Re: How to set asterisk NOT to answer incoming lines?
(Steven Critchfield)
10. Limit outgoing trunk calls (Mike Sander)
11. RE: Agentcallbackogin withoutanyuserinputafter extension is
dialed. (Florian Overkamp)
----------------------------------------------------------------------
Message: 1
Date: Fri, 14 Jan 2005 19:00:11 +1300
From: Matt Riddell <matt.riddell@sineapps.com>
Subject: Re: [Asterisk-Users] Problem patching asterisk CVS with
SpanDSP
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <41E75FEB.3040305@sineapps.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Keith LeClaire Jr wrote:> I'm trying to patch the current asterisk CVS with
spandsp-0.0.1k.tar.gz.
> Everything compiles fine but when I go to patch the asterisk/apps/Makefile
> it fails:
:-))))))))))))))))))))))
Sorry, that's my excuse for the biggest smile ever.
I just posted the solution yesterday/day before for this exact thing.
Have you just subscribed or were you here yesterday too?
:-)
Drop me a line off list if you would like me to talk you though this
(free of course). The reason I say off-list is because the solution
will already end up in the mailing list...
This is one of the simplest patches in the world to apply. I can talk
you through it, or you could have a look (hint +xxx means add xxx, don't
forget that the spaces are actually tabs in the Makefile).
--
Cheers,
Matt Riddell
_______________________________________________
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
------------------------------
Message: 2
Date: Fri, 14 Jan 2005 08:05:36 +0200
From: "Dan" <danto@rdslink.ro>
Subject: [Asterisk-Users] DIAX 0.9.9g more features and higher
stability
To: <asterisk-users@lists.digium.com>
Message-ID: <003501c4f9ff$27c1de50$0121a8c0@dantenc4010>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
Hi all,
DIAX 0.9.9g is available for download (including the updated help file and
web page) from the following locations:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
What's new in 0.9.9g (from 0.9.9f):
- during a call, accept DTMF tones as monitored events to trigger output
commands
- call timer on the phone display
- Swedish language added
- can run a command from the monitoring definition form, to test it
- ENTER key validate all fields in the Registration form
- you can select both preffered and accepted codecs
- do not autoresize main form when receiving a call and monitoring activated
- use /m switch to start DIAX minimized
- saving only main form position, all others auto positioning relative to
the main form
solved bugs:
- crash when trying to dial without registration server defined
- Config Audio form positioning issue
- not saving the main form when closing the app from the systray
- X10 send error if CM11/12 interface has some commands in the receiver
buffer
- error if trying to delete for the second time the log file
- unexpected crashes when registered with IAXTEL and/or other remote servers
As usual, please send me your feedback.
Best regards,
Dan
------------------------------
Message: 3
Date: Thu, 13 Jan 2005 22:07:46 -0800 (PST)
From: Gonzalo Gasca Meza <xomeboy@yahoo.com>
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test
chan_unicall
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <20050114060746.34380.qmail@web60707.mail.yahoo.com>
Content-Type: text/plain; charset="us-ascii"
Miguel,
Congrats, i was testing your R2/MFC link, and I was able to made lots of
calls, all of them worked fine.Thanks for setting up this link.
When i hang up, there were no dead air, music on hold worked fine, when I
called to a conference worked fine also, busy line Telmex recording worked
also fine. Please let me know if there is anything I can help you with or if
you want to test something.
Thanks again!
---------------------------------
Do you Yahoo!?
Yahoo! Mail - Easier than ever with enhanced search. Learn more.
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Message: 4
Date: Fri, 14 Jan 2005 17:14:26 +1100
From: Howard Lowndes <lannet@lannet.com.au>
Subject: Re: [Asterisk-Users] Updated kphone 4.0.5, asterisk v1.0.3
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <1105683264.4323.11.camel@lan-255-17.lan.lannet.com.au>
Content-Type: text/plain
On Fri, 2005-01-14 at 15:09, Andrew McRory wrote:> I have uploaded kphone and asterisk CVS stable. These packages are built
> for Fedora Core 1 and this asterisk release should fix the non-root
> permissions problem I worte about...
