Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

[Peter Brown]

At 11:20 11/01/04 +0800, you wrote:
Anton Tinchev wrote:

Just spended ~ hour googling - all boards are based on GC-XX or I750X 
Chipsets - all for Xeons. There also some boards for Pentium 3.
Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 
800Mhz FSB.
Thanks
Unless one has appeared in the last couple of weeks, there are none. In 
fact, the only one I know of for any kind of non-Xeon Pentium is the Dell 
600SC. That one isn't an 800MHz bus machine.

Regards,
Steve
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Re: [Asterisk-Users] Asterisk Development Updates

[Greg Boehnlein]

On Sat, 10 Jan 2004, Jared Smith wrote:

> On Sat, 2004-01-10 at 20:25, Greg Boehnlein wrote:
> 
> > Awesome! I'm game to create Asterisk RPMS when the stable branch comes 
> > out!
> 
> Great... I was going to do the same... maybe we should join forces and
> make better RPMS!  (I've already got a semi-decent .spec file done.)
> 
> Jared Smith

Sounds like a plan. I run a fairly large, well connect Data center, so I 
can provide a development environment, and centralized download 
repository!

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Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[Chris Albertson]

Anyone who does not like the fact that all code must be
"disclaimed", sent through Digum to CVS and that GPL'd
code can't go in can fix that problem.  All you need to
do is copy the current CVS and use that to start your
own project.  You can call it "Asterisk Prime" or "Star"
and make up your own rules within the limits of the GPL.

So there, if you want the rules changed _you_ can change
them.  But I'll bet you don't want the rules changed so badly
that you would go to that much trouble.  I and I assume
most everyone here whould rather put up the way it is than
duplicate Mark's efforts

--- Steven Critchfield <[EMAIL PROTECTED]> wrote:
> On Sat, 2004-01-10 at 18:47, [EMAIL PROTECTED] wrote:
> 
> > I have always been suspicious of centralized control and
> dictatorship,
> > benevolent or otherwise. After thinking for some time about the 
> > licensing structure of code for Asterisk, I am not sure that
> > their motives are so innocuous and altrusitic, or at least
> > this is not reflected so well in the fine print. After learning
> > that "all code must pass through Mark", I am even less sure.
> > It means that Digium remains in a position of control and 
> > dominance over what is ostensibly communal property.
> 
> I seem to remember at one point that all code in the official linux
> kernel had to go though Linus. Did we suffer? I don't think so. All
> code
> going through Mark isn't a bad thing. If you look through the cvs
> logs,
> you might see there are 3 or so commiters right now. I know jeremy is
> able to commit, but I think he is limited(probably self imposed) to
> the

=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

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RE : [Asterisk-Users] 2nd call leg status?

[SW]

Hi Folks,

Wonder whether this question found an answer ?

I too have a similar question that I can't find an answer so far.

Let me first share my dial plan;

exten => _011.,1,Authenticate(/etc/asterisk/auth.txt |a)
exten => _011.,2,Playback(Pls-wait-while-I-connect)
exten => _011.,3,Absolutetimeout(3600)
exten => _011.,4,Dial(H323/${EXTEN:[EMAIL PROTECTED],70)
exten => _011.,5,NoCDR()  <=if no answer cdr is not written
exten => _011.,6,Busy
exten => _011.,105,NoCDR()  < if called party is busy cdr not wirtten
exten => _011.,106,Busy
exten => i,1,NoCDR() <== if authentication failed cdr not written
exten => i,2,Hangup

Here are my observations

(a) Since Authenticate function is present in my dial paln,  disposition
fieled in cdr always show Answered, so with that I can't figure out whether
H323 leg is successfully answered or not.
(b) If the H323 g/w sends the busy signal then CDR is not written, If the
g/w rings and timed out then again CDR is not written (as expected we have
priorities set for extensions)
(c) Now if the called party is ringing and originating party just hang-up, A
CDR is written. I have no way to differentiate that with a very short answer
call.

I think this behavior is a incorrect.

If * answered a call and show up disposition as Answered, then that call is
a completed call. So there should be one record in the CDR.

Then for the second leg there should be another record, as now * is
originating a call again and expecting other side(in this case the h323 g/w)
to respond.

If the both legs are considered as a single call, then cdr should show the
disposition of the final end point.

Please show me if there is a way that I could generate two records in CDR
for this kind of a call, or any other solution for this problem.

Cheers

SW



Message: 2
Date: Thu, 8 Jan 2004 17:27:07 + (GMT)
From: "Stephen J. Wilcox" <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] 2nd call leg status?
Reply-To: [EMAIL PROTECTED]

Hi,
 okay heres what I want to do .. simple ivr, we take a call, answer it, play
a
menu, dial out based on options. No problems so far.

The CDR always shows the call as answered as I answer the 1st leg to play
the
prompts, I am actually more interested in if the 2nd leg - the outbound
part -
has been answered or not before the call is hungup. How can I get this and
record the information in the CDR?

Steve


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[Asterisk-Users] a constructive proposal: tie the marshals to a cvs server

[asterisk]

perhaps i have been too ascerbic on the list as of late. 
for that i offer my apologies.

create a second cvs server, bug marshalls have commit access.
they commit patches (tagged with bug id) to this server.
users wanting bleeding-edge stuff, use that server and don't
have to keep downloading and patching from the web site.
digium at their discretion, imports changes into canonical source.

i will happily make such a server available.

-w
-- 
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\ /No HTML/RTF in email
 X No Word docs in email
/ \  Respect for open standards
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Re: [Asterisk-Users] default music source for SIP channel

[Olle E. Johansson]

Lance Arbuckle wrote:

Thanks Kevin, but boy, do I feel dumb.  Maybe someone could update the
MusicOnHold wiki page and add SetMusicOnHold to the "Also See" section.
Added to wiki and submitted a patch with new "show application musiconhold" help text
to reflect this.
/O

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Re: [Asterisk-Users] default music source for SIP channel

[Olle E. Johansson]

Lance Arbuckle wrote:


So, why does zapata.conf accept musiconhold=class yet sip.conf ignores a
similar statement ?  Can anyone give me an example of how to control the
MOH class for a SIP channel ?
Because no one added that feature to chan_sip.c

For now, use setmusiconhold, then you're welcome to submit a bug report to
bugs.digium.com. If you're a programmer, give it a try and create a patch.
If you aren't familiar with programming, maybe someone else will find
the inspiration to fix this. At least, it will be filed in the repository.
Thank you for your help.

/O

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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[info-lists]

Chandra said:
> i also had the same problem temporarily i solved my problem with both
> outside NAT. u can also do it if both inside NAT. * outside NAT and
> Budgetone behind NAT simply doesn't seem to work. if u ever solve this
> problem please let me know too.
>
> thanks
>
> cm
>

I am able to use my Grandstream very well from behind NAT going to FWD. It
seems that a proxy server is needed outside NAT in order for SIP to work
correctly.  Havn't heard of any SIP phones that can jump through NAT
without a problem IAX, on the other hand, seems to work fine.

Robert
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Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

[Balaji NJL]

Hi All,

i just applied this patch. i need to test whether its
working. Can someone
connect to my server and leave me a vm at extension
2000.

Server : ojoobala.com

Phone
Extension : 2005
pwd   : mytest
auth: md5.

pl leave a vm on extension 2000.

thanks a lot,
-B
- Original Message - 
From: "listas iPfone" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 2:10 AM
Subject: Re: [Asterisk-Users] Asterisk behind NAT <<
How to do it.


> Hi
>
> The version 1.260 of chan_sip.c already have that
patch?:
>
>
http://bugs.digium.com/file_download.php?file_id=430&type=bug
>
> thanks!
>
> Miklos
>
>
> - Original Message - 
> From: "Leif Madsen" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, November 28, 2003 2:10 AM
> Subject: [Asterisk-Users] Asterisk behind NAT << How
to do it.
>
>
> > Thanks to ww and his patch on bug #104, I have
successfully implemented
> > Asterisk behind NAT without using STUN or anything
crazy.  It's quite
> > straight forward.
> >
> > Until this gets tested enough and put into CVS,
you will have to patch
> > your chan_sip.c file to do this.  I'm sure within
the next few days this
> > will get put merged into CVS if no one finds any
problems.
> >
> > I tried this on chan_sip.c version 1.249 (the
version the patch was
> > written for) and the latest as of today 1.258. 
Both work great.
> >
> > Open ports 5060 and your RTP range (found in
/etc/asterisk/rtp.conf).
> > Default is 1 -> 2
> >
> > Forward ports 5060 and your RTP range to your
internal Asterisk box.
> >
> > For your sip.conf, you need to add three lines:
> >
> > ; sip.conf snippet
> > [general]
> > port=5060   ; make sure you
have this line :)
> > inside_net=192.168.1.100; this is the
internal ip address of
> > the;
> > asterisk server
> > inside_mask=255.255.255.0   ; internal ip
mask.  /24 as this example
> > outside_addr=216.239.33.100 ; this can also be
a FQDN! ie.
> > ; my.domain.com
> > ; ... plus whatever else you have in your sip.conf
> >
> > Download the patch at:
> >
http://bugs.digium.com/file_download.php?file_id=430&type=bug
> >
> > Either update your Asterisk or verify you have at
least version 1.249 of
> > chan_sip.c:
> >
> > cd /usr/src/asterisk/channels/
> > cvs status chan_sip.c
> >
> >
===
> > File: chan_sip.cStatus: Locally Modified
> >
> >Working revision:1.258
> >Repository revision: 1.258
> > /usr/cvsroot/asterisk/channels/chan_sip.c,v
> >
> > While in pwd /usr/src/asterisk/channels/
> > patch -p0 < /path/to/patch
> >
> > Nothing should fail.
> >
> > cd /usr/src/asterisk/
> > make
> > cp /usr/src/asterisk/channels/chan_sip.so
/usr/lib/asterisk/modules/
> >
> > Restart your Asterisk and try it.  If you want to
call a NAT'd Asterisk
> > box, my Free World Dialup number is 18924. 
Currently online.
> >
> > -- 
> > Leif Madsen <[EMAIL PROTECTED]>
> > http://www.hacklocalhost.com
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
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> [EMAIL PROTECTED]
>
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Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

[Anton Tinchev]

Peter Brown wrote:

At 11:20 11/01/04 +0800, you wrote:

Anton Tinchev wrote:

Just spended ~ hour googling - all boards are based on GC-XX or 
I750X Chipsets - all for Xeons. There also some boards for Pentium 3.
Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 
with 800Mhz FSB.
Thanks


Unless one has appeared in the last couple of weeks, there are none. 
In fact, the only one I know of for any kind of non-Xeon Pentium is 
the Dell 600SC. That one isn't an 800MHz bus machine.

Regards,
Steve
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Asus NRL-LS533 Server Board
Single Intel Pentium 4 processors up to 2.8GHz+ North Bridge: 
ServerWorks CMIC-SL South Bridge: ServerWorks CSB6 Up to 4GB 
registered PC1600 ECC DDR RAM, 4 DIMM slots Single channel Ultra160 
SCSI 1 x BroadCom 5702C 32bit PCI Gigabit Ethernet controller for 
NRL-LS533 5 x 64bit/33MHz/3.3V PCI slots ASUS Server Management 
Software included TrendMicroTM ServerProtect anti-virus software full 
users version for enterprises FROM TWINCUBE.COM

Peter Brown

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Yes, but they all are with 533 Mhz FSB
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Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

[Balaji NJL]

i like the idea of not requiring to open 1 ports
in the firewall.

Do i need to change rtf.conf to from 1 - 2 to
16384 and 16394.

thanks,
-B 
- Original Message - 
From: "Craig Waddington" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, December 27, 2003 3:43 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT <<
How to do it.


> Hi
> 
> I have SIP working on NAT using X-lite phones. 
> 
> On my Cisco 827H ADSL router I forwarded ports 5060,
16394, 16384 to my
> * (10.1.0.0).
> 
> 16394,16384 being RTP.
> 
> In X-lite set the RTP port to use 16394 instead of
the default 8000.
> 
> Works great over the internet. Didn't need patches
or anything else.
> 
> I hope that helps you.
> 
> -C
> 
> 
> www.ntfs.org
> 
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
Behalf Of Balaji NJL
> Sent: 27 December 2003 08:34
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk behind NAT <<
How to do it.
> 
> Hi All,
> 
> i tried to apply this patch and i got the following
> error. The chan_sip.c
> version i hv is 1.265
> 
> hv any one tried this patch on this latest chan_sip
> version.
> 
> thanks,
> -B
> 
> chan_sip.o: In function `load_module':
> chan_sip.o(.text+0x15ebf): undefined reference to
> `ast_rtp_proto_register'
> chan_sip.o(.text+0x15ee0): undefined reference to
> `ast_register_application'
> chan_sip.o: In function `delete_users':
> chan_sip.o(.text+0x15fc1): undefined reference to
> `ast_free_ha'
> chan_sip.o(.text+0x1604d): undefined reference to
> `ast_sched_del'
> chan_sip.o: In function `prune_peers':
> chan_sip.o(.text+0x16167): undefined reference to
> `ast_sched_del'
> chan_sip.o(.text+0x1618d): undefined reference to
> `ast_sched_del'
> chan_sip.o: In function `unload_module':
> chan_sip.o(.text+0x162bd): undefined reference to
> `ast_channel_unregister'
> chan_sip.o(.text+0x162ce): undefined reference to
> `ast_unregister_application'
> chan_sip.o(.text+0x16337): undefined reference to
> `ast_softhangup'
> chan_sip.o(.text+0x1636c): undefined reference to
> `ast_log'
> chan_sip.o(.text+0x163ab): undefined reference to
> `pthread_cancel'
> chan_sip.o(.text+0x163be): undefined reference to
> `pthread_kill'
> chan_sip.o(.text+0x163d1): undefined reference to
> `pthread_join'
> chan_sip.o(.text+0x16418): undefined reference to
> `ast_log'
> chan_sip.o(.text+0x164b8): undefined reference to
> `ast_log'
> collect2: ld returned 1 exit status
> make: *** [chan_sip.so] Error 1
> 
> - Original Message - 
> From: "listas iPfone" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, December 09, 2003 2:10 AM
> Subject: Re: [Asterisk-Users] Asterisk behind NAT <<
> How to do it.
> 
> 
> > Hi
> >
> > The version 1.260 of chan_sip.c already have that
> patch?:
> >
> >
>
http://bugs.digium.com/file_download.php?file_id=430&type=bug
> >
> > thanks!
> >
> > Miklos
> >
> >
> > - Original Message - 
> > From: "Leif Madsen" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Friday, November 28, 2003 2:10 AM
> > Subject: [Asterisk-Users] Asterisk behind NAT <<
How
> to do it.
> >
> >
> > > Thanks to ww and his patch on bug #104, I have
> successfully implemented
> > > Asterisk behind NAT without using STUN or
anything
> crazy.  It's quite
> > > straight forward.
> > >
> > > Until this gets tested enough and put into CVS,
> you will have to patch
> > > your chan_sip.c file to do this.  I'm sure
within
> the next few days this
> > > will get put merged into CVS if no one finds any
> problems.
> > >
> > > I tried this on chan_sip.c version 1.249 (the
> version the patch was
> > > written for) and the latest as of today 1.258. 
> Both work great.
> > >
> > > Open ports 5060 and your RTP range (found in
> /etc/asterisk/rtp.conf).
> > > Default is 1 -> 2
> > >
> > > Forward ports 5060 and your RTP range to your
> internal Asterisk box.
> > >
> > > For your sip.conf, you need to add three lines:
> > >
> > > ; sip.conf snippet
> > > [general]
> > > port=5060   ; make sure you
> have this line :)
> > > inside_net=192.168.1.100; this is the
> internal ip address of
> > > the;
> > > asterisk server
> > > inside_mask=255.255.255.0   ; internal ip
> mask.  /24 as this example
> > > outside_addr=216.239.33.100 ; this can also
be
> a FQDN! ie.
> > > ; my.domain.com
> > > ; ... plus whatever else you have in your
sip.conf
> > >
> > > Download the patch at:
> > >
>
http://bugs.digium.com/file_download.php?file_id=430&type=bug
> > >
> > > Either update your Asterisk or verify you have
at
> least version 1.249 of
> > > chan_sip.c:
> > >
> > > cd /usr/src/asterisk/channels/
> > > cvs status chan_sip.c
> > >
> > >
>
===
> > > File: chan_sip.cStatus: Locally Modified
> > >
> > >Working revision:1.258
> > >Repository revision: 1.258

[Asterisk-Users] High Level of CVS activity

[Dave Cotton]

I don't think I've ever seen so many commits to the CVS, I'm not
complaining. To save lots of checkout/test cycles can a message be
posted when the current surge is over?


