Try deleting the line
pritrustusercid=yes
in zapata.conf
maxx
- Original Message -
From: Bastian Schern [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 20, 2004 8:34 PM
Subject: [Asterisk-Users] Incoming MSN via ZapHFC - to SIP
Hi there,
I've got a small problem with
Hi,
you can also load the wct4xxp driver with
modprobe wct4xxp t1e1override=1
Does the same job, remotely.
Kind regards,
Martin List-Petersen
On Fri, 2004-08-20 at 22:11, Arnaud Pignard wrote:
Ok, i haven't get in hand the card and make remote hardware install.
It's certainely the
Hi,
I realize that this topic has been hashed and rehashed a lot of times on
this list, but none of the information I was able to find quite fits my
case.
I seem to have disconnect supervision, namely the light on the handset
turns off for ~2 seconds when the other side hangs up, and I have
I just added a second X100P card to my * server, altough it seems to
be working * seems to be ignoring it:
zaptel.conf:
-
fxsks=1-2
loadzone=us
defaultzone=us
zapata.conf:
--
context=inbound-analog
signalling=fxs_ks
group=1
channel = 1
I had issues with two sipuras after I upgraded the firmware, after
emails and troubleshooting they sent me an RMA. They fixed one and
replaced the other.
On Fri, 20 Aug 2004 07:43:56 -0500, Matt Schulte [EMAIL PROTECTED] wrote:
Anyone have experience with Sipura's? Anyone know if they offer a
Thanks to all those who responded to my post.
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Woods
Sent: 21 August 2004 01:05
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] telnet and Root
Chris Shaw wrote:
If you really want to be able
Hello everyone,
I've some stange problems with my AVM FritzBox Fon
(http://www.avm.de/de/Presse/Informationen/2004/2004_06_22.php3)
and Asterisk.
I want to build up an internal SIP Network for testing
purposes.
I can phone with my Grandstream BT-101 to other BT-101
or SIP-Software, but not to
I just deployed * on my home system last Sunday. 2x since then the Zap
hardware seems to have malfunctioned on some way.
One time it would just screech out one FXS, even though it would ring. The
other time * would bridge to my FXO but it never got out on the line. I have
a new TDM400 with 3
On Sat, 2004-08-21 at 06:42, Michael George wrote:
I just deployed * on my home system last Sunday. 2x since then the Zap
hardware seems to have malfunctioned on some way.
One time it would just screech out one FXS, even though it would ring. The
other time * would bridge to my FXO but it
hello everybody ,
i got a test setup with
lanphone 101 connected to *
and a welltech fxo 3802 also to *
the extensions are configure so that i can dial from the lanphone to
the fxo.
although once on the FXO and having the dialtone, no of my dtmf
dialing is being processed on
This doesn't answer your question completely, but I have noticed that
inserting and removing the kernal modules doesn't work all that well and
that rebooting is a better answer at that point.
Have you verified that you are not IRQ sharing? * really doesn't like that,
even though other
Hi,
Sorry, in my last mail I wrote wcfxs instead of what I actually used, wcfxo. I just got two digium x100p clones and installed asterisk on fedora
core 2 which took some tweaking. After getting asterisk up I installed
the zaptel stuff - then modprobed zaptel, wcfxo,
which worked fine.
show channels only shows active calls. So if no calls are active, there are
no channels active to be displayed.
Try 'zap show channels' first.
Then 'asterisk -c' and call one of the cards and see what pops up on the
screen. It should give you clues as to the problem.
Lyle
- Original
Although it shouldn't make a difference, try:
channel = 1-2
As well, did you run ztcfg after you installed the new card? I've found
sometimes I've had run ztcfg a couple times before Asterisk would kick
in and recognize a new card.
On Sat, 2004-08-21 at 02:49 -0500, spectro wrote:
I just added
On Fri, 20 Aug 2004, Roy Sigurd Karlsbakk wrote:
Is it possible to find out what source/destination operator you're
connected to from the ISDN signalling? Number lookups no-longer work
since number porting begun, and access to the national database for
these numbers cost too much :(
That
Chris Shaw wrote:
These phones can be
found for as low as $65 USD and have EXCELLENT voice quality and can
even work behind NAT...
I disagree. I think the Grandstreams sound like crap (at least on the
remote end, I've never used one. Just heard people on one.). Lots of bad
echo.
--
Daniel
Hi all,
quick question...i am using autocreatepeer to get asterisk to work with SER
without having to specify each UA in sip.conf and in ser separately.
2 questions:
1. obviously this is not very secure. assuming i block incoming requests on
the port asterisk is running SIP on (excluding requests
I'm trying to somewhat reduce the security risk of Asterisk, by loading less
modules. In my installation I use SIP and IAX2 for incoming calls,
and that's it. No voicemail, no call parking, it just plays back voice
clips.
I can remove /etc/asterisk/modules.conf modules one by one:
Hi Navit,
My company is developing asterisk compatible HW SW.
If you would like to contact me, we might be able to work on your wishlist.
Eran Gal - Xorcom
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Navnit Chachan
Sent: Tuesday,
Simone Ricci [EMAIL PROTECTED] wrote:
Anyone knows if there's a way to select the bitrate of those codecs
supporting multiple bitrates (eg. g.726)? I've tried searching and
googling a lot, but without useful results...
Asterisk's G.726 codec only supports the 32-bit mode. You can't select
on 8/21/04 13:37, Daniel Jimenez at [EMAIL PROTECTED] wrote:
I disagree. I think the Grandstreams sound like crap (at least on the
remote end, I've never used one. Just heard people on one.). Lots of bad
echo.
I would tend to agree with Daniel. and the workmanship on them are
terrible.
Hi All,
A friend has asked me to get a new Zultys 4x5 working with his *
server. I've been over the web gui but don't see where all the
registration info gets entered in the phone. Is there someone on-list
who has Zultys phones and can advise me as to their setup? The
manufacturers docs are
Dnia roda, 18 sierpnia 2004 16:56, Walt Reed napisa:
OK, I'm going nuts here trying to correctly identify null values,
specifically when callerID info is not available.
FYI, I'm running Asterisk CVS-HEAD-08/17/04-13:08:53, and Bison 1.875a
(debian Sid).
A snippit of my dialplan looks like
I've got a Dual Xeon system Digium Quad T1 and an
Idle T1 circuit that I can experiment with.I've been wanting to use Redhat
with software Raid 1 on an Asterisk server