>
> ftp://ftp.linuxsys.com/pub/releases/FC1/
OK, there are a number of issues I have detected.
The error message about closing other applications using the sound card
is definitly repated to the SIP SUBSCRIBE packets.
When I run it from an xterm, on hangup it seg faults. This does not
happen when I run it from a KDE panel button.
The DTMF tones generated from the on-screen keypad appear not to be
recognised by *.
--
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
------------------------------------------
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
------------------------------------------
"Flatter government, not fatter government;
Get rid of the Australian states."
------------------------------
Message: 5
Date: Fri, 14 Jan 2005 08:16:22 +0200
From: "Rob Lith" <rob@connection-telecom.com>
Subject: [Asterisk-Users] RE: [Asterisk-biz] SS7 and Asterisk solution
To: "'Commercial and Business-Oriented Asterisk Discussion'"
<asterisk-biz@lists.digium.com>, <rehan1@rehan.com>
Cc: asterisk-users@lists.digium.com
Message-ID: <200501140616.j0E6GPag004475@aphrodite.dbuzz.net>
Content-Type: text/plain; charset="us-ascii"
Tracy, one example I can think of is here in South Africa, when VoIP is
deregulated on the 1st February the very first trick the incumbent monopoly
is going to pull out of its hat it saying that to interconnect with them
you're going to need SS7 - if there is a 'soft' way of doing this in
* then
they'll come up with some excuse that its not approved by the regulator/it
not carrier grade....
Regards
Rob
> -----Original Message-----
> From: asterisk-biz-bounces@lists.digium.com
> [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of
> Tracy R Reed
> Sent: 13 January 2005 23:23
> To: rehan1@rehan.com; Commercial and Business-Oriented
> Asterisk Discussion
> Cc: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-biz] SS7 and Asterisk solution
>
> On Thu, Jan 13, 2005 at 01:44:16PM -0600, Rehan Ahmed spake thusly:
> > can u point us to where we can buy cheap ss7 solution
>
> Can you tell me why you think you need one?
>
> --
> Tracy Reed http://copilotcom.com
> This message is cryptographically signed for your protection.
> Info: http://copilotconsulting.com/sig
>
------------------------------
Message: 6
Date: Fri, 14 Jan 2005 16:30:25 +1000
From: "Joshua McAdam" <josh@tlmtech.com>
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4
server
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users@lists.digium.com>
Message-ID:
<20050114063022.1077221B83@mailsrv01-syd.hosting.mipt.com.au>
Content-Type: text/plain; charset="us-ascii"
Has anyone logged a support issue with HP on this one?
I still haven't been able to get it working so far,
So I'm going to log a support issue here in australia to see what HP can do
about this and was wondering if anyone else has.
Josh
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alexander
Lopez
Sent: Monday, 10 January 2005 4:22 PM
To: karlp@fortephones.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Make sure you has a span defined for each port on the TE410P. With out
signaling it would not take interrupts.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Karl H.
Putz
Sent: Monday, January 10, 2005 12:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4
server
I have been having this exact problem with a Tatung dual EMT-64 server
as
well.
I have been trying to get a TE410P running and all looks great, driver
loads, runs ztcfg OK, etc. but no interrupts are ever processed.
One additional piece of info that I have not seen in this thread is that
I
am able to successfully start and run a T100P card in this system. In
the
same PCI slot, wct1xxp driver built from the same CVS HEAD version as
the
wct4xxp.
Just hoping this might shed some light on the problem for any Digium
folks
monitoring the forum.
Karl Putz
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------------------------------
Message: 7
Date: Fri, 14 Jan 2005 01:37:14 -0500
From: "John Bittner" <john@simlab.net>
Subject: [Asterisk-Users] Polycom Shared Call Appearance
To: <asterisk-users@lists.digium.com>
Message-ID: <200501140137156.SM00436@johnb2>
Content-Type: text/plain; charset="us-ascii"
Has anyone got Polycom Shared Call Appearance working with
Asterisk ?