-- 
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Directeur
Linux Autrement
193 rue Marcel Cerdan
84270 Vedene
04 90 23 30 81

IAX 17004902330 FWD 42651

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RE: [Asterisk-Users] Asterisk behind NAT << How to do it.

[Craig Waddington]

Balaji.

I just left rtf.conf at default. Though I guess it wouldn't hurt to
change it to test.

Does it currently work for you with the settings I provided?

Craig.


www.ntfs.org


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL
Sent: 11 January 2004 10:35
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

i like the idea of not requiring to open 1 ports
in the firewall.

Do i need to change rtf.conf to from 1 - 2 to
16384 and 16394.

thanks,
-B 
- Original Message - 
From: "Craig Waddington" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, December 27, 2003 3:43 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT <<
How to do it.


> Hi
> 
> I have SIP working on NAT using X-lite phones. 
> 
> On my Cisco 827H ADSL router I forwarded ports 5060,
16394, 16384 to my
> * (10.1.0.0).
> 
> 16394,16384 being RTP.
> 
> In X-lite set the RTP port to use 16394 instead of
the default 8000.
> 
> Works great over the internet. Didn't need patches
or anything else.
> 
> I hope that helps you.
> 
> -C
> 
> 
> www.ntfs.org
> 
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
Behalf Of Balaji NJL
> Sent: 27 December 2003 08:34
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk behind NAT <<
How to do it.
> 
> Hi All,
> 
> i tried to apply this patch and i got the following
> error. The chan_sip.c
> version i hv is 1.265
> 
> hv any one tried this patch on this latest chan_sip
> version.
> 
> thanks,
> -B
> 
> chan_sip.o: In function `load_module':
> chan_sip.o(.text+0x15ebf): undefined reference to
> `ast_rtp_proto_register'
> chan_sip.o(.text+0x15ee0): undefined reference to
> `ast_register_application'
> chan_sip.o: In function `delete_users':
> chan_sip.o(.text+0x15fc1): undefined reference to
> `ast_free_ha'
> chan_sip.o(.text+0x1604d): undefined reference to
> `ast_sched_del'
> chan_sip.o: In function `prune_peers':
> chan_sip.o(.text+0x16167): undefined reference to
> `ast_sched_del'
> chan_sip.o(.text+0x1618d): undefined reference to
> `ast_sched_del'
> chan_sip.o: In function `unload_module':
> chan_sip.o(.text+0x162bd): undefined reference to
> `ast_channel_unregister'
> chan_sip.o(.text+0x162ce): undefined reference to
> `ast_unregister_application'
> chan_sip.o(.text+0x16337): undefined reference to
> `ast_softhangup'
> chan_sip.o(.text+0x1636c): undefined reference to
> `ast_log'
> chan_sip.o(.text+0x163ab): undefined reference to
> `pthread_cancel'
> chan_sip.o(.text+0x163be): undefined reference to
> `pthread_kill'
> chan_sip.o(.text+0x163d1): undefined reference to
> `pthread_join'
> chan_sip.o(.text+0x16418): undefined reference to
> `ast_log'
> chan_sip.o(.text+0x164b8): undefined reference to
> `ast_log'
> collect2: ld returned 1 exit status
> make: *** [chan_sip.so] Error 1
> 
> - Original Message - 
> From: "listas iPfone" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, December 09, 2003 2:10 AM
> Subject: Re: [Asterisk-Users] Asterisk behind NAT <<
> How to do it.
> 
> 
> > Hi
> >
> > The version 1.260 of chan_sip.c already have that
> patch?:
> >
> >
>
http://bugs.digium.com/file_download.php?file_id=430&type=bug
> >
> > thanks!
> >
> > Miklos
> >
> >
> > - Original Message - 
> > From: "Leif Madsen" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Friday, November 28, 2003 2:10 AM
> > Subject: [Asterisk-Users] Asterisk behind NAT <<
How
> to do it.
> >
> >
> > > Thanks to ww and his patch on bug #104, I have
> successfully implemented
> > > Asterisk behind NAT without using STUN or
anything
> crazy.  It's quite
> > > straight forward.
> > >
> > > Until this gets tested enough and put into CVS,
> you will have to patch
> > > your chan_sip.c file to do this.  I'm sure
within
> the next few days this
> > > will get put merged into CVS if no one finds any
> problems.
> > >
> > > I tried this on chan_sip.c version 1.249 (the
> version the patch was
> > > written for) and the latest as of today 1.258. 
> Both work great.
> > >
> > > Open ports 5060 and your RTP range (found in
> /etc/asterisk/rtp.conf).
> > > Default is 1 -> 2
> > >
> > > Forward ports 5060 and your RTP range to your
> internal Asterisk box.
> > >
> > > For your sip.conf, you need to add three lines:
> > >
> > > ; sip.conf snippet
> > > [general]
> > > port=5060   ; make sure you
> have this line :)
> > > inside_net=192.168.1.100; this is the
> internal ip address of
> > > the;
> > > asterisk server
> > > inside_mask=255.255.255.0   ; internal ip
> mask.  /24 as this example
> > > outside_addr=216.239.33.100 ; this can also
be
> a FQDN! ie.
> > > ; my.domain.com
> > > ; ... plus whatever else you have in your
sip.conf
> > >
> > > Download the patch at:
> > >
>
http://bugs.digium.com/file_download.php?file_id=4

Re: [Asterisk-Users] default music source for SIP channel

[Lance Arbuckle]

"Olle E. Johansson" wrote:
> 
> Lance Arbuckle wrote:
> >
> >
> > So, why does zapata.conf accept musiconhold=class yet sip.conf ignores a
> > similar statement ?  Can anyone give me an example of how to control the
> > MOH class for a SIP channel ?
> Because no one added that feature to chan_sip.c

ah, I thought I was going nuts, yet again :)

> 
> For now, use setmusiconhold, then you're welcome to submit a bug report to
> bugs.digium.com. If you're a programmer, give it a try and create a patch.
> If you aren't familiar with programming, maybe someone else will find
> the inspiration to fix this. At least, it will be filed in the repository.
> 
> Thank you for your help.
> 
> /O

I'd be happy to work on this but, alas, I haven't a programming clue.  I
don't think this is a big deal bug wise, but it sure would be nice from
a consistancy standpoint :)
Thanks for the reply...

-Lance
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Re: [Asterisk-Users] far end disconnect supervision

[Rich Adamson]

> > Personal opinion is that throwing a channel bank at * (other then for
> > better echo cancellation on trunks) is like taking a shower with your
> > socks on; don't see any practical use. But, I'm sure some do.
>  
> This statement assumes a lot. 
> 
> 1. That you don't want to use analog phones.
> 2. That you need enough lines for a T1/PRI circuit. In this case the most
> economical way to deploy a setup of say 5-16 voice lines is with a channel bank
> and a T1 card in the * server. If there is some less expensive option please
> share :-). 

Note the comment in the statement that says "other then ... trunks".
Full agreement that most channel banks will handle echo cancellation,
impedence matching, etc, very well. They've been doing it for years.

The comment was oriented towards using a channel bank to connect analog
phones to *, and was not intended to start another thread, flames, etc.
And, we've all seen the discussion relative to the cost of sip phones
vs analog many times over, use of left over slots for phones, etc, etc.
If you have a specific case where that makes sense for you or your 
customer, knock yourself out (that would be the "some do" part). 
No need to rehash the "I'm cheaper then you are" discussions again.



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[Asterisk-Users] More Success on the Cisco 7920 and SCCP !!!!!

[Jan Czmok]

Hi All,

have some decent success on the 7920 "activation" in Asterisk.

Latest status:
chan_skinny does NOT work with 7920
chan_sccp does WORK with 7920 (!!)

however:

to remove coredumping the chan_sccp just comment out the MWI
(messagewaitingindicator), then it compiles fine.

Then change sccp_helper.c:  return "P0060302" instead of the old value.

and voila:

Phone is registering to Asterisk :-)

But currently:
--
After registering to Asterisk it received a "off-hook" message from the
7960 and then "Call Ended" on the Display (curious about that !!!).

After that the phone reboots and the stuff repeats

Hope to find more answers soon, but it should lead people in the right
direction.

--jan


-- 
Jan Czmok, Network Engineering & Support, Global Access Telecomm, Inc.
Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED]
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[Asterisk-Users] Re: [Asterisk-Dev] More Success on the Cisco 7920 and SCCP !!!!!

[Siggi Langauf]

Hi Jan,

the 7920 is on my todo list for quite a few days now, and I've had
experience similar to yours...

On Sun, 11 Jan 2004, Jan Czmok wrote:

> Latest status:
> chan_skinny does NOT work with 7920
> chan_sccp does WORK with 7920 (!!)

Yup.
One should add that you'd better use the 0.2 release or a CVS snapshot.
(Note for CVS: Zozo's server is a bit picky about its clients. You have to
 add the trailing "/" to your CVSROOT, exactly as documented. And even
 then, I was only able to make one checkout, whereas "cvs update" in that
 tree always failed. Some versions of CVS clients didn't work at all.
 One working version is the 1:1.12.2-2 Debian package)

> however:
>
> to remove coredumping the chan_sccp just comment out the MWI
> (messagewaitingindicator), then it compiles fine.

Haven't had any trouble with that. (In fact, MWI works fine on a 7960 with
chan_sccp here.) When do you get core dumps and do you have app_voicemail
loaded/configured?

> Then change sccp_helper.c:  return "P0060302" instead of the old value.
>
> and voila:
>
> Phone is registering to Asterisk :-)
>
> But currently:
> --
> After registering to Asterisk it received a "off-hook" message from the
> 7960 and then "Call Ended" on the Display (curious about that !!!).

That seems to be normal for the 7920. I've sniffed the registration
procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's
doing the same thing. Maybe that's some odd way of testing if the
CallManager ("CCM") really works...

> After that the phone reboots and the stuff repeats

Same thing here.
CCM does quite a few things in different order compared to chan_sccp, but
apart from that, the registration procedure seems quite similar.
I'm still looking into the detailed differences (which is a bit hard, as
there doesn't seem to be any tool like "diff" for ethereal traces).

> Hope to find more answers soon, but it should lead people in the right
> direction.

I hope so, too. And I'll let you know if there is any news.

That said: The 7920 is definitely beta quality, at most: I have
experienced dropped connections as well as the phone ignoring remote
hangup, on more that 50% of my calls in a 100% Cisco environment, with a
Cisco AiroNet access point within 3m, direct line-of-sight. So I guess it
will take a year or two (just as with the 7960/7940) before Cisco gets the
firmware in a state that can be considered production quality.

Cheers,
Siggi
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Re: [Asterisk-Users] High Level of CVS activity

[Greg Boehnlein]

On Sun, 11 Jan 2004, Dave Cotton wrote:

> I don't think I've ever seen so many commits to the CVS, I'm not
> complaining. To save lots of checkout/test cycles can a message be
> posted when the current surge is over?

Most likely that is the frenzy before the release of 0.7.0 on Monday, the 
12th! ;)

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Development Process comment and Email list suggestion

[Siggi Langauf]

On Fri, 9 Jan 2004 [EMAIL PROTECTED] wrote:

[...]
> Regarding the email list: A number of people have suggested creating more
> email lists. I think this is not a good idea because there will be even
> more cross posting than there is now between -dev and -users.

That's a very valid point.

[...]
> various modules of R/3 (ie: topic areas) were managed by using (and
> inforcing) topic keywords in the subject line.  Enforcing the list
> guidelines was a 2 hour or so task each night for the moderators (we had
> several that rotated weekly shifts) but it made the list usage fairly
> easy.  Subscribers could pick the topics of interest and then (assuming
> people followed the guidelines) their messages from the list were
> restricted to those topics.  Dealing with chronic violators was always a
> judgement call by the moderator-on-duty: if a person was INTENTIONALLY not
> following guidelines then they were blocked. If they didn't understand the
> guideline then it was explained again to them and life went on.

Generally, I like this idea. Note however, that there doesn't seem to be
any dedicated moderator workforce for this list at all. (At least I've
never seen any of the posts ever approved or rejected by a moderator that
I've posted from the wrong address...)
So for educating people we'd need some voluntary moderators, first. (No,
I'm not volunteering.)

[...]
> There would definately be a cost for the ListServe license (since there is
> a commercial profit from the list)  but I think this software is best able
> to handle what we need to do.  If Mark and Digium want to go in this
> direction I would be glad to coordinate the moderation.

*hehe* Caught you here.
Luckily, it doesn't take ListServe or any license. From the NEWS file[1]
of MailMan 2.1:

- Topic Filters
o A new feature has been added called "Topic Filters".  A list
  administrator can create topics, which are essentially
  regular expression matches against Subject: and Keyword:
  headers (including such pseudo-headers if they appear in the
  first few lines of the body of a message).

  List members can then `subscribe' to various topics, which
  allows them to filter out any messages that don't match a
  topic, or to filter out any message that does match a
  topic.  This can be useful for high volume lists where not
  everyone will be interested in every message.

As that's exactly the feature you described, a simple upgrade of Digium's
MailMan would cut it.
As you're volunteering to organize the moderation, that should be a
realistic plan, especially if you consider that the upgrade brings quite
some bug fixes and improved performance due to better MTA integration...

Cheers,
Siggi

[1]
http://cvs.sourceforge.net/viewcvs.py/mailman/mailman/NEWS?rev=2.43&view=markup
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Re: [Asterisk-Users] far end disconnect supervision

[Philipp von Klitzing]

Hi!

> I've not implemented any form of channel bank with *, so can't offer much
> help on specific vendor/models. Since there are a fair number of folks
> using them on the list, try posting a new thread with channel bank in
> the subject, and summarize the responses in the wiki.

Yes, please do so at:
http://www.voip-info.org/
tiki-index.php?page=Asterisk+hardware+channel+bank+check

Cheers, Philipp


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Re: [Asterisk-Users] default music source for SIP channel

[Philipp von Klitzing]

Hi!

> Thanks Kevin, but boy, do I feel dumb.  Maybe someone could update the
> MusicOnHold wiki page and add SetMusicOnHold to the "Also See"
> section. 

Just login to the Wiki and do that yourself. That's the idea of a Wiki!
Greez, Philipp



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[Asterisk-Users] New Version of SJPhone

[admin]

I just installed the new version of SJPhone and it 
appears that it cannot work with * anymore?

RE : [Asterisk-Users] 2nd call leg status?

[Freddi Hansen]

Hi Folks,

Wonder whether this question found an answer ?

I too have a similar question that I can't find an answer so far.

Let me first share my dial plan;

exten => _011.,1,Authenticate(/etc/asterisk/auth.txt |a)
exten => _011.,2,Playback(Pls-wait-while-I-connect)
exten => _011.,3,Absolutetimeout(3600)
exten => _011.,4,Dial(H323/${EXTEN:[EMAIL PROTECTED],70)
exten => _011.,5,NoCDR()  <=if no answer cdr is not written
exten => _011.,6,Busy
exten => _011.,105,NoCDR()  < if called party is busy cdr not wirtten
exten => _011.,106,Busy
exten => i,1,NoCDR() <== if authentication failed cdr not written
exten => i,2,Hangup
Here are my observations

(a) Since Authenticate function is present in my dial paln,  disposition
fieled in cdr always show Answered, so with that I can't figure out 
whether
H323 leg is successfully answered or not.
(b) If the H323 g/w sends the busy signal then CDR is not written, If the
g/w rings and timed out then again CDR is not written (as expected we 
have
priorities set for extensions)
(c) Now if the called party is ringing and originating party just 
hang-up, A
CDR is written. I have no way to differentiate that with a very short 
answer
call.

I think this behavior is a incorrect.

If * answered a call and show up disposition as Answered, then that 
call is
a completed call. So there should be one record in the CDR.

Then for the second leg there should be another record, as now * is
originating a call again and expecting other side(in this case the 
h323 g/w)
to respond.

If the both legs are considered as a single call, then cdr should show 
the
disposition of the final end point.

Please show me if there is a way that I could generate two records in CDR
for this kind of a call, or any other solution for this problem.
Cheers

SW


Hi,
I did have the same problem. I simply issue a ResetCDR(w) as the last 
thing before using
the Dial application. This will reset the 'answered' flag and the CDR 
you get from the Dial
will contain the correct value. I do later do a small of backend 
processing to get the correct
A-line time related to the call.
There may be smarter way of doing this but this quick hack works fine 
for me
Freddi





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[Asterisk-Users] Cisco 79xx Ringtones

[Adthrawn]

Hi,

I'm after two very specific ringtones for the 79xx's...

A dog barking, and a horse either galloping or neighing.

I've tried making the sounds, but for some bizarre reason they're not 
working. I used to make quite a few ringtones for the 79xx's, but I 
seem to have forgotten how to do it! And to top things off, I can't 
even find the documentation on Cisco's site for making new ringtones.

I do recall, you had to set the sample length to a divisible, something 
like 800? And there was a maximum sample length too...

Best,
Ad.
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[Asterisk-Users] Strange problem with call hangup on Budgetone 102 Phones

[Jon Fautley]

Hi,
 
I've got Asterisk configured and working (sort of) 
with an Eicon Diva Server 2M ISDN card (connected to S0 bus of another PBX). 
This * box is on a 'live', non-nat IP address.
I also have a couple of budgetone phones, one 
behind NAT and one not. When I place an outgoing call, I get the following 
messages:
 
-- Executing Dial("SIP/filbert-9876", 
"CAPI/288:333") in new stack    -- creating pipe for 
PLCI=-1   > sent CONNECT_REQ MN 
=0x5    -- Called 288:333    -- Setting up 
echo canceller (PLCI=0x201, function=1, options=2, 
tail=64)   > sent FACILITY_REQ 
(PLCI=0x201)    -- CAPI[contr1/288]/0 answered 
SIP/filbert-9876    -- Echo canceller successfully set up 
(PLCI=0x201)WARNING[9226]: File chan_sip.c, Line 464 (retrans_pkt): Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request)    -- CAPI 
Hangingup   > sent DISCONNECT_B3_REQ 
NCCI=0xa0201   > sent DISCONNECT_REQ 
PLCI=0x201    -- removed pipe for PLCI = 0x201  == 
Spawn extension (sip, 9333, 1) exited non-zero on 
'SIP/filbert-9876'WARNING[9226]: File chan_sip.c, Line 464 (retrans_pkt): 
Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 103 (Request)
I can hear the voicemail service (extn. 333) answer 
correctly, but then after about 5 seconds i'll get the WARNING message and the 
system will hangup.
 
Here's a snippet from my sip.conf 
file:
 
---
[general]port = 5060bindaddr = 
0.0.0.0
context = 
sip-incomingsrvlookup=noqualify=yesdisallow=allallow=alawallow=ulaw
 
[filbert]type=friendhost=dynamicdtmfmode=infocontext=sipcallerid="Jon 
Fautley" 
<200>nat=yespickupgroup=1reinvite=nocanreinvite=nodisallow=allallow=ulaw
 
 
Any ideas?
 
Many thanks,
 
Jon

[Asterisk-Users] macro error "exited non-zero"

[Jimmy Riley]

  On this macro I keep getting exited non-zero on s,3, but s,3 is doing what
it is suppose to do but the macro stops. Is there a way to make a macro
ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4.

Also if I just run this line from the command line I don't get an error.
[EMAIL PROTECTED] monitor]# sox in.wav in-rev.wav reverse
[EMAIL PROTECTED] monitor]#


[macro-record-cleanup]

exten => s,1,GotoIf($[${CALLFILENAME} = ${FOO}]?11:2)
exten => s,2,SetVar(MONITORDIR=/var/spool/asterisk/monitor)

exten => s,3,System(/usr/local/bin/sox ${MONITORDIR}/${CALLFILENAME}-in.wav
${MONITORDIR}/${CALLFILENAME}-in-rev.wav reverse)
exten => s,4,System(/usr/local/bin/sox ${MONITORDIR}/${CALLFILENAME}-out.wav
${MONITORDIR}/${CALLFILENAME}-out-rev.wav reverse)

exten => s,5,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav
${MONITORDIR}/${CALLFILENAME}-out.wav)

exten => s,6,System(/usr/local/bin/soxmix
${MONITORDIR}/${CALLFILENAME}-in-rev.wav
${MONITORDIR}/${CALLFILENAME}-out-rev.wav
${MONITORDIR}/${CALLFILENAME}-rev.gsm)

exten => s,7,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in-rev.gsm
${MONITORDIR}/${CALLFILENAME}-out-rev.gsm)

exten => s,8,System(/usr/local/bin/sox ${MONITORDIR}/${CALLFILENAME}-rev.gsm
${MONITORDIR}/${CALLFILENAME}.gsm reverse)

exten => s,9,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-rev.gsm)

exten => s,10,System(sox ${MONITORDIR}/${CALLFILENAME}.gsm -g
${MONITORDIR}/${CALLFILENAME}.wav
exten => s,11,NoOp



== Spawn extension (sip, 18005551212, 2) exited non-zero on 'SIP/one-8e46'
-- Executing Macro("SIP/one-8e46", "record-cleanup") in new stack
-- Executing GotoIf("SIP/one-8e46", "0?11:2") in new stack
-- Goto (macro-record-cleanup,s,2)
-- Executing SetVar("SIP/one-8e46",
"MONITORDIR=/var/spool/asterisk/monitor") in new stack
-- Executing System("SIP/one-8e46", "/usr/local/bin/sox
/var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav
/var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav
reverse") in new stack
WARNING[1209277232]: File app_system.c, Line 57 (system_exec): Unable to
execute '/usr/local/bin/sox
/var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav
/var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav
reverse'
  == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
'SIP/one-8e46' in macro 'record-cleanup'
  == Spawn extension (sip, s, 1) exited non-zero on 'SIP/one-8e46'


Jimmy Riley
Network Administrator
VeriCore
985-626-1701 X1103

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Re: [Asterisk-Users] High Level of CVS activity

[Tilghman Lesher]

On Sunday 11 January 2004 04:36, Dave Cotton wrote:
> I don't think I've ever seen so many commits to the CVS, I'm not
> complaining. To save lots of checkout/test cycles can a message be
> posted when the current surge is over?

Yes, there were a few of us going through the bugtracker last night
and trying various patches for the 0.7.0 release.  The ironic thing is
that approximately the same time that you posted is the same time
that a whole lot of the trackers quit and went to bed (4:30 a.m. CST).

-Tilghman

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[Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

[Richard Grinnell]

Dell - PowerEdge 400SC Server Under $300 with MIR
Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz
FSB 

For those of you who aren`t familiar with the 400SC, 
this server is an Intel i875P chipset based server
with an 8x AGP slot. 
It is compatible with 533Mhz and 800Mhz processors
(hyperthreaded too), 
there`s built in 2 channel S-ATA, there are 6 USB 2.0
ports, 
there`s built in 10/100/1000 ethernet, 4 PCI slots 
(with both 5.0v and 3.3v universal support), 
ADI 198x audio, and loads of other features. 
These puppies are incredibly fast and rock stable. 
And this deal comes with 1 year parts and Onsite labor
service too! 

Go to gotapex.com and Find on page: 400SC

Richard Grinnell

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Re: [Asterisk-Users] High Level of CVS activity

[Dave Cotton]

On Sun, 2004-01-11 at 18:00, Tilghman Lesher wrote:
> Yes, there were a few of us going through the bugtracker last night
> and trying various patches for the 0.7.0 release.  The ironic thing is
> that approximately the same time that you posted is the same time
> that a whole lot of the trackers quit and went to bed (4:30 a.m. CST).

Yes I noticed the deluge stopped almost immediately afterwards, thought
it was something I said :).  I'll update as soon as 0.7.0 is announced
as in place. The timezone thing seems to get in the way often, last week
I had someone call my * at 02.00 CET from Brazil! Really must get out of
hours configured in my dial plan.

-- 
Dave Cotton <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

[Dave Cotton]

On Sun, 2004-01-11 at 18:06, Richard Grinnell wrote:
> Dell - PowerEdge 400SC Server Under $300 with MIR
> Intel P4 Processor at 2.8GHz, 512KB Cache, 800MHz
> FSB 

Yet again the trans-Atlantic con 399¤ here = $500 when it should be 235¤

Just paid 5000¤ for a 2600SC but it is nice, if only it did not make so
much noise.

-- 
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Cisco 79xx Ringtones

[Craig Waddington]

Customizing the Cisco SIP IP Phone Ring Types

The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By
default, your ring type options will be those two choices. However,
using the RINGLIST.DAT file, you can customize the ring types that are
available to the Cisco SIP IP phone users.




Step 1   Create a pulse code modulation (PCM) file of the desired ring
types and store the PCM files in the root directory of your TFTP server.
PCM files must contain no header information and comply with the
following format guidelines:


8000 Hz sampling rate 
8 bits per sample 
ulaw compression 

Step 2   Using a ASCII editor, open the RINGLIST.DAT file and for each
of the ring types you are adding, specify the name as you want it to
display on the Ring Type menu, press Tab, and then specify the filename
of the ring type. For example, the format of a pointer in your
RINGLIST.DAT file should appear similar to the following:

Ring Type 1 ringer1.pcm


Step 3   After defining pointers for each of the ring types you are
adding, save your modifications and close the RINGLIST.DAT file.


http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administr
ation_guide_chapter09186a0080087511.html#1042487


-C


www.ntfs.org

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adthrawn
Sent: 11 January 2004 16:27
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 79xx Ringtones

Hi,

I'm after two very specific ringtones for the 79xx's...

A dog barking, and a horse either galloping or neighing.

I've tried making the sounds, but for some bizarre reason they're not 
working. I used to make quite a few ringtones for the 79xx's, but I 
seem to have forgotten how to do it! And to top things off, I can't 
even find the documentation on Cisco's site for making new ringtones.

I do recall, you had to set the sample length to a divisible, something 
like 800? And there was a maximum sample length too...

Best,
Ad.

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RE: [Asterisk-Users] Cisco 79xx Ringtones

[Brian West]

http://www.bkw.org/~brian/cisco/7960-ringtones/rings/

bkw

On Sun, 11 Jan 2004, Craig Waddington wrote:

> Customizing the Cisco SIP IP Phone Ring Types
>
> The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By
> default, your ring type options will be those two choices. However,
> using the RINGLIST.DAT file, you can customize the ring types that are
> available to the Cisco SIP IP phone users.
>
>
> 
> 
> Step 1   Create a pulse code modulation (PCM) file of the desired ring
> types and store the PCM files in the root directory of your TFTP server.
> PCM files must contain no header information and comply with the
> following format guidelines:
>
>
> 8000 Hz sampling rate
> 8 bits per sample
> ulaw compression
>
> Step 2   Using a ASCII editor, open the RINGLIST.DAT file and for each
> of the ring types you are adding, specify the name as you want it to
> display on the Ring Type menu, press Tab, and then specify the filename
> of the ring type. For example, the format of a pointer in your
> RINGLIST.DAT file should appear similar to the following:
>
> Ring Type 1 ringer1.pcm
>
>
> Step 3   After defining pointers for each of the ring types you are
> adding, save your modifications and close the RINGLIST.DAT file.
>
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administr
> ation_guide_chapter09186a0080087511.html#1042487
>
>
> -C
>
>
> www.ntfs.org
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Adthrawn
> Sent: 11 January 2004 16:27
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Cisco 79xx Ringtones
>
> Hi,
>
> I'm after two very specific ringtones for the 79xx's...
>
> A dog barking, and a horse either galloping or neighing.
>
> I've tried making the sounds, but for some bizarre reason they're not
> working. I used to make quite a few ringtones for the 79xx's, but I
> seem to have forgotten how to do it! And to top things off, I can't
> even find the documentation on Cisco's site for making new ringtones.
>
> I do recall, you had to set the sample length to a divisible, something
> like 800? And there was a maximum sample length too...
>
> Best,
> Ad.
>
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Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

[Tom Johnson]

Got one, and it is rock solid and totally quiet. Had an Athlon XP1800+
before with a screaming 7k RPM fan. Never even knew the clock on the wall
actually ticked until I shutdown that noisemaker and fired up this 400sc. It
is a bit picky about memory to take advantage of the bus speed, stick with
the good stuff or it will fall back to a slower speed.

Tom Johnson
StarWISP Broadband

- Original Message - 
From: "Richard Grinnell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, January 11, 2004 11:06 AM
Subject: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary
Pentium4, not Xeon)


>
> Dell - PowerEdge 400SC Server Under $300 with MIR
> Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz
> FSB
>
> For those of you who aren`t familiar with the 400SC,
> this server is an Intel i875P chipset based server
> with an 8x AGP slot.
> It is compatible with 533Mhz and 800Mhz processors
> (hyperthreaded too),
> there`s built in 2 channel S-ATA, there are 6 USB 2.0
> ports,
> there`s built in 10/100/1000 ethernet, 4 PCI slots
> (with both 5.0v and 3.3v universal support),
> ADI 198x audio, and loads of other features.
> These puppies are incredibly fast and rock stable.
> And this deal comes with 1 year parts and Onsite labor
> service too!
>
> Go to gotapex.com and Find on page: 400SC
>
> Richard Grinnell
>
> __
> Do you Yahoo!?
> Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes
> http://hotjobs.sweepstakes.yahoo.com/signingbonus
> ___
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Re: [Asterisk-Users] macro error "exited non-zero"

[Andres]

On Sunday 11 January 2004 11:47, Jimmy Riley wrote:
>   On this macro I keep getting exited non-zero on s,3, but s,3 is doing
> what it is suppose to do but the macro stops. Is there a way to make a
> macro ignore errors and continue to s,4? I have the lattes ver of sox
> 12.17.4.