If Asterisk doesn't support this, I am willing to put up a
bounty of 1000 to get it to work.
John Bittner
Simlab.net
Shared Call Appearance Signaling
A shared line is an address of record managed by a server.
The server allows multiple
endpoints to register locations against the address of
record.
SoundPointR IP supports shared call appearances (SCA) using
the SUBSCRIBENOTIFY
method in the "SIP Specific Event Notification" framework
(RFC 3265).
The events used are:
. "call-info" for call appearance state notification.
"line-seize for the phone to ask to seize the line
------------------------------
Message: 8
Date: Fri, 14 Jan 2005 00:43:05 -0600
From: "Julio Tejera" <jat@realityfirewall.net>
Subject: Re: [Asterisk-Users] SER vs Asterisk for SIP
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <015d01c4fa04$4d96d400$e101a8c0@Aceituno>
Content-Type: text/plain; charset="iso-8859-1"
* is a "middleware"
HTH
-------
Ing. Julio Alvarez Tejera
Unix Trends
*BSD, Solaris & Linux
VoIP & CT Solutions Finder
Asterisk PBX Consultant
Costa Rica Land +506-359-9753
USA Toll Free +1-888-899-6269
---------------
"extremely stable systems"
----- Original Message -----
From: "Ashling O'Driscoll" <ashling.odriscoll@cit.ie>
To: <asterisk-users@lists.digium.com>
Sent: Thursday, January 13, 2005 10:57 AM
Subject: RE: [Asterisk-Users] SER vs Asterisk for SIP
>From my (fairly limited) understanding, I think the fundamental
difference is that Asterisk is a pbx (offering all the features
associated with a pbx, voicemail, call transfer, call detail
recording etc) whereas SER is just a sip proxy (albeit a good one).
Therefore Asterisk deals in terms of phones extensions whereas if you
want a system that can contact clients with sip urls, ser will have
to be set up. Also the audio i.e. rtp stream, traverses asterisk i.e.
it acts as a middle man holding onto the call, and if you want the
audio to go peer to peer (which it ideally should with sip), ser is
also needed.
Aisling.
---- Original Message ----
From: vicky@freebsdcluster.net
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] SER vs Asterisk for SIP
Date: Thu, 13 Jan 2005 17:50:39 +0100
>Why is SER considered a better SIPserver than asterisk , why is it
>that SER
>can handle more clients than asterisk can. And if this is just cause
>of say
>poor SIP handling code in asterisk then is there anything being done
>to fix
>it. Just wanted to know why SER claims to be better than asterisk as
>a SIP
>server. ?
>
>--
>regards
>Vikram (http://www.vicramresearch.com)
>_______________________________________________
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>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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------------------------------
Message: 9
Date: Fri, 14 Jan 2005 00:51:50 -0600
From: Steven Critchfield <critch@basesys.com>
Subject: Re: [Asterisk-Users] How to set asterisk NOT to answer
incoming lines?
To: C F <shmaltz@gmail.com>, Asterisk Users Mailing List -
Non-Commercial Discussion <asterisk-users@lists.digium.com>
Message-ID: <1105685510.13831.154.camel@critch>
Content-Type: text/plain
On Thu, 2005-01-13 at 21:09 -0500, C F wrote:> The definition of normal in the case of PBX implementations is up to
> the customer.
You sure are acting like a 'tard lately.
No a customer does not define normal, the market defines normal. A
customer defines an implementation. That implementation is either normal
or an exception/deviation of normal.
> On Thu, 13 Jan 2005 10:44:51 -0600, Steven Critchfield
> <critch@basesys.com> wrote:
> > On Thu, 2005-01-13 at 16:08 +0000, Patrick Lidstone (Personal e-mail)
> > wrote:
> >
> > > I don't think Kelly's response is correct, at least for
TDM FXO
boards.> > > I could not find a way of preventing the FXO board grabbing the
line
> > > when it rang, and subsequent enquiries on this list at the time
> > > suggested that it wasn't actually possible - which is a pity,
as it
> > > means it is impossible to piggy back Asterisk on a POTS line with
other> > > auto-answering equipment (e.g. data collection terminals).