Are you using RedHat 9?

>
> Also if I just run this line from the command line I don't get an error.
> [EMAIL PROTECTED] monitor]# sox in.wav in-rev.wav reverse
> [EMAIL PROTECTED] monitor]#
>
>
> [macro-record-cleanup]
>
> exten => s,1,GotoIf($[${CALLFILENAME} = ${FOO}]?11:2)
> exten => s,2,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
>
> exten => s,3,System(/usr/local/bin/sox ${MONITORDIR}/${CALLFILENAME}-in.wav
> ${MONITORDIR}/${CALLFILENAME}-in-rev.wav reverse)
> exten => s,4,System(/usr/local/bin/sox
> ${MONITORDIR}/${CALLFILENAME}-out.wav
> ${MONITORDIR}/${CALLFILENAME}-out-rev.wav reverse)
>
> exten => s,5,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav
> ${MONITORDIR}/${CALLFILENAME}-out.wav)
>
> exten => s,6,System(/usr/local/bin/soxmix
> ${MONITORDIR}/${CALLFILENAME}-in-rev.wav
> ${MONITORDIR}/${CALLFILENAME}-out-rev.wav
> ${MONITORDIR}/${CALLFILENAME}-rev.gsm)
>
> exten => s,7,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in-rev.gsm
> ${MONITORDIR}/${CALLFILENAME}-out-rev.gsm)
>
> exten => s,8,System(/usr/local/bin/sox
> ${MONITORDIR}/${CALLFILENAME}-rev.gsm ${MONITORDIR}/${CALLFILENAME}.gsm
> reverse)
>
> exten => s,9,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-rev.gsm)
>
> exten => s,10,System(sox ${MONITORDIR}/${CALLFILENAME}.gsm -g
> ${MONITORDIR}/${CALLFILENAME}.wav
> exten => s,11,NoOp
>
>
>
> == Spawn extension (sip, 18005551212, 2) exited non-zero on 'SIP/one-8e46'
> -- Executing Macro("SIP/one-8e46", "record-cleanup") in new stack
> -- Executing GotoIf("SIP/one-8e46", "0?11:2") in new stack
> -- Goto (macro-record-cleanup,s,2)
> -- Executing SetVar("SIP/one-8e46",
> "MONITORDIR=/var/spool/asterisk/monitor") in new stack
> -- Executing System("SIP/one-8e46", "/usr/local/bin/sox
> /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav
> /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav
> reverse") in new stack
> WARNING[1209277232]: File app_system.c, Line 57 (system_exec): Unable to
> execute '/usr/local/bin/sox
> /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav
> /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav
> reverse'
>   == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
> 'SIP/one-8e46' in macro 'record-cleanup'
>   == Spawn extension (sip, s, 1) exited non-zero on 'SIP/one-8e46'
>
>
> Jimmy Riley
> Network Administrator
> VeriCore
> 985-626-1701 X1103
>
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Re: [Asterisk-Users] Asteriks as SIP<>H323 Proxy?

[Siggi Langauf]

On Sat, 10 Jan 2004, Arnd Vehling wrote:

> is it possible to use Asteriks for translating SIP to H323 and vice versa?
> I am looking to implement the following Setup
>
> SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC
>
> Basicly i want SIP fones to talk to H323 fones and and SIP Fones to
> access PSTN Gateway(s) in a H323 network.
>
> Anyone got something similiar running? Any ideas?

Yes, Asterisk can do that fine.
Unless you have an H.323 device with multiple IP addresses, like a CCM
cluster with Skinny phones. In that case, you'll need chan_oh323.

In any case, you'll have to install recent versions of openh323 (and
pwlib) and build the H.323 channel driver (whichever you choose) manually.
It won't work with all devices, though. (I have experienced some trouble
when codecs don't match on H.323 and SIP sides, but that can usually be
fixed by configuring allowed codecs explicitly, if all devices are under
your control.)
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RE: [Asterisk-Users] macro error "exited non-zero"

[Jimmy Riley]

Jimmy Riley
Network Administrator
VeriCore
985-626-1701 X1103
-Original Message-
From: Andres [mailto:[EMAIL PROTECTED] 
Sent: January 11, 2004 12:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] macro error "exited non-zero"

On Sunday 11 January 2004 11:47, Jimmy Riley wrote:
>   On this macro I keep getting exited non-zero on s,3, but s,3 is doing
> what it is suppose to do but the macro stops. Is there a way to make a
> macro ignore errors and continue to s,4? I have the lattes ver of sox
> 12.17.4.

Are you using RedHat 9? Yes

>
> Also if I just run this line from the command line I don't get an error.
> [EMAIL PROTECTED] monitor]# sox in.wav in-rev.wav reverse
> [EMAIL PROTECTED] monitor]#
>
>
> [macro-record-cleanup]
>
> exten => s,1,GotoIf($[${CALLFILENAME} = ${FOO}]?11:2)
> exten => s,2,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
>
> exten => s,3,System(/usr/local/bin/sox
${MONITORDIR}/${CALLFILENAME}-in.wav
> ${MONITORDIR}/${CALLFILENAME}-in-rev.wav reverse)
> exten => s,4,System(/usr/local/bin/sox
> ${MONITORDIR}/${CALLFILENAME}-out.wav
> ${MONITORDIR}/${CALLFILENAME}-out-rev.wav reverse)
>
> exten => s,5,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav
> ${MONITORDIR}/${CALLFILENAME}-out.wav)
>
> exten => s,6,System(/usr/local/bin/soxmix
> ${MONITORDIR}/${CALLFILENAME}-in-rev.wav
> ${MONITORDIR}/${CALLFILENAME}-out-rev.wav
> ${MONITORDIR}/${CALLFILENAME}-rev.gsm)
>
> exten => s,7,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in-rev.gsm
> ${MONITORDIR}/${CALLFILENAME}-out-rev.gsm)
>
> exten => s,8,System(/usr/local/bin/sox
> ${MONITORDIR}/${CALLFILENAME}-rev.gsm ${MONITORDIR}/${CALLFILENAME}.gsm
> reverse)
>
> exten => s,9,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-rev.gsm)
>
> exten => s,10,System(sox ${MONITORDIR}/${CALLFILENAME}.gsm -g
> ${MONITORDIR}/${CALLFILENAME}.wav
> exten => s,11,NoOp
>
>
>
> == Spawn extension (sip, 18005551212, 2) exited non-zero on 'SIP/one-8e46'
> -- Executing Macro("SIP/one-8e46", "record-cleanup") in new stack
> -- Executing GotoIf("SIP/one-8e46", "0?11:2") in new stack
> -- Goto (macro-record-cleanup,s,2)
> -- Executing SetVar("SIP/one-8e46",
> "MONITORDIR=/var/spool/asterisk/monitor") in new stack
> -- Executing System("SIP/one-8e46", "/usr/local/bin/sox
> /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav
> /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav
> reverse") in new stack
> WARNING[1209277232]: File app_system.c, Line 57 (system_exec): Unable to
> execute '/usr/local/bin/sox
> /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav
> /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav
> reverse'
>   == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
> 'SIP/one-8e46' in macro 'record-cleanup'
>   == Spawn extension (sip, s, 1) exited non-zero on 'SIP/one-8e46'
>
>
> Jimmy Riley
> Network Administrator
> VeriCore
> 985-626-1701 X1103
>
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Re: [Asterisk-Users] far end disconnect supervision

[Jonathan Moore]

Quoting Rich Adamson <[EMAIL PROTECTED]>:

> > > Personal opinion is that throwing a channel bank at * (other then for
> > > better echo cancellation on trunks) is like taking a shower with your
> > > socks on; don't see any practical use. But, I'm sure some do.
> >  
> > This statement assumes a lot. 
> > 
> > 1. That you don't want to use analog phones.
> > 2. That you need enough lines for a T1/PRI circuit. In this case the most
> > economical way to deploy a setup of say 5-16 voice lines is with a channel
> bank
> > and a T1 card in the * server. If there is some less expensive option
> please
> > share :-). 
> 
> Note the comment in the statement that says "other then ... trunks".
> Full agreement that most channel banks will handle echo cancellation,
> impedence matching, etc, very well. They've been doing it for years.
> 
> The comment was oriented towards using a channel bank to connect analog
> phones to *, and was not intended to start another thread, flames, etc.
> And, we've all seen the discussion relative to the cost of sip phones
> vs analog many times over, use of left over slots for phones, etc, etc.
> If you have a specific case where that makes sense for you or your 
> customer, knock yourself out (that would be the "some do" part). 
> No need to rehash the "I'm cheaper then you are" discussions again.
> 

Yes, I don't want to start a flame war and I am not really even advocating
analog phones or a discussion of the relative merits of analog/sip. We are
actually going the SIP route. My comment is more from the purspective of someone
new to phone systems and what was important for me to learn. I initially thought
we would lease a T1/PRI (which is what I normally think of as a "trunk" line,
but obviously that is my limited experience with telephony single single POTS
lines are also trunks), but because our call traffic is mostly internal we only
need somewhere in the neiborhood of 6-12 phone lines to the PSTN. It took a
while for me to realize that it was cheaper to buy a channel bank and a T1 card
than to purchase multiple single fxo cards. 



Visit Winfield Public Schools at http://usd465.com
-
This mail sent through IMP: http://horde.org/imp/
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RE : [Asterisk-Users] 2nd call leg status?

[SW]

Thanks Freddi,

Cool, it works, infact better yet ResetCDR(). ResetCDR() show just one
record in cdr, ResetCDR(w) will crete two records with the one for first
call leg, which I do not want :).

Cheers

SW

Date: Sun, 11 Jan 2004 17:27:08 +0100
From: Freddi Hansen <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: RE : [Asterisk-Users] 2nd call leg status?
Reply-To: [EMAIL PROTECTED]

Hi,
I did have the same problem. I simply issue a ResetCDR(w) as the last
thing before using
the Dial application. This will reset the 'answered' flag and the CDR
you get from the Dial
will contain the correct value. I do later do a small of backend
processing to get the correct
A-line time related to the call.
There may be smarter way of doing this but this quick hack works fine
for me
Freddi


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[Asterisk-Users] SpeakFree

[Lists]

If you did not see slashdot today, check out this anoucment.


http://www.fourmilab.ch/speakfree/


Michael

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Re: [Asterisk-Users] macro error "exited non-zero"

[Andres]

On Sunday 11 January 2004 14:11, Jimmy Riley wrote:
> Jimmy Riley
> Network Administrator
> VeriCore
> 985-626-1701 X1103
> -Original Message-
> From: Andres [mailto:[EMAIL PROTECTED]
> Sent: January 11, 2004 12:31 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] macro error "exited non-zero"
>
> On Sunday 11 January 2004 11:47, Jimmy Riley wrote:
> >   On this macro I keep getting exited non-zero on s,3, but s,3 is doing
> > what it is suppose to do but the macro stops. Is there a way to make a
> > macro ignore errors and continue to s,4? I have the lattes ver of sox
> > 12.17.4.
>
> Are you using RedHat 9? Yes
Then edit your /etc/init.d/asterisk to include the line:
export LD_ASSUME_KERNEL=2.4.1
before you start the daemon.

I don't know why but it fixed our problem (same as yours)
>
> > Also if I just run this line from the command line I don't get an error.
> > [EMAIL PROTECTED] monitor]# sox in.wav in-rev.wav reverse
> > [EMAIL PROTECTED] monitor]#
> >
> >
> > [macro-record-cleanup]
> >
> > exten => s,1,GotoIf($[${CALLFILENAME} = ${FOO}]?11:2)
> > exten => s,2,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
> >
> > exten => s,3,System(/usr/local/bin/sox
>
> ${MONITORDIR}/${CALLFILENAME}-in.wav
>
> > ${MONITORDIR}/${CALLFILENAME}-in-rev.wav reverse)
> > exten => s,4,System(/usr/local/bin/sox
> > ${MONITORDIR}/${CALLFILENAME}-out.wav
> > ${MONITORDIR}/${CALLFILENAME}-out-rev.wav reverse)
> >
> > exten => s,5,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav
> > ${MONITORDIR}/${CALLFILENAME}-out.wav)
> >
> > exten => s,6,System(/usr/local/bin/soxmix
> > ${MONITORDIR}/${CALLFILENAME}-in-rev.wav
> > ${MONITORDIR}/${CALLFILENAME}-out-rev.wav
> > ${MONITORDIR}/${CALLFILENAME}-rev.gsm)
> >
> > exten => s,7,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in-rev.gsm
> > ${MONITORDIR}/${CALLFILENAME}-out-rev.gsm)
> >
> > exten => s,8,System(/usr/local/bin/sox
> > ${MONITORDIR}/${CALLFILENAME}-rev.gsm ${MONITORDIR}/${CALLFILENAME}.gsm
> > reverse)
> >
> > exten => s,9,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-rev.gsm)
> >
> > exten => s,10,System(sox ${MONITORDIR}/${CALLFILENAME}.gsm -g
> > ${MONITORDIR}/${CALLFILENAME}.wav
> > exten => s,11,NoOp
> >
> >
> >
> > == Spawn extension (sip, 18005551212, 2) exited non-zero on
> > 'SIP/one-8e46' -- Executing Macro("SIP/one-8e46", "record-cleanup") in
> > new stack -- Executing GotoIf("SIP/one-8e46", "0?11:2") in new stack
> > -- Goto (macro-record-cleanup,s,2)
> > -- Executing SetVar("SIP/one-8e46",
> > "MONITORDIR=/var/spool/asterisk/monitor") in new stack
> > -- Executing System("SIP/one-8e46", "/usr/local/bin/sox
> > /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav
> > /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav
> > reverse") in new stack
> > WARNING[1209277232]: File app_system.c, Line 57 (system_exec): Unable to
> > execute '/usr/local/bin/sox
> > /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in.wav
> > /var/spool/asterisk/monitor/11012004-10:40:08-one-18005551212-in-rev.wav
> > reverse'
> >   == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
> > 'SIP/one-8e46' in macro 'record-cleanup'
> >   == Spawn extension (sip, s, 1) exited non-zero on 'SIP/one-8e46'
> >
> >
> > Jimmy Riley
> > Network Administrator
> > VeriCore
> > 985-626-1701 X1103
> >
> > ___
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Re: [Asterisk-Users] Cisco 79xx Ringtones

[Siggi Langauf]

Hi,

On Sun, 11 Jan 2004, Adthrawn wrote:

> I'm after two very specific ringtones for the 79xx's...
>
> A dog barking, and a horse either galloping or neighing.
[...]
> I do recall, you had to set the sample length to a divisible, something
> like 800? And there was a maximum sample length too...

Almost.
The complete specs are on
http://cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00800c2fa1.html#21431

That page also shows the complete procedure.
If your phone doesn't use Skinny firmware, you might have to update
RINGLIST.DAT instead of ringlist.xml (though I think the new SIP loads
should also work with the xml file).

HTH,
Siggi

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Re: [Asterisk-Users] SpeakFree

[Miguel Cavazos]

sad, yes but who needs speakfreely when you have asterisk and soft/hard
phones. The author seems really unmotivated so let him find the path
maybe he could join asterisk development team :)

Miguel
On Sun, 2004-01-11 at 19:33, Lists wrote:
> If you did not see slashdot today, check out this anoucment.
> 
> 
> http://www.fourmilab.ch/speakfree/
> 
> 
> Michael
> 
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RE: [Asterisk-Users] SIP reload configuration problem /* New subject */

[Christopher Raper]

New users arent being added from the sip.conf file...
let me play with it and get back to you in a few weeks when I know what I am doing! 
Newbie remember!
Dont want to send you down the wrong track and then work out that its me doing 
something wrong.

Thanks for your help


Cheers
Chris

-Original Message-
From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
Sent: Friday, 9 January 2004 7:24 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP reload configuration problem /* New
subject */


Christopher Raper wrote:
> When creating users in the sip.conf file, they do not appear when running the "sip 
> show users" 
 > command from the CLI until i restart. A reload doesnt make them appear.
The appear for me after a reload. (running the latest and most dangerous CVS, many 
patches are
added by malcolm - thank you!)

While on the topic, note that some peer data based on their registration status is 
saved in the
Asterisk database, and when doing a reload, peers already logged in will stay logged 
in.
This is a feature, not a bug :-)

Other than that, the SIP channel should be reset and new users added.

Maybe there's something else that is causing you trouble with your users?

/Olle

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[Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[David Burr]

We have a new contest starting today!

The first three members to post 300 messages at http://www.asterisk.bz 
will win a _80Gig Hard Drive!_

Its quite simple. Messages must be asterisk related.

http://www.asterisk.bz Alternative to the asterisk-users list
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RE: [Asterisk-Users] SIP reload configuration problem /* New subject */

[Brian West]

accually latest cvs has alot of minor and a few major fixes  Worth the
try.. its building up the 0.7.0 for monday. :)

bkw

On Mon, 12 Jan 2004, Christopher Raper wrote:

> New users arent being added from the sip.conf file...
> let me play with it and get back to you in a few weeks when I know what I am doing! 
> Newbie remember!
> Dont want to send you down the wrong track and then work out that its me doing 
> something wrong.
>
> Thanks for your help
>
>
> Cheers
> Chris
>
> -Original Message-
> From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
> Sent: Friday, 9 January 2004 7:24 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] SIP reload configuration problem /* New
> subject */
>
>
> Christopher Raper wrote:
> > When creating users in the sip.conf file, they do not appear when running the "sip 
> > show users"
>  > command from the CLI until i restart. A reload doesnt make them appear.
> The appear for me after a reload. (running the latest and most dangerous CVS, many 
> patches are
> added by malcolm - thank you!)
>
> While on the topic, note that some peer data based on their registration status is 
> saved in the
> Asterisk database, and when doing a reload, peers already logged in will stay logged 
> in.
> This is a feature, not a bug :-)
>
> Other than that, the SIP channel should be reset and new users added.
>
> Maybe there's something else that is causing you trouble with your users?
>
> /Olle
>
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Re: [Asterisk-Users] SpeakFree

[Brian West]

Since its public domain software we can use the encrypition part in IAX2
:)

bkw

On Sun, 11 Jan 2004, Miguel Cavazos wrote:

> sad, yes but who needs speakfreely when you have asterisk and soft/hard
> phones. The author seems really unmotivated so let him find the path
> maybe he could join asterisk development team :)
>
> Miguel
> On Sun, 2004-01-11 at 19:33, Lists wrote:
> > If you did not see slashdot today, check out this anoucment.
> >
> >
> > http://www.fourmilab.ch/speakfree/
> >
> >
> > Michael
> >
> > ___
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RE: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[Scott Stingel]

Oh, I thought it was a contest for top posters!  Darn!

David, it's a cool and clean format, but what's the matter with using the
Wiki that people already have devoted much blood sweat and tears to?   It's
not too bad - try it!

Regards
Scott

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]    
URL:www.evtmedia.com    



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Burr
Sent: Sunday, January 11, 2004 9:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive


We have a new contest starting today!

The first three members to post 300 messages at http://www.asterisk.bz 
will win a _80Gig Hard Drive!_

Its quite simple. Messages must be asterisk related.

http://www.asterisk.bz Alternative to the asterisk-users list
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RE: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[daryl]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of David Burr
> Sent: Sunday, January 11, 2004 4:31 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive
> 
> 
> We have a new contest starting today!
> 
> The first three members to post 300 messages at 
> http://www.asterisk.bz 
> will win a _80Gig Hard Drive!_
> 
> Its 
> quite simple. Messages must be asterisk related.

I can guarantee you that very few people who know anything about * will
be posting to that site.

It's a horrible interface, as nearly all forums that try to duplicate
mailing lists are.  Do you really think Linux/UNIX CLI guys want to deal
with a web site where  will
do just fine?

Highly technical forums, especially those related to primarily CLI
tools, fail miserably.

Have you not been reading this list as to the feelings of the regular
contributors and answer-providers on this issue?

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 
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[Asterisk-Users] Asterisk on FreeBSD 4.9

[NetOne Administrator]

Hi all!I'm trying to set up Asterisk on FreeBSD 4.9 to route
calls to H.323 GK.I have installed asterisk using the
ports.It seems to be running OK, but when i try to dial through
h323, it segfaults.I'm using X-Lite as SIP client, i have set up my
h323.conf:[general]port = 1721bindaddr = 0.0.0.0tos =
lowdelaydtmfmode = rfc2833context = OutnoFastStart =
yesnoH245Tunneling = nogatekeeper = 01.23.45.67AllowGKRouted =
noallow = all
[NET1-BG]type=h323prefix=0context=Incand in my
extensions.conf there's a line like this:exten =>
_0.,1,Dial(H323/0889811777,20) which i think should dial the
phone entered, no matter what number is dialed by the client (if it start
with 0, of course).Anyone with suggestion? where am i
wrong?Doichin DokovNetOne - Silistra0889 / 811-777

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[Asterisk-Users] Asterisk on FreeBSD 4.9?

[NetOne Administrator]

Hi all


I'm trying to set up Asterisk on FreeBSD 4.9 to route
calls to H.323 GK.
I have installed asterisk using the ports.
It seems to be running OK, but when i try to dial through
h323, it segfaults

I'm using X-Lite as SIP client, i have set up my
h323.conf
[general
port = 1721
bindaddr = 0.0.0.0
tos = lowdelaydtmfmode = rfc2833
context = Out
noFastStart = yes
noH245Tunneling = no
gatekeeper = 01.23.45.67
AllowGKRouted = no
allow = all

[NET1-BG]
type=h323
prefix=0
context=Inc


and in my extensions.conf there's a line like this:
exten = _0.,1,Dial(H323/0889811777,20)

which i think should dial the phone entered, no matter what number is
dialed by the client (if it start with 0, of course).

Anyone with suggestion? where am i wrong?

Doichin Dokov
NetOne - Silistra
0889 / 811-777
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[Asterisk-Users] "friendly" dial tone frequency combinations

[Andrew Thompson]

Hello List,

I just started messing with the settings on a SPA-2000, and it has a really
nasty alternative dial tone that I want to make go away. I'm not too hip on
how the two numbers interact, so my results haven't been good.

(I'm in the US, so I'm bias'ed towards US tones.)

Default(I'm ok with this one): [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2)

OutsideLineTone(Bad, really bad. No, worse than that): [EMAIL PROTECTED];10(*/0/1)

My only moderately successful alteration is: [EMAIL PROTECTED],[EMAIL 
PROTECTED];10(*/0/1+2)

I was wondering if any of you had worked out some dial tone combinations
that were user-friendly.

As an aside, some of you I'm sure know what the US defaults are so can you
tell me what the pieces of these configurations mean? I know the two 3digit
numbers are frequencies. I'm guessing that the -19 is gain? The rest of them
don't mean anything at all to me...

Thanks,
Andrew.


Andrew Thompson

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Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[Brancaleoni Matteo]

Hi
> http://www.asterisk.bz Alternative to the asterisk-users list

nothing against this forum, but this made me think.
I noticed that some people loose their time in
setting up doc sites... the idea is great, but
since there're already grown sites (oej's wiki),
why not stopping into doing something already
done and spend that time writing docs, for example ?

matteo

-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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RE: [Asterisk-Users] Newbie Question-Looking for Feedback

[woody+asterisk]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Christopher Raper
> Sent: Thursday, 8 January 2004 10:06
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Newbie Question-Looking for Feedback
> 
> Greetings all. I am new to the Asterisk world! Found it very 
> impressive so far!

Good.  And you are in Australia too!  For all the seppos who don't know
about Australia, it is just like Iowa except we have pet kangaroos instead
of pitchforks :-)

> In relation to the below.
> 
> I have worked with Alcatel PBX's for the last 3 years. 
> Alcatel OxE supports SIP and H323 as well.

Asterisk does both too, as well as IAX which works very well through NAT.

> As far as SIP goes I have also found the Xlite to be good for 
> soft phones. I am using one now.
> Check out www.xten.com Xlite is free and easy to use. I also 
> have been given a Pingtel SIP to play with.
> http://www.pingtel.com/ 

Many on the list use Xten, and I've heard of Pingtel, I'm not sure how well
it goes with Asterisk,
many use Grandstream, Snom, or Ci$co hardphones, and DIAX softphone.

> As far as H323 terminals go I have 
> not played with all that many, however the simple Microsoft 
> netmeeting works for testing purpose anyway.

http://openh323.org/ has many H323 apps for linux

> Now a question to all you experts out there, and this may 
> seem VERY stupid, but I have configured the sip phone and 
> have it logged in and can dial 500 to get to the sample 
> messages etc. However i cannot work out how to give the sip 
> termainal a number that can be dialled. I would assume that 
> it needs to be in the dialplan, so I have added it in via the 
> extensions.conf file, however I am sure that I have stuffed 
> the config somewhere. Can someone please point me in the 
> right direction. Would be much appreciated. Also, do i need 
> hardware to make a SIP to SIP call... eg. Compressors etc.

You also need to define the extension in /etc/asterisk/sip.conf

E.g.

[woody]
type=friend
insecure=yes
username=woody
secret=bogus
host=dynamic
defaultip=192.168.2.76

And in extensions.conf:

exten => 1976,1,Dial(SIP/woody,15,tr)

You might want to look at the wiki http://www.voip-info.org/wiki-Asterisk
which is a good place to find out how to do stuff with Asterisk.

cheers,
Woody


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[Asterisk-Users] analog or sip ? was far end disconnect supervision

[Lance Arbuckle]

Thanks to everyone that responded to my channel bank question.  Ive
decided that the Adit 600 would be a good choice.
Then I got to thinking about SIP phones and wondered if their quality
has progressed to the point that they can be deployed to customers who
"just want their phones to work" and wouldn't tolerate any SIP hickups. 
As for pricing, I would think the SIP phones would need to be in the
$200 price range to be competative with analog or ADSI phones plus a
channel bank.  I know there are lots of variables that figure into the
analog vs SIP question like number of incoming lines and how they're
delivered and the number of extensions etc   I guess what would be
helpfull to me would be some general rules of thumb that you asterisk
experts use to determine what type of extension phones to recommend for
a given customer.
Thanks

-Lance
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Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

[Brancaleoni Matteo]

hi.

> Then I got to thinking about SIP phones and wondered if their quality
> has progressed to the point that they can be deployed to customers who
> "just want their phones to work" and wouldn't tolerate any SIP hickups. 

so for that use Cisco. beside I like GS budgetones and wanna see them
work reliably, if you need a rock solid phone, cisco rules, for now :/

> As for pricing, I would think the SIP phones would need to be in the
> $200 price range to be competative with analog or ADSI phones plus a
> channel bank.  
cisco 7905

matteo

-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

[Philipp von Klitzing]

Hi!

> As for pricing, I would think the SIP phones would need to be in the
> $200 price range to be competative with analog or ADSI phones 

That would make it SNOM then, I believe. Or go look at MGCP phones.

By the way, is anyone here using the SNOM 100 or 105? If yes, could you 
drop a short note on differences between the the two 200 and the 105 that 
appear to be important to you when making a buy decision for use with 
Asterisk?  

Thanks, Philipp


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[Asterisk-Users] Re: [Asterisk-Dev] More Success on the Cisco 7920 and SCCP !!!!!

[Martin Bene]

Hi Siggi,

> > 7960 and then "Call Ended" on the Display (curious about that !!!).
> 
> That seems to be normal for the 7920. I've sniffed the registration
> procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's
> doing the same thing. Maybe that's some odd way of testing if the
> CallManager ("CCM") really works...
> 
> > After that the phone reboots and the stuff repeats
> 
> Same thing here.
> CCM does quite a few things in different order compared to 
> chan_sccp, but
> apart from that, the registration procedure seems quite similar.
> I'm still looking into the detailed differences (which is a 
> bit hard, as
> there doesn't seem to be any tool like "diff" for ethereal traces).

Since I've got a 7920 myself and am trying to get things to work: 
If you've still got access to the cisco stuff: could you make available a
tcpdump file (tcpdump -w) of a successfull callmanager registration?

I'd really like to see what the successfull tftp and skinny sessions look
like and try to duplicate that w/ asterisk.

Thanks, Martin
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Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[David Burr]

I think wiki is one of the best resources around.  I think fourms could 
be an addidional resource for the newbie, an alternative to a listserv.

Scott Stingel wrote:

Oh, I thought it was a contest for top posters!  Darn!

David, it's a cool and clean format, but what's the matter with using the
Wiki that people already have devoted much blood sweat and tears to?   It's
not too bad - try it!
Regards
Scott
Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]    
URL:www.evtmedia.com    



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Burr
Sent: Sunday, January 11, 2004 9:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive
We have a new contest starting today!

The first three members to post 300 messages at http://www.asterisk.bz 
will win a _80Gig Hard Drive!_

Its quite simple. Messages must be asterisk related.

http://www.asterisk.bz Alternative to the asterisk-users list
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Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[David Burr]

Forums are more desirable for the newbie user. I realize people are set 
the their ways. thats fine :)

[EMAIL PROTECTED] wrote:

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of David Burr
Sent: Sunday, January 11, 2004 4:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

We have a new contest starting today!

The first three members to post 300 messages at 
http://www.asterisk.bz 
will win a _80Gig Hard Drive!_

Its 
quite simple. Messages must be asterisk related.
   

I can guarantee you that very few people who know anything about * will
be posting to that site.
It's a horrible interface, as nearly all forums that try to duplicate
mailing lists are.  Do you really think Linux/UNIX CLI guys want to deal
with a web site where  will
do just fine?
Highly technical forums, especially those related to primarily CLI
tools, fail miserably.
Have you not been reading this list as to the feelings of the regular
contributors and answer-providers on this issue?
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp 
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Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[Miguel Cavazos]

The following errors occurred during your registration:
  * The username you entered as your referrer could not be located.

cant create a username

Miguel
On Sun, 2004-01-11 at 22:53, Brancaleoni Matteo wrote:
> Hi
> > http://www.asterisk.bz Alternative to the asterisk-users list
> 
> nothing against this forum, but this made me think.
> I noticed that some people loose their time in
> setting up doc sites... the idea is great, but
> since there're already grown sites (oej's wiki),
> why not stopping into doing something already
> done and spend that time writing docs, for example ?
> 
> matteo
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[Asterisk-Users] T1 Sync clarification

[John Brown (CV)]

Hi List,

After reading a bunch of the docs, list post archives, it
still seems that a clear definition of how to clock the T100P
card is muddy.

zttool says that the link is "INTERNALLY CLOCKED", 

does this mean the T100P is providing clock, or does
this mean the T100P is getting clock from the T1 line
side (ergo getting clock from the Telco) ??