> >
> > It isn't normal to put a PBX on a line shared with other
equipment. It
> > is normal to route the other equipment through the PBX.
> >
> > --
> > Steven Critchfield <critch@basesys.com>
> >
> > _______________________________________________
> > 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
--
Steven Critchfield <critch@basesys.com>
------------------------------
Message: 10
Date: Fri, 14 Jan 2005 18:00:26 +1100
From: "Mike Sander" <mike@corporatebankinginternational.com>
Subject: [Asterisk-Users] Limit outgoing trunk calls
To: <asterisk-users@lists.digium.com>
Message-ID: <20050114070028.9F8D9EE9AB@mail.tyneinternational.com>
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Message: 11
Date: Fri, 14 Jan 2005 08:26:10 +0100
From: "Florian Overkamp" <florian@obsimref.com>
Subject: RE: [Asterisk-Users] Agentcallbackogin
withoutanyuserinputafter extension is dialed.
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <E1CpLr1-0004Ao-00@clio>
Content-Type: text/plain; charset="us-ascii"
Hi,
> -----Original Message-----
> Ok, maybe this is an ignorant question but...... where in memory does
> asterisk store the information and how do I access it?
It's not an ignorant question, but it is like I've stated a few times
now:
The agent information asterisk has is in its own memory and cannot be
accessed easily (you could probably write an AGI script that executes 'show
agents' and parses the output though). That is exactly why you make your
dialplan so every time an agent logs on or off you store your own copy if
the info in the asterisk database where it is available to you for future
reference.
BTW, I know agent technology is a bit better in CVS-HEAD but for my
customers sake (where I have to run a stable branch) I kicked out usage of
agents and now emulate it all with a few AGI scripts.
Florian
------------------------------
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End of Asterisk-Users Digest, Vol 6, Issue 199
**********************************************
Andrew Kohlsmith
2005-Jan-14 08:28 UTC
[Asterisk-Users] Spandsp....And garble incoming fax
On January 14, 2005 10:14 am, Luis Mata wrote:> Latest csv from asterisk, > spandsp-0.0.1k.tar.gzThat's a very old version of spandsp.> redhat 7.3What is the exact version of libtiff? I find you will have strange results with anything other than *stock* 3.5.7. -A.
check out my bug post, I have yet to recieve a successful fax using rxfax. http://www.opencall.org/mantis/bug_view_page.php?bug_id=0000019 and I'm using newest versions of everything. -Matthew ----- Original Message ----- From: "Luis Mata" <mataluis@xtremenetworks.biz> To: <asterisk-users@lists.digium.com> Sent: Friday, January 14, 2005 9:14 AM Subject: [Asterisk-Users] Spandsp....And garble incoming fax> Hello: > > I have successfully install spandsp and patch asterisk with it. Butwhen> I received a Fax is garble or shrink. Does any one know why???... Am usinga> PRI T100P card to receive the fax and save it to a tiff file... Any help > will be greatly appreciated. Here are the versions. > > Latest csv from asterisk, > spandsp-0.0.1k.tar.gz > redhat 7.3 > T100P has its own IRQ. > > Any help will be greatly appreciated... > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > asterisk-users-request@lists.digium.com > Sent: Friday, January 14, 2005 2:28 AM > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 6, Issue 199 > > 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-owner@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: Problem patching asterisk CVS with SpanDSP (Matt Riddell) > 2. DIAX 0.9.9g more features and higher stability (Dan) > 3. R2/MFC Mexico FREE calls to test chan_unicall (Gonzalo Gasca Meza) > 4. Re: Updated kphone 4.0.5, asterisk v1.0.3 (Howard Lowndes) > 5. RE: [Asterisk-biz] SS7 and Asterisk solution (Rob Lith) > 6. RE: TE410P card in an HP-Compaq DL380 G4 server (Joshua McAdam) > 7. Polycom Shared Call Appearance (John Bittner) > 8. Re: SER vs Asterisk for SIP (Julio Tejera) > 9. Re: How to set asterisk NOT to answer incoming lines? > (Steven Critchfield) > 10. Limit outgoing trunk calls (Mike Sander) > 11. RE: Agentcallbackogin withoutanyuserinputafter extension is > dialed. (Florian Overkamp) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 14 Jan 2005 19:00:11 +1300 > From: Matt Riddell <matt.riddell@sineapps.com> > Subject: Re: [Asterisk-Users] Problem patching asterisk CVS with > SpanDSP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <41E75FEB.3040305@sineapps.