If you have sync = 0   then zttool says internally clocked

if sync > 0  then zttool says "Digium..."  and link
goes into an error condition.

Thus the million dollar question is this:

What should the SYNC value be if you want to clock from 
the TELCO ?

Maybe zttool is reporting things in error ??

thanks mucho




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Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

[Miguel Cavazos]

sip phones have alot of nice features and they really work, you can try
some phones under $200 yes, but about the analog phones, people like to
have there cordless phones, or there micky mouse phone or garfield phone
so consider that.

You loss some features but your customers get the phones they want to
have in there room, office, kitchen, living room, etc. Besides you can
get cool atas under $100 USD GS or sipura.

Sip phones get old and look ugly, analog can be replace at any moment.

Miguel
On Sun, 2004-01-11 at 23:07, Lance Arbuckle wrote:
> Thanks to everyone that responded to my channel bank question.  Ive
> decided that the Adit 600 would be a good choice.
> Then I got to thinking about SIP phones and wondered if their quality
> has progressed to the point that they can be deployed to customers who
> "just want their phones to work" and wouldn't tolerate any SIP hickups. 
> As for pricing, I would think the SIP phones would need to be in the
> $200 price range to be competative with analog or ADSI phones plus a
> channel bank.  I know there are lots of variables that figure into the
> analog vs SIP question like number of incoming lines and how they're
> delivered and the number of extensions etc   I guess what would be
> helpfull to me would be some general rules of thumb that you asterisk
> experts use to determine what type of extension phones to recommend for
> a given customer.
> Thanks
> 
> -Lance
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[Asterisk-Users] NuFone Network H323 configuration?

[SamW]

I am using Nu Fone Network's h323 drivers.

I can place H323 calls using following in extensions.conf file,

exten => _1732.,1,Dial(H323/[EMAIL PROTECTED])

If I need to use h323.conf to do the same I cannot configure h323 to do the 
same. I get everyone is busy message and I do not see IP packets being 
generated by * trying to communicate to 192.168.1.2. Can someone point out 
what I am doing wrong here.

extensions.conf
---
exten => _1732.,1,Dial(H323/[EMAIL PROTECTED])
h323.conf
--
[h323-out]
type=h323
host=192.168.1.2
In addition in the * console, when I issue following command I do not see 
anything. Is there something that need to be done with codec registration 
for asterisk ?

host01*CLI> h.323 show codecs
host01*CLI>
In addition I see following H323 related messages during asterisk start in 
the Console.

 == Creating H.323 Endpoint
WARNING[1074494944]: File chan_h323.c, Line 216 (build_alias): Keyword 
203.115.15.1 does not make sense in type=h323
WARNING[1074494944]: File chan_h323.c, Line 216 (build_alias): Keyword all 
does not make sense in type=h323
WARNING[1074494944]: File chan_h323.c, Line 216 (build_alias): Keyword g729 
does not make sense in type=h323
WARNING[1074494944]: File chan_h323.c, Line 216 (build_alias): Keyword gsm 
does not make sense in type=h323
WARNING[1074494944]: File chan_h323.c, Line 216 (build_alias): Keyword ulaw 
does not make sense in type=h323
WARNING[1074494944]: File chan_h323.c, Line 216 (build_alias): Keyword alaw 
does not make sense in type=h323
  == Adding alias "slt-h323-o" to endpoint
  == Registered channel type 'H323' (The NuFone Network's Open H.323 
Channel Driver)
  == H.323 listener started



Thank you,

SamW

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Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

[Rich Adamson]

> Thanks to everyone that responded to my channel bank question.  Ive
> decided that the Adit 600 would be a good choice.
> Then I got to thinking about SIP phones and wondered if their quality
> has progressed to the point that they can be deployed to customers who
> "just want their phones to work" and wouldn't tolerate any SIP hickups. 
> As for pricing, I would think the SIP phones would need to be in the
> $200 price range to be competative with analog or ADSI phones plus a
> channel bank.  I know there are lots of variables that figure into the
> analog vs SIP question like number of incoming lines and how they're
> delivered and the number of extensions etc   I guess what would be
> helpfull to me would be some general rules of thumb that you asterisk
> experts use to determine what type of extension phones to recommend for
> a given customer.

Lots of choices ranging from about $80 to $700 (and more) depending upon
manufacturer, model, features, etc. Believe the wiki has some references
to many of them.

For business use, I've had excellent experience with the Cisco 7960 
(refurb ~$350 with the power cube), and average-moving-to-good/excellent
with the Snom 200 (using the latest firmware). Both are probably 
considered higher-end multiline business sip phones by most on this list.
There are others but I've not attempted to eval those.

Your customer is likely to drive the decision "if" you let them eval 
a few different models. Since you indicated that you're just getting
started with *, etc, pure guess is that most business sales are likely
to require a mixture of multi-line and single-line insturments. 

There has been a fair amount of list traffic relative to how various
phones support nat, call transfer, music on hold, speaker volume, 
call waiting tones, and other issues. Best guess is that you would 
likely only sell a select set of single-line and multi-line units purely 
from a support perspective, with actual proposals based on specific
requirements (eg, a location needs nat therefore this model, business
office with all internal phones likely a different model, another
business with an unlimited checkbook gets a Cisco ;).

Rule of thumb...
 - don't give an executive or check-writer a cheap phone, or one that
   is so lite-weight they pull it around their desk
 - find a single-line instrument or two you are comfortable supporting
   (seems like the list has suggested at least one vendor's cheap phone
has a high mortality rate that might be worth striking from your list)
 - understand where the ata-186 kind of boxes fit (and where they don't
   fit from a real business perpective)
 - understand the value (or lack thereof) for the phone having an 
   internal switch with two RJ45 jacks (and who's phones don't work very
   well with this)
 - keep a sharp eye on the sip marketplace going forward ;)
 - understand the value of QoS in switches
 - find a supplier that delivers & invoices reliably, and will work with
   you on defective units

If you're looking for opinions on specific models, I'm sure you'll get
a number of responses from those with favorites.



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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Steve]

On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote:
> I'm using Asterisk on a open server (no firewall or NAT) and trying to
> communicate with a Grandstream BudgeTone 102 SIP phone which is behind
> NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS
> about a week ago.  My problem is that I'm only getting half-duplex
> communication -- I can hear voice from the Asterisk server but the server
> does not understand any voice from me.  From the console "sip debug" shows
> that the SIP part is working fine and DTMF via SIP INFO works.


I use OpenBSD firewalls with NAT and redirect and it works just as it's 
supposed to. 

That's not even half duplex. In half duplex each side Can talk, but only one 
at a time. It seems to be an error with configuring your firewall. (One 
common error is to only turn on redirect. But you also need to Allow the 
traffic to flow...

-- 
Steve

__
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 and willing to handle things, or life 
   will find a way to get you good!
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Re: [Asterisk-Users] Chagres Technologies, Inc

[Steve]

On Saturday 10 January 2004 10:22 pm, Sean Cheesman wrote:
> time to take this off-list.

Pleeze!

-- 
Steve

__
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 and willing to handle things, or life 
   will find a way to get you good!
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Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?

[Tilghman Lesher]

On Sunday 11 January 2004 16:18, NetOne Administrator wrote:
> I'm trying to set up Asterisk on FreeBSD 4.9 to route
> calls to H.323 GK.
> I have installed asterisk using the ports.
> It seems to be running OK, but when i try to dial through
> h323, it segfaults

I want you to look at the headers of my reply and note that I'm running
my mail client on FreeBSD.

Now my advice:  run your Asterisk server on Linux.

-Tilghman

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[Asterisk-Users] More words for Allison

[John Todd]

Here's the latest batch of words to get shipped out to Allison Smith. 
Please submit reasonably small changes to me by tomorrow 10:00 AM 
Eastern time, and I'll add them.

As usual, donations to what will be a ~$110 USD expense would be 
appreciated, as I am paying for this round out of my pocket.  Please 
send to paypal address "[EMAIL PROTECTED]".  I did not include all 
possible symbols on a North American keyboard, as it was getting 
exhausting and possibly silly.

calls
abandons
staffing
average Speed of Answer
Sorry, but the user's mailbox can't accept more messages.
Please enter the conference call number for the conference you wish to join.
fortieth
fiftieth
Please enter the conference pin number.
That pin is invalid for this conference.
[The alphabet - a through z, like "ayy", "bee", "cee", etc.]
zed
space
dash
dot
comma
slash
exclamation point
ampersand
percent
at sign [we want a verbalization of the @ symbol]
with
plus
equals
left bracket
right bracket
open parenthesis
close parenthesis
pipe
backslash
comma
period
quote
greater than
less than
chance of
cloudy
sunny
sun
turning to
rainy
rain
partly
partially
mostly
snowy
snow
scattered
patchy
wind
windy
miles per hour
kilometers per hour
knots per hour
storm
warning
watch
thunderstorm
hail
weather
lightning
fog
foggy
sleet
sleeting
clear
clearing
freezing
freeze
hurricane
tornado
severe
later
morning
afternoon
evening
late
early
changing
in the
Alpha
Bravo
Charlie
Delta
Echo
Foxtrot
Golf
Hotel
India
Juliet
Kilo
Lima
Mike
November
Oscar
Papa
Quebec
Romeo
Sierra
Tango
Uniform
Victor
Whiskey
Xray
Yankee
Zulu
Niner
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RE: [Asterisk-Users] T1 Sync clarification

[Don Pobanz]

On Sunday, January 11, 2004 5:41 PM,  John Brown (CV) 
[SMTP:[EMAIL PROTECTED] wrote:
> Hi List,
>
> After reading a bunch of the docs, list post archives, it
> still seems that a clear definition of how to clock the T100P
> card is muddy.
>
> zttool says that the link is "INTERNALLY CLOCKED",
>
> does this mean the T100P is providing clock, or does
> this mean the T100P is getting clock from the T1 line
> side (ergo getting clock from the Telco) ??
>

This was really confusing for me when I started. Let me explain it this 
way.
If you want to run on the internal T100P clock then set sync to '0'
To derive the timing from the incoming T1 line (loop timing) set sync 
to '1'

>
> If you have sync = 0   then zttool says internally clocked
>
> if sync > 0  then zttool says "Digium..."  and link
> goes into an error condition.

I don't know what this would be.

>
> Thus the million dollar question is this:
>
> What should the SYNC value be if you want to clock from
> the TELCO ?

sync should be set to 1 to time from telco.

>
> Maybe zttool is reporting things in error ??
>
> thanks mucho

--
Don Pobanz
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[Asterisk-Users] zttool and errors

[John Brown (CV)]

It appears that  zttool doesn't actually report T1 span
errors.

If I inject BPV's, crc errors, framing errors, etc into
a T1 span, the counters on zttool  don't change.



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Re: [Asterisk-Users] T1 Sync clarification

[John Brown (CV)]

THank you.  Thats what I thought it should be.

Off to call the telco and tell them they are mucked up.


On Sun, Jan 11, 2004 at 06:54:11PM -0600, Don Pobanz wrote:
> On Sunday, January 11, 2004 5:41 PM,  John Brown (CV) 
> [SMTP:[EMAIL PROTECTED] wrote:
> > Hi List,
> >
> > After reading a bunch of the docs, list post archives, it
> > still seems that a clear definition of how to clock the T100P
> > card is muddy.
> >
> > zttool says that the link is "INTERNALLY CLOCKED",
> >
> > does this mean the T100P is providing clock, or does
> > this mean the T100P is getting clock from the T1 line
> > side (ergo getting clock from the Telco) ??
> >
> 
> This was really confusing for me when I started. Let me explain it this 
> way.
> If you want to run on the internal T100P clock then set sync to '0'
> To derive the timing from the incoming T1 line (loop timing) set sync 
> to '1'
> 
> >
> > If you have sync = 0   then zttool says internally clocked
> >
> > if sync > 0  then zttool says "Digium..."  and link
> > goes into an error condition.
> 
> I don't know what this would be.
> 
> >
> > Thus the million dollar question is this:
> >
> > What should the SYNC value be if you want to clock from
> > the TELCO ?
> 
> sync should be set to 1 to time from telco.
> 
> >
> > Maybe zttool is reporting things in error ??
> >
> > thanks mucho
> 
> --
> Don Pobanz
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RE: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[daryl]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of David Burr
> Sent: Sunday, January 11, 2004 6:32 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive
> 
> 
> Forums are more desirable for the newbie user. I realize 
> people are set 
> the their ways. thats fine :)

That's my point.

So who's going to answer all of the newbie questions?

Other newbies?

That's the problem.

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 
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Re: [Asterisk-Users] More words for Allison

[Brian West]

Add Celsius Fahrenheit

bkw

On Sun, 11 Jan 2004, John Todd wrote:

>
> Here's the latest batch of words to get shipped out to Allison Smith.
> Please submit reasonably small changes to me by tomorrow 10:00 AM
> Eastern time, and I'll add them.
>
> As usual, donations to what will be a ~$110 USD expense would be
> appreciated, as I am paying for this round out of my pocket.  Please
> send to paypal address "[EMAIL PROTECTED]".  I did not include all
> possible symbols on a North American keyboard, as it was getting
> exhausting and possibly silly.
>
> calls
> abandons
> staffing
> average Speed of Answer
> Sorry, but the user's mailbox can't accept more messages.
> Please enter the conference call number for the conference you wish to join.
> fortieth
> fiftieth
> Please enter the conference pin number.
> That pin is invalid for this conference.
> [The alphabet - a through z, like "ayy", "bee", "cee", etc.]
> zed
> space
> dash
> dot
> comma
> slash
> exclamation point
> ampersand
> percent
> at sign [we want a verbalization of the @ symbol]
> with
> plus
> equals
> left bracket
> right bracket
> open parenthesis
> close parenthesis
> pipe
> backslash
> comma
> period
> quote
> greater than
> less than
> chance of
> cloudy
> sunny
> sun
> turning to
> rainy
> rain
> partly
> partially
> mostly
> snowy
> snow
> scattered
> patchy
> wind
> windy
> miles per hour
> kilometers per hour
> knots per hour
> storm
> warning
> watch
> thunderstorm
> hail
> weather
> lightning
> fog
> foggy
> sleet
> sleeting
> clear
> clearing
> freezing
> freeze
> hurricane
> tornado
> severe
> later
> morning
> afternoon
> evening
> late
> early
> changing
> in the
>
> Alpha
> Bravo
> Charlie
> Delta
> Echo
> Foxtrot
> Golf
> Hotel
> India
> Juliet
> Kilo
> Lima
> Mike
> November
> Oscar
> Papa
> Quebec
> Romeo
> Sierra
> Tango
> Uniform
> Victor
> Whiskey
> Xray
> Yankee
> Zulu
> Niner
>
> ___
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Re: [Asterisk-Users] More words for Allison

[John Brown (CV)]

How about  Dollars  Cents  Euros


On Sun, Jan 11, 2004 at 07:36:45PM -0500, John Todd wrote:
> 
> Here's the latest batch of words to get shipped out to Allison Smith. 
> Please submit reasonably small changes to me by tomorrow 10:00 AM 
> Eastern time, and I'll add them.
> 
> As usual, donations to what will be a ~$110 USD expense would be 
> appreciated, as I am paying for this round out of my pocket.  Please 
> send to paypal address "[EMAIL PROTECTED]".  I did not include all 
> possible symbols on a North American keyboard, as it was getting 
> exhausting and possibly silly.
> 
> calls
> abandons
> staffing
> average Speed of Answer
> Sorry, but the user's mailbox can't accept more messages.
> Please enter the conference call number for the conference you wish to join.
> fortieth
> fiftieth
> Please enter the conference pin number.
> That pin is invalid for this conference.
> [The alphabet - a through z, like "ayy", "bee", "cee", etc.]
> zed
> space
> dash
> dot
> comma
> slash
> exclamation point
> ampersand
> percent
> at sign [we want a verbalization of the @ symbol]
> with
> plus
> equals
> left bracket
> right bracket
> open parenthesis
> close parenthesis
> pipe
> backslash
> comma
> period
> quote
> greater than
> less than
> chance of
> cloudy
> sunny
> sun
> turning to
> rainy
> rain
> partly
> partially
> mostly
> snowy
> snow
> scattered
> patchy
> wind
> windy
> miles per hour
> kilometers per hour
> knots per hour
> storm
> warning
> watch
> thunderstorm
> hail
> weather
> lightning
> fog
> foggy
> sleet
> sleeting
> clear
> clearing
> freezing
> freeze
> hurricane
> tornado
> severe
> later
> morning
> afternoon
> evening
> late
> early
> changing
> in the
> 
> Alpha
> Bravo
> Charlie
> Delta
> Echo
> Foxtrot
> Golf
> Hotel
> India
> Juliet
> Kilo
> Lima
> Mike
> November
> Oscar
> Papa
> Quebec
> Romeo
> Sierra
> Tango
> Uniform
> Victor
> Whiskey
> Xray
> Yankee
> Zulu
> Niner
> 
> ___
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[Asterisk-Users] Forward call with response required to accept

[Glenn Dalgliesh]

I am looking for a way to Forward to a external or 
internal number and require a digit(s) in order to complete 
forward.
 
Example:
 
PSTN1 Calls * dials PSTN2
    
if PSTN2 presses proper digits bridge the PSTN1 and 
PSTN2
if 
no response return to a context
 
Reasons: 2 actually
    1. call is forwarded to cell 
phone but If cell is out of range, turned off, or they don't answer I don't want 
the calling party to get connected to the Cell phones VM
    2. Call is forwarded to outside 
number and I want a level of security that ensures that the person that the call 
is intended for is present.
 
Any help would be appreciated
Thanks 
 

Re: [Asterisk-Users] More words for Allison

[Gary]

Sex
drigs
rock & roll 

???:-)


On Sun, 11 Jan 2004 19:20:11 -0600 (CST), Brian West wrote:

>Add Celsius Fahrenheit
>
>bkw
>
>On Sun, 11 Jan 2004, John Todd wrote:
>
>>
>> Here's the latest batch of words to get shipped out to Allison Smith.
>> Please submit reasonably small changes to me by tomorrow 10:00 AM
>> Eastern time, and I'll add them.
>>
>> As usual, donations to what will be a ~$110 USD expense would be
>> appreciated, as I am paying for this round out of my pocket.  Please
>> send to paypal address "[EMAIL PROTECTED]".  I did not include all
>> possible symbols on a North American keyboard, as it was getting
>> exhausting and possibly silly.
>>
>> calls
>> abandons
>> staffing
>> average Speed of Answer
>> Sorry, but the user's mailbox can't accept more messages.
>> Please enter the conference call number for the conference you wish to join.
>> fortieth
>> fiftieth
>> Please enter the conference pin number.
>> That pin is invalid for this conference.
>> [The alphabet - a through z, like "ayy", "bee", "cee", etc.]
>> zed
>> space
>> dash
>> dot
>> comma
>> slash
>> exclamation point
>> ampersand
>> percent
>> at sign [we want a verbalization of the @ symbol]
>> with
>> plus
>> equals
>> left bracket
>> right bracket
>> open parenthesis
>> close parenthesis
>> pipe
>> backslash
>> comma
>> period
>> quote
>> greater than
>> less than
>> chance of
>> cloudy
>> sunny
>> sun
>> turning to
>> rainy
>> rain
>> partly
>> partially
>> mostly
>> snowy
>> snow
>> scattered
>> patchy
>> wind
>> windy
>> miles per hour
>> kilometers per hour
>> knots per hour
>> storm
>> warning
>> watch
>> thunderstorm
>> hail
>> weather
>> lightning
>> fog
>> foggy
>> sleet
>> sleeting
>> clear
>> clearing
>> freezing
>> freeze
>> hurricane
>> tornado
>> severe
>> later
>> morning
>> afternoon
>> evening
>> late
>> early
>> changing
>> in the
>>
>> Alpha
>> Bravo
>> Charlie
>> Delta
>> Echo
>> Foxtrot
>> Golf
>> Hotel
>> India
>> Juliet
>> Kilo
>> Lima
>> Mike
>> November
>> Oscar
>> Papa
>> Quebec
>> Romeo
>> Sierra
>> Tango
>> Uniform
>> Victor
>> Whiskey
>> Xray
>> Yankee
>> Zulu
>> Niner
>>
>> ___
>> Asterisk-Users mailing list
>> [EMAIL PROTECTED]
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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Re: [Asterisk-Users] question re voicemail

[Glenn Dalgliesh]

I think this is the syntax you are looking 
for
 
[sip]exten => 
5104112978,1,Dial(SIP/5104112978,20,tr)exten 
=> 5104112978,2,Voicemail,u5104112978
exten => 
5104112978,102,Voicemail,b5104112978
 

  - Original Message - 
  From: 
  Jess Magnaye 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Monday, January 05, 2004 4:28 
  PM
  Subject: [Asterisk-Users] question re 
  voicemail
  
  Hi,
  I just setup my * with digium. I started testing 
  voicemail first between atas, and i am not sure why it is not prompting me any 
  when the call is not answered or if busy.  i only get continuous 
  ringback and the following message: 
   
  asterisk*CLI>     -- 
  Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new 
  stack    -- Called 5104112978    -- 
  SIP/5104112978-3f88 is ringing    -- Nobody picked up in 
  2 ms
   
  I wonder if my u and 
  b config is correct, mispelled, or something else is 
  missing.  Note that ata to ata via * 
  works, as well as getting to VoicemailMain via extension 
  1234.    Please help.  My config are found below.  I 
  appreciate your help.
   
   
  sip.conf
  ---
   
  [6882332]type=friendusername=6882332secret=testhost=dynamicdefaultip=172.30.200.27dtmfmode=rfc2833mailbox=6882332callerid 
  = "test1" <6882332>context=sip
  [5104112978]type=friendusername=5104112978secret=testhost=dynamic;canreinvite=nodefaultip=172.30.200.26dtmfmode=rfc2833mailbox=5104112978callerid 
  = "test2" <5104112978>context=sip
  extensions.conf
  
  voicemail.conf
  -
   
  [default]6882332 => 
  6882332,test1,[EMAIL PROTECTED]
  5104112978 => 5104112978,test2, [EMAIL PROTECTED]
  9011 => 9011,Asterisk,[EMAIL PROTECTED] 
  => ,Nada,[EMAIL PROTECTED]

RE: [Asterisk-Users] More words for Allison

[woody+asterisk]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of John Todd
> Sent: Monday, 12 January 2004 11:37
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] More words for Allison
> 

> knots per hour

I'm a land-lubber, but I think knots is a speed unit (like Miles Per Hour),
so I think you want "knots" here, not "knots per hour", if you are talking
wind speed.

One small request:

"All Your Base Are Belong To Us"

(Grammar deliberately wrong)
Optionally a second recording spoken in a monotone.

Cheers,
Woody



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[Asterisk-Users] WTS (200) AC Power Adapters for Cisco 7910 / 7940 / 7960 IP Phones

[Sales]

Have (200) Brand New power cubes (AC Power Adapter 
with AC Cord) - compatible with Cisco CP-7910, CP-7940, CP-7960 and equivalent 
"G" models.
 
$25/ea - Minimum Purchase (10) Units.
 
Email [EMAIL PROTECTED] if 
interested.
 
Regards
 
Cory 
Andrews***b2 
Technologies***
web - www.ValueResale.com email - [EMAIL PROTECTED]

[Asterisk-Users] possible solution to PRI T100P dropped call issue

[John Brown (CV)]

To recap:

T100P card wouldn't sync with the telco using line side
clocking ( span=1,1,0.) 

Had to use  internal clocking (span=1,0,0...)

zttool  showed Tx/Rx Levels as  0/ 1


For the grins of it I replaced the T100P card with
another newer card from inventory.

This newer card has the same rev on the ASIC / FPGA
but doesn't have any of the various jumper headers
installed on the board.

Set clocking to line side
sync=1,1,0..

Poof, card comes up, span comes up. 

zttool  now shows  Tx/Rx Levels as  0/ 0

No other changes have been made to the various
configs on the system.


It seems in the end that this was some strange hardware
failure on the T100P card.


:


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Re: [Asterisk-Users] More words for Allison

[Tilghman Lesher]

On Sunday 11 January 2004 19:26, John Brown (CV) wrote:
> How about  Dollars  Cents  Euros

Dollars, Cents, Zulu, and At are already done.

-Tilghman

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[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2448 - 10 msgs

[Adthrawn]

John,

Pounds
Sterling
Screw You (purely optional :-)
Your Call Is Being Connected
Please Wait
Sorry, Your Call Will Be Answered Soon
Your Call Is Important. (That is why we have not yet bothered to answer 
it, instead dancing around the office high on tip-ex.)

And for the girlfriend filter:

Sorry, that number has not been recognized. Please hangup and try again.
or
This number is no longer in operation, Please Hangup.
or
Welcome to the FBI Regional Headquarters, how may I help?
or
This number is busy. Please try again later.
or
The network is busy. Please try again later.
or
Your call could not be connected. Please try again later.
Best,
Ad.
Possibly something to frighten the telesales?
How about: "Thank you for calling. There is nobody available to take 
your call. If you leave a long, loud message, we will return your call 
at the most inconvenient time - most likely at 2am or whenever you are 
asleep. Whilst we have been distracting you, our phone system has 
captured the exact whereabouts of your dialing location. We now know 
where you live. Thank you, and enjoy your day."

On 12 Jan 2004, at 1:56 am, [EMAIL PROTECTED] 
wrote:

Message: 9
From: "Gary" <[EMAIL PROTECTED]>
To: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Date: Mon, 12 Jan 2004 11:34:49 +1000
Subject: Re: [Asterisk-Users] More words for Allison
Reply-To: [EMAIL PROTECTED]
Sex
drigs
rock & roll
???:-)

On Sun, 11 Jan 2004 19:20:11 -0600 (CST), Brian West wrote:

Add Celsius Fahrenheit

bkw

On Sun, 11 Jan 2004, John Todd wrote:

Here's the latest batch of words to get shipped out to Allison Smith.
Please submit reasonably small changes to me by tomorrow 10:00 AM
Eastern time, and I'll add them.
As usual, donations to what will be a ~$110 USD expense would be
appreciated, as I am paying for this round out of my pocket.  Please
send to paypal address "[EMAIL PROTECTED]".  I did not include all
possible symbols on a North American keyboard, as it was getting
exhausting and possibly silly.
calls
abandons
staffing
average Speed of Answer
Sorry, but the user's mailbox can't accept more messages.
Please enter the conference call number for the conference you wish 
to join.
fortieth
fiftieth
Please enter the conference pin number.
That pin is invalid for this conference.
[The alphabet - a through z, like "ayy", "bee", "cee", etc.]
zed
space
dash
dot
comma
slash
exclamation point
ampersand
percent
at sign [we want a verbalization of the @ symbol]
with
plus
equals
left bracket
right bracket
open parenthesis
close parenthesis
pipe
backslash
comma
period
quote
greater than
less than
chance of
cloudy
sunny
sun
turning to
rainy
rain
partly
partially
mostly
snowy
snow
scattered
patchy
wind
windy
miles per hour
kilometers per hour
knots per hour
storm
warning
watch
thunderstorm
hail
weather
lightning
fog
foggy
sleet
sleeting
clear
clearing
freezing
freeze
hurricane
tornado
severe
later
morning
afternoon
evening
late
early
changing
in the

Alpha
Bravo
Charlie
Delta
Echo
Foxtrot
Golf
Hotel
India
Juliet
Kilo
Lima
Mike
November
Oscar
Papa
Quebec
Romeo
Sierra
Tango
Uniform
Victor
Whiskey
Xray
Yankee
Zulu
Niner
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Re: [Asterisk-Users] More words for Allison

[Tilghman Lesher]

On Sunday 11 January 2004 18:36, John Todd wrote:
> Here's the latest batch of words to get shipped out to Allison Smith.
> Please submit reasonably small changes to me by tomorrow 10:00 AM
> Eastern time, and I'll add them.
> storm
> warning
> watch
> thunderstorm
> hail
> weather
> lightning
> fog
> foggy
> sleet
> sleeting
> clear
> clearing
> freezing
> freeze
> hurricane
> tornado
> severe

approximately
dopplar radar
indicated
local authorities
sighted
National Weather Service
has issued a
moving
north
south
east
west
Persons in the immediate path of the
are advised to seek shelter immediately.

-Tilghman

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Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

[Lance Arbuckle]

Rich Adamson wrote:
> 
> > Thanks to everyone that responded to my channel bank question.  Ive
> > decided that the Adit 600 would be a good choice.
> > Then I got to thinking about SIP phones and wondered if their quality
> > has progressed to the point that they can be deployed to customers who
> > "just want their phones to work" and wouldn't tolerate any SIP hickups.
> > As for pricing, I would think the SIP phones would need to be in the
> > $200 price range to be competative with analog or ADSI phones plus a
> > channel bank.  I know there are lots of variables that figure into the
> > analog vs SIP question like number of incoming lines and how they're
> > delivered and the number of extensions etc   I guess what would be
> > helpfull to me would be some general rules of thumb that you asterisk
> > experts use to determine what type of extension phones to recommend for
> > a given customer.
> 
> Lots of choices ranging from about $80 to $700 (and more) depending upon
> manufacturer, model, features, etc. Believe the wiki has some references
> to many of them.
> 
> For business use, I've had excellent experience with the Cisco 7960
> (refurb ~$350 with the power cube), and average-moving-to-good/excellent
> with the Snom 200 (using the latest firmware). Both are probably
> considered higher-end multiline business sip phones by most on this list.
> There are others but I've not attempted to eval those.
> 
> Your customer is likely to drive the decision "if" you let them eval
> a few different models. Since you indicated that you're just getting
> started with *, etc, pure guess is that most business sales are likely
> to require a mixture of multi-line and single-line insturments.
> 
> There has been a fair amount of list traffic relative to how various
> phones support nat, call transfer, music on hold, speaker volume,
> call waiting tones, and other issues. Best guess is that you would
> likely only sell a select set of single-line and multi-line units purely
> from a support perspective, with actual proposals based on specific
> requirements (eg, a location needs nat therefore this model, business
> office with all internal phones likely a different model, another
> business with an unlimited checkbook gets a Cisco ;).
> 
> Rule of thumb...
>  - don't give an executive or check-writer a cheap phone, or one that
>is so lite-weight they pull it around their desk
>  - find a single-line instrument or two you are comfortable supporting
>(seems like the list has suggested at least one vendor's cheap phone
> has a high mortality rate that might be worth striking from your list)
>  - understand where the ata-186 kind of boxes fit (and where they don't
>fit from a real business perpective)
>  - understand the value (or lack thereof) for the phone having an
>internal switch with two RJ45 jacks (and who's phones don't work very
>well with this)
>  - keep a sharp eye on the sip marketplace going forward ;)
>  - understand the value of QoS in switches
>  - find a supplier that delivers & invoices reliably, and will work with
>you on defective units
> 
> If you're looking for opinions on specific models, I'm sure you'll get
> a number of responses from those with favorites.