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Keith LeClaire Jr wrote: > > I'm trying to patch the current asterisk CVS with spandsp-0.0.1k.tar.gz. > > Everything compiles fine but when I go to patch theasterisk/apps/Makefile> > it fails: > > :-)))))))))))))))))))))) > > Sorry, that's my excuse for the biggest smile ever. > > I just posted the solution yesterday/day before for this exact thing. > > Have you just subscribed or were you here yesterday too? > > :-) > > Drop me a line off list if you would like me to talk you though this > (free of course). The reason I say off-list is because the solution > will already end up in the mailing list... > > This is one of the simplest patches in the world to apply. I can talk > you through it, or you could have a look (hint +xxx means add xxx, don't > forget that the spaces are actually tabs in the Makefile). > > -- > Cheers, > > Matt Riddell > _______________________________________________ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) > > > ------------------------------ > > Message: 2 > Date: Fri, 14 Jan 2005 08:05:36 +0200 > From: "Dan" <danto@rdslink.ro> > Subject: [Asterisk-Users] DIAX 0.9.9g more features and higher > stability > To: <asterisk-users@lists.digium.com> > Message-ID: <003501c4f9ff$27c1de50$0121a8c0@dantenc4010> > Content-Type: text/plain; format=flowed; charset="iso-8859-1"; > reply-type=original > > Hi all, > > DIAX 0.9.9g is available for download (including the updated help file and > web page) from the following locations: > http://www.laser.com/dante > or > http://www.geocities.com/tdanro > > What's new in 0.9.9g (from 0.9.9f): > > - during a call, accept DTMF tones as monitored events to trigger output > commands > - call timer on the phone display > - Swedish language added > - can run a command from the monitoring definition form, to test it > - ENTER key validate all fields in the Registration form > - you can select both preffered and accepted codecs > - do not autoresize main form when receiving a call and monitoringactivated> - use /m switch to start DIAX minimized > - saving only main form position, all others auto positioning relative to > the main form > > solved bugs: > - crash when trying to dial without registration server defined > - Config Audio form positioning issue > - not saving the main form when closing the app from the systray > - X10 send error if CM11/12 interface has some commands in the receiver > buffer > - error if trying to delete for the second time the log file > - unexpected crashes when registered with IAXTEL and/or other remoteservers> > > As usual, please send me your feedback. > > > Best regards, > Dan > > > > > ------------------------------ > > Message: 3 > Date: Thu, 13 Jan 2005 22:07:46 -0800 (PST) > From: Gonzalo Gasca Meza <xomeboy@yahoo.com> > Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test > chan_unicall > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <20050114060746.34380.qmail@web60707.mail.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > > Miguel, > > Congrats, i was testing your R2/MFC link, and I was able to made lots of > calls, all of them worked fine.Thanks for setting up this link. > > When i hang up, there were no dead air, music on hold worked fine, when I > called to a conference worked fine also, busy line Telmex recording worked > also fine. Please let me know if there is anything I can help you with orif> you want to test something. > > Thanks again! > > > > > > > > > > --------------------------------- > Do you Yahoo!? > Yahoo! Mail - Easier than ever with enhanced search. Learn more. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: >http://lists.digium.com/pipermail/asterisk-users/attachments/20050113/87c34f> 80/attachment-0001.htm > > ------------------------------ > > Message: 4 > Date: Fri, 14 Jan 2005 17:14:26 +1100 > From: Howard Lowndes <lannet@lannet.com.au> > Subject: Re: [Asterisk-Users] Updated kphone 4.0.5, asterisk v1.0.3 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <1105683264.4323.11.camel@lan-255-17.lan.lannet.com.au> > Content-Type: text/plain > > On Fri, 2005-01-14 at 15:09, Andrew McRory wrote: > > I have uploaded kphone and asterisk CVS stable. These packages are built > > for Fedora Core 1 and this asterisk release should fix the non-root > > permissions problem I worte about... > > > > ftp://ftp.linuxsys.com/pub/releases/FC1/ > > OK, there are a number of issues I have detected. > > The error message about closing other applications using the sound card > is definitly repated to the SIP SUBSCRIBE packets. > > When I run it from an xterm, on hangup it seg faults. This does not > happen when I run it from a KDE panel button. > > The DTMF tones generated from the on-screen keypad appear not to be > recognised by *. > -- > Howard. > LANNet Computing Associates; > Your Linux people <http://www.lannetlinux.com> > ------------------------------------------ > "When you just want a system that works, you choose Linux; > when you want a system that just works, you choose Microsoft." > ------------------------------------------ > "Flatter government, not fatter government; > Get rid of the Australian states." > > > > > ------------------------------ > > Message: 5 > Date: Fri, 14 Jan 2005 08:16:22 +0200 > From: "Rob Lith" <rob@connection-telecom.com> > Subject: [Asterisk-Users] RE: [Asterisk-biz] SS7 and Asterisk solution > To: "'Commercial and Business-Oriented Asterisk Discussion'" > <asterisk-biz@lists.digium.com>, <rehan1@rehan.com> > Cc: asterisk-users@lists.digium.com > Message-ID: <200501140616.j0E6GPag004475@aphrodite.dbuzz.net> > Content-Type: text/plain; charset="us-ascii" > > Tracy, one example I can think of is here in South Africa, when VoIP is > deregulated on the 1st February the very first trick the incumbentmonopoly> is going to pull out of its hat it saying that to interconnect with them > you're going to need SS7 - if there is a 'soft' way of doing this in *then> they'll come up with some excuse that its not approved by the regulator/it > not carrier grade.... > > Regards > Rob > > > -----Original Message----- > > From: asterisk-biz-bounces@lists.digium.com > > [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of > > Tracy R Reed > > Sent: 13 January 2005 23:23 > > To: rehan1@rehan.com; Commercial and Business-Oriented > > Asterisk Discussion > > Cc: asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-biz] SS7 and Asterisk solution > > > > On Thu, Jan 13, 2005 at 01:44:16PM -0600, Rehan Ahmed spake thusly: > > > can u point us to where we can buy cheap ss7 solution > > > > Can you tell me why you think you need one? > > > > -- > > Tracy Reed http://copilotcom.com > > This message is cryptographically signed for your protection. > > Info: http://copilotconsulting.com/sig > > > > > > > ------------------------------ > > Message: 6 > Date: Fri, 14 Jan 2005 16:30:25 +1000 > From: "Joshua McAdam" <josh@tlmtech.com> > Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 > server > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: > <20050114063022.1077221B83@mailsrv01-syd.hosting.mipt.com.au> > Content-Type: text/plain; charset="us-ascii" > > Has anyone logged a support issue with HP on this one? > > I still haven't been able to get it working so far, > So I'm going to log a support issue here in australia to see what HP cando> about this and was wondering if anyone else has. > > Josh > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alexander > Lopez > Sent: Monday, 10 January 2005 4:22 PM > To: karlp@fortephones.com; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server > > Make sure you has a span defined for each port on the TE410P. With out > signaling it would not take interrupts. > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Karl H. > Putz > Sent: Monday, January 10, 2005 12:38 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 > server > > I have been having this exact problem with a Tatung dual EMT-64 server > as > well. > > I have been trying to get a TE410P running and all looks great, driver > loads, runs ztcfg OK, etc. but no interrupts are ever processed. > > One additional piece of info that I have not seen in this thread is that > I > am able to successfully start and run a T100P card in this system. In > the > same PCI slot, wct1xxp driver built from the same CVS HEAD version as > the > wct4xxp. > > Just hoping this might shed some light on the problem for any Digium > folks > monitoring the forum. > > > Karl Putz > > > > _______________________________________________ > 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 > Date: Fri, 14 Jan 2005 01:37:14 -0500 > From: "John Bittner" <john@simlab.net> > Subject: [Asterisk-Users] Polycom Shared Call Appearance > To: <asterisk-users@lists.digium.com> > Message-ID: <200501140137156.SM00436@johnb2> > Content-Type: text/plain; charset="us-ascii" > > Has anyone got Polycom Shared Call Appearance working with > Asterisk ? > > If Asterisk doesn't support this, I am willing to put up a > bounty of 1000 to get it to work. > > John Bittner > Simlab.net > > > > Shared Call Appearance Signaling > A shared line is an address of record managed by a server. > The server allows multiple > endpoints to register locations against the address of > record. > SoundPointR IP supports shared call appearances (SCA) using > the SUBSCRIBENOTIFY > method in the "SIP Specific Event Notification" framework > (RFC 3265). > The events used are: > . "call-info" for call appearance state notification. > "line-seize for the phone to ask to seize the line > > > > ------------------------------ > > Message: 8 > Date: Fri, 14 Jan 2005 00:43:05 -0600 > From: "Julio Tejera" <jat@realityfirewall.net> > Subject: Re: [Asterisk-Users] SER vs Asterisk for SIP > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <015d01c4fa04$4d96d400$e101a8c0@Aceituno> > Content-Type: text/plain; charset="iso-8859-1" > > * is a "middleware" > > HTH > > ------- > Ing. Julio Alvarez Tejera > Unix Trends > *BSD, Solaris & Linux > VoIP & CT Solutions Finder > Asterisk PBX Consultant > Costa Rica Land +506-359-9753 > USA Toll Free +1-888-899-6269 > --------------- > "extremely stable systems" > > > ----- Original Message ----- > From: "Ashling O'Driscoll" <ashling.odriscoll@cit.ie> > To: <asterisk-users@lists.digium.com> > Sent: Thursday, January 13, 2005 10:57 AM > Subject: RE: [Asterisk-Users] SER vs Asterisk for SIP > > > > >From my (fairly limited) understanding, I think the fundamental > difference is that Asterisk is a pbx (offering all the features > associated with a pbx, voicemail, call transfer, call detail > recording etc) whereas SER is just a sip proxy (albeit a good one). > > Therefore Asterisk deals in terms of phones extensions whereas if you > want a system that can contact clients with sip urls, ser will have > to be set up. Also the audio i.e. rtp stream, traverses asterisk i.e. > it acts as a middle man holding onto the call, and if you want the > audio to go peer to peer (which it ideally should with sip), ser is > also needed. > > Aisling. > ---- Original Message ---- > From: vicky@freebsdcluster.net > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] SER vs Asterisk for SIP > Date: Thu, 13 Jan 2005 17:50:39 +0100 > > >Why is SER considered a better SIPserver than asterisk , why is it > >that SER > >can handle more clients than asterisk can. And if this is just cause > >of say > >poor SIP handling code in asterisk then is there anything being done > >to fix > >it. Just wanted to know why SER claims to be better than asterisk as > >a SIP > >server. ? > > > >-- > >regards > >Vikram (http://www.vicramresearch.com) > >_______________________________________________ > >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 > > > >-------------------Legal > >Disclaimer--------------------------------------- > > > >The above electronic mail transmission is confidential and intended > >only for the person to whom it is addressed. Its contents may be > >protected by legal and/or professional privilege. Should it be > >received by you in error please contact the sender at the above > >quoted email address. Any unauthorised form of reproduction of this > >message is strictly prohibited. The Institute does not guarantee the > >security of any information electronically transmitted and is not > >liable if the information contained in this communication is not a > >proper and complete record of the message as transmitted by the > >sender nor for any delay in its receipt. > > > > > > -------------------LegalDisclaimer---------------------------------------> > The above electronic mail transmission is confidential and intended onlyfor> the person to whom it is addressed. Its contents may be protected by legal > and/or professional privilege. Should it be received by you in errorplease> contact