I haven't a clue, yet

Correct me if I'm wrong, but it sounds like you're firmly in the SIP
camp. While I like the idea of adding extensions by simply plugging a
phone into the network, knowing what some of my potential Asterisk
customers have for data network hardware makes me cringe when I think
about adding 20,30 or 40 Sip phones to the mix :)

So, I was thinking that perhaps going the analog route with a nice ADSI
screen phone might be best for those customers that are either (a)too
cheap to buy cisco's, (b)reluctant to replace network hardware,
(c)afraid of technology...etc. etc.  From what I've pieced together from
googling the list archives it seems like this approach would offer the
customer a solid system today that could grow with SIP phones as
Asterisk and SIP mature a bit more.

Can anyone share what their favorite analog business phones are ?
Is ADSI a good way to go ?  If so, which models are your favorites?

Thanks everyone :)

-Lance
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[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2448 - 10 msgs

[John Todd]

Some I've added, some already existed, some I declined.  :-)

JT


Pounds
Sterling
Screw You (purely optional :-)
Your Call Is Being Connected
Please Wait
Sorry, Your Call Will Be Answered Soon
Your Call Is Important. (That is why we have not yet bothered to 
answer it, instead dancing around the office high on tip-ex.)

And for the girlfriend filter:

Sorry, that number has not been recognized. Please hangup and try again.
or
This number is no longer in operation, Please Hangup.
or
Welcome to the FBI Regional Headquarters, how may I help?
or
This number is busy. Please try again later.
or
The network is busy. Please try again later.
or
Your call could not be connected. Please try again later.
Best,
Ad.
Possibly something to frighten the telesales?
How about: "Thank you for calling. There is nobody available to take 
your call. If you leave a long, loud message, we will return your 
call at the most inconvenient time - most likely at 2am or whenever 
you are asleep. Whilst we have been distracting you, our phone 
system has captured the exact whereabouts of your dialing location. 
We now know where you live. Thank you, and enjoy your day."

[snip]
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[Asterisk-Users] sip and x-lite

[Ing Isianto Istiadi]

Dear all, 
Can you give me the configurations for x-lite and sip in *. 
Thanks



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Re: [Asterisk-Users] More words for Allison

[Steve Underwood]

John Todd wrote:

hurricane
tornado
You missed typhoon!

Regards,
Steve
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Re: [Asterisk-Users] sip and x-lite

[Chandra]

try this...
http://www.fnords.org/~eric/asterisk/

cm

- Original Message - 
From: "Ing Isianto Istiadi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 12, 2004 7:50 AM
Subject: [Asterisk-Users] sip and x-lite


> 
> 
> Dear all, 
> Can you give me the configurations for x-lite and sip in *. 
> Thanks
> 
> 
> 
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> 

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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Chandra]

can u give me the configuration for the firewall??  with the same
configuration i can't even talk or hear... its giving me the RTP Read Error
whenever one picks up the phone.

cm

- Original Message -
From: "Steve" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 12, 2004 6:09 AM
Subject: Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)


> On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote:
> > I'm using Asterisk on a open server (no firewall or NAT) and trying to
> > communicate with a Grandstream BudgeTone 102 SIP phone which is behind
> > NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from
CVS
> > about a week ago.  My problem is that I'm only getting half-duplex
> > communication -- I can hear voice from the Asterisk server but the
server
> > does not understand any voice from me.  From the console "sip debug"
shows
> > that the SIP part is working fine and DTMF via SIP INFO works.
>
>
> I use OpenBSD firewalls with NAT and redirect and it works just as it's
> supposed to.
>
> That's not even half duplex. In half duplex each side Can talk, but only
one
> at a time. It seems to be an error with configuring your firewall. (One
> common error is to only turn on redirect. But you also need to Allow the
> traffic to flow...
>
> --
> Steve
>
> __
> You actually need to constantly be alert
>  and willing to handle things, or life
>will find a way to get you good!
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>


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Re: [Asterisk-Users] possible solution to PRI T100P dropped call issue

[Walker Haddock]

I have a T100P connected to an xspedius T1.  We occasionally have calls disconnect.  I 
just ran zttool and I get Tx/Rx Levels as 0/2.

My zaptel.conf has --> span=1,0,0,esf,b8zs

I can switch out the T100P and see what happens and report back.

Walker

On Sun, Jan 11, 2004 at 07:07:07PM -0700,  John Brown (CV) wrote:
> 
> To recap:
> 
> T100P card wouldn't sync with the telco using line side
> clocking ( span=1,1,0.) 
> 
> Had to use  internal clocking (span=1,0,0...)
> 
> zttool  showed Tx/Rx Levels as  0/ 1
> 
> 
> For the grins of it I replaced the T100P card with
> another newer card from inventory.
> 
> This newer card has the same rev on the ASIC / FPGA
> but doesn't have any of the various jumper headers
> installed on the board.
> 
> Set clocking to line side
> sync=1,1,0..
> 
> Poof, card comes up, span comes up. 
> 
> zttool  now shows  Tx/Rx Levels as  0/ 0
> 
> No other changes have been made to the various
> configs on the system.
> 
> 
> It seems in the end that this was some strange hardware
> failure on the T100P card.
> 
> 
> :
> 
> 
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Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Steven Ringwald]

Steve wrote:

  On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote:
  
  
I'm using Asterisk on a open server (no firewall or NAT) and trying to
communicate with a Grandstream BudgeTone 102 SIP phone which is behind
NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS
about a week ago.  My problem is that I'm only getting half-duplex
communication -- I can hear voice from the Asterisk server but the server
does not understand any voice from me.  From the console "sip debug" shows
that the SIP part is working fine and DTMF via SIP INFO works.

  
  

I use OpenBSD firewalls with NAT and redirect and it works just as it's 
supposed to. 

That's not even half duplex. In half duplex each side Can talk, but only one 
at a time. It seems to be an error with configuring your firewall. (One 
common error is to only turn on redirect. But you also need to Allow the 
traffic to flow...
  


I am having problems similar to Owen's. Just for grins, can you tell me
which ports you opened up? I opened the following:

tcp 4569 192.168.2.212
udp 4569 192.168.2.212
udp 5036 192.168.2.212
udp 5060 192.168.2.212
tcp 2:21000 192.168.2.212
udp 2:21000 192.168.2.212

192.168.2.212 is the IP of the Asterisk box within my firewall. I have
no trouble connecting to it on the local LAN, but if I go remote, it
always wants to connect via a bridge connection to the other BudgeTone
phone.


[ringwald]
fromuser=ringwald
disallow=all
host=dynamic
allow=ulaw
type=friend
username=ringwald
secret=MySecret
canreinvite=no
reinvite=no
nat=yes
dtmfmode=inband ; Choices are inband, rfc2833, or info

Any help that you can provide would be greatly appreciated.

Steve Ringwald

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Owen Kelso]

Thanks for all of your responses.

I confirmed that the phone works perfectly without NAT or through a IPSec
VPN (yeah, I know, same thing).

I've concluded that the Netgear router (FVS318) performing the NAT is
corrupting the outgoing RTP packets.  Traces confirmed that the BudgeTone
is sending them out with a UDP checksum of 0 but the next hop after the
Netgear router they are set to a non-zero value (an incorrect one). 
Asterisk is never even seeing the packets because the kernel is
recognizing them as corrupt and dropping them, hence the recvfrom()
"Resource temporarily unavailable" errors in rtp.c.

I'm going to write Netgear to see what they have to say about it.  If I
make any progress I'll post to the list...thanks again, Owen
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[Asterisk-Users] questions

[Ing Isianto Istiadi]

Dear all,
I have activated call waiting (but since my pstn doesn't support call
waiting, I can't test it with the pstn), and I have 3 fxses. But when I call
the extentions (if that extention is already called), then I got the busy
tones. Is it possible to use call waiting for fxs phone?



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Re: [Asterisk-Users] More words for Allison

[Andrew Thompson]

 Original Message - 
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, January 11, 2004 7:36 PM
Subject: [Asterisk-Users] More words for Allison


> As usual, donations to what will be a ~$110 USD expense would be 
> appreciated, as I am paying for this round out of my pocket.  Please 
> send to paypal address mailto:[EMAIL PROTECTED]


- Original Message - 
From: <[EMAIL PROTECTED]>
To: 
Sent: Sunday, January 11, 2004 11:26 PM
Subject: Receipt for your Payment


> Dear Andrew Thompson,
> 
> This email confirms that you sent $5.00 USD to [EMAIL PROTECTED]
> 

Anyone who added to John's list at least going to match me?

Note: I will not be raised, the rest of it goes to stockpile diapers...


Andrew Thompson http://aktzero.com/


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Re: [Asterisk-Users] questions

[Andrew Thompson]

- Original Message -
From: "Ing Isianto Istiadi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, January 11, 2004 11:30 PM
Subject: [Asterisk-Users] questions


> Dear all,
> I have activated call waiting (but since my pstn doesn't support call
> waiting, I can't test it with the pstn), and I have 3 fxses. But when I
call
> the extentions (if that extention is already called), then I got the busy
> tones. Is it possible to use call waiting for fxs phone?
>

Where did you activate call waiting?

What kind of hardware do you have? Are you even using *? 0.5, or CVS? If
CVS, from when?

More info = less questions


Andrew Thompson http://aktzero.com/

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Re: [Asterisk-Users] More words for Allison

[John Brown (CV)]

We sent $50 USD for the cause

john brown
chagres technologies

On Mon, Jan 12, 2004 at 12:10:09AM -0500, Andrew Thompson wrote:
>  Original Message - 
> From: "John Todd" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, January 11, 2004 7:36 PM
> Subject: [Asterisk-Users] More words for Allison
> 
> 
> > As usual, donations to what will be a ~$110 USD expense would be 
> > appreciated, as I am paying for this round out of my pocket.  Please 
> > send to paypal address mailto:[EMAIL PROTECTED]
> 
> 
> - Original Message - 
> From: <[EMAIL PROTECTED]>
> To: 
> Sent: Sunday, January 11, 2004 11:26 PM
> Subject: Receipt for your Payment
> 
> 
> > Dear Andrew Thompson,
> > 
> > This email confirms that you sent $5.00 USD to [EMAIL PROTECTED]
> > 
> 
> Anyone who added to John's list at least going to match me?
> 
> Note: I will not be raised, the rest of it goes to stockpile diapers...
> 
> 
> Andrew Thompson http://aktzero.com/
> 
> 
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Re: [Asterisk-Users] More words for Allison

[Brian Capouch]

Andrew Thompson wrote:
 Original Message - 
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, January 11, 2004 7:36 PM
Subject: [Asterisk-Users] More words for Allison



As usual, donations to what will be a ~$110 USD expense would be 
appreciated, as I am paying for this round out of my pocket.  Please 
send to paypal address mailto:[EMAIL PROTECTED]


- Original Message - 
From: <[EMAIL PROTECTED]>
To: 
Sent: Sunday, January 11, 2004 11:26 PM
Subject: Receipt for your Payment



Dear Andrew Thompson,

This email confirms that you sent $5.00 USD to [EMAIL PROTECTED]



Anyone who added to John's list at least going to match me?

I didn't add any words to the list, but I did just send $20. . . .

Don't know where that leaves me.

B.
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Re: [Asterisk-Users] Cisco 79xx Ringtones

[Lion Templin]

Siggi Langauf wrote:


I'm after two very specific ringtones for the 79xx's...

If you want some classic office phone ringers:

http://www.leonine.com/~lion/phones.php

These are Merlin rings.  For some, myself included, they're a bit nostalgic.

Lion

--
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= 612-605-3613 x3001 FWD 94117 =
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RE: [Asterisk-Users] More words for Allison

[Scott Bennett]

+$20 from us

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Sunday, January 11, 2004 10:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] More words for Allison

Andrew Thompson wrote:
>  Original Message - 
> From: "John Todd" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, January 11, 2004 7:36 PM
> Subject: [Asterisk-Users] More words for Allison
> 
> 
> 
>>As usual, donations to what will be a ~$110 USD expense would be 
>>appreciated, as I am paying for this round out of my pocket.  Please 
>>send to paypal address mailto:[EMAIL PROTECTED]
> 
> 
> 
> - Original Message - 
> From: <[EMAIL PROTECTED]>
> To: 
> Sent: Sunday, January 11, 2004 11:26 PM
> Subject: Receipt for your Payment
> 
> 
> 
>>Dear Andrew Thompson,
>>
>>This email confirms that you sent $5.00 USD to [EMAIL PROTECTED]
>>
> 
> 
> Anyone who added to John's list at least going to match me?
> 

I didn't add any words to the list, but I did just send $20. . . .

Don't know where that leaves me.

B.
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RE: [Asterisk-Users] More words for Allison

[calvis]

I just sent $20.00 to [EMAIL PROTECTED]

I am new to the list so I don't really know what I am donating to, but the
whole Asterisk program sounds pretty cool and I hope to work myself to
setting up an experimental system to play with it in the near future.

Calvis
Redmond, WA

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Sunday, January 11, 2004 4:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] More words for Allison


Here's the latest batch of words to get shipped out to Allison Smith. 
Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern
time, and I'll add them.

As usual, donations to what will be a ~$110 USD expense would be
appreciated, as I am paying for this round out of my pocket.  Please send to
paypal address "[EMAIL PROTECTED]".  I did not include all possible symbols
on a North American keyboard, as it was getting exhausting and possibly
silly.

calls
abandons
staffing
average Speed of Answer
Sorry, but the user's mailbox can't accept more messages.
Please enter the conference call number for the conference you wish to join.
fortieth
fiftieth
Please enter the conference pin number.
That pin is invalid for this conference.
[The alphabet - a through z, like "ayy", "bee", "cee", etc.] zed space dash
dot comma slash exclamation point ampersand percent at sign [we want a
verbalization of the @ symbol] with plus equals left bracket right bracket
open parenthesis close parenthesis pipe backslash comma period quote greater
than less than chance of cloudy sunny sun turning to rainy rain partly
partially mostly snowy snow scattered patchy wind windy miles per hour
kilometers per hour knots per hour storm warning watch thunderstorm hail
weather lightning fog foggy sleet sleeting clear clearing freezing freeze
hurricane tornado severe later morning afternoon evening late early changing
in the

Alpha
Bravo
Charlie
Delta
Echo
Foxtrot
Golf
Hotel
India
Juliet
Kilo
Lima
Mike
November
Oscar
Papa
Quebec
Romeo
Sierra
Tango
Uniform
Victor
Whiskey
Xray
Yankee
Zulu
Niner

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