the sender at the above quoted email address. Any unauthorisedform> of reproduction of this message is strictly prohibited. The Institute does > not guarantee the security of any information electronically transmittedand> is not liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the sender nor > for any delay in its receipt. > > _______________________________________________ > 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: 9 > Date: Fri, 14 Jan 2005 00:51:50 -0600 > From: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] How to set asterisk NOT to answer > incoming lines? > To: C F <shmaltz@gmail.com>, Asterisk Users Mailing List - > Non-Commercial Discussion <asterisk-users@lists.digium.com> > Message-ID: <1105685510.13831.154.camel@critch> > Content-Type: text/plain > > On Thu, 2005-01-13 at 21:09 -0500, C F wrote: > > The definition of normal in the case of PBX implementations is up to > > the customer. > > You sure are acting like a 'tard lately. > > No a customer does not define normal, the market defines normal. A > customer defines an implementation. That implementation is either normal > or an exception/deviation of normal. > > > On Thu, 13 Jan 2005 10:44:51 -0600, Steven Critchfield > > <critch@basesys.com> wrote: > > > On Thu, 2005-01-13 at 16:08 +0000, Patrick Lidstone (Personal e-mail) > > > wrote: > > > > > > > I don't think Kelly's response is correct, at least for TDM FXO > boards. > > > > I could not find a way of preventing the FXO board grabbing the line > > > > when it rang, and subsequent enquiries on this list at the time > > > > suggested that it wasn't actually possible - which is a pity, as it > > > > means it is impossible to piggy back Asterisk on a POTS line with > other > > > > auto-answering equipment (e.g. data collection terminals). > > > > > > It isn't normal to put a PBX on a line shared with other equipment. It > > > is normal to route the other equipment through the PBX. > > > > > > -- > > > Steven Critchfield <critch@basesys.com> > > > > > > _______________________________________________ > > > 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 > -- > Steven Critchfield <critch@basesys.com> > > > > ------------------------------ > > Message: 10 > Date: Fri, 14 Jan 2005 18:00:26 +1100 > From: "Mike Sander" <mike@corporatebankinginternational.com> > Subject: [Asterisk-Users] Limit outgoing trunk calls > To: <asterisk-users@lists.digium.com> > Message-ID: <20050114070028.9F8D9EE9AB@mail.tyneinternational.com> > Content-Type: text/plain; charset="windows-1250" > > Skipped content of type multipart/alternative-------------- next part > -------------- > A non-text attachment was scrubbed... > Name: not available > Type: image/jpeg > Size: 3649 bytes > Desc: not available > Url : >http://lists.digium.com/pipermail/asterisk-users/attachments/20050114/b60b16> 94/attachment-0001.jpeg > > ------------------------------ > > Message: 11 > Date: Fri, 14 Jan 2005 08:26:10 +0100 > From: "Florian Overkamp" <florian@obsimref.com> > Subject: RE: [Asterisk-Users] Agentcallbackogin > withoutanyuserinputafter extension is dialed. > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <E1CpLr1-0004Ao-00@clio> > Content-Type: text/plain; charset="us-ascii" > > Hi, > > > -----Original Message----- > > Ok, maybe this is an ignorant question but...... where in memory does > > asterisk store the information and how do I access it? > > It's not an ignorant question, but it is like I've stated a few times now: > The agent information asterisk has is in its own memory and cannot be > accessed easily (you could probably write an AGI script that executes'show> agents' and parses the output though). That is exactly why you make your > dialplan so every time an agent logs on or off you store your own copy if > the info in the asterisk database where it is available to you for future > reference. > > BTW, I know agent technology is a bit better in CVS-HEAD but for my > customers sake (where I have to run a stable branch) I kicked out usage of > agents and now emulate it all with a few AGI scripts. > > Florian > > > > > ------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 6, Issue 199 > ********************************************** > > > _______________________________________________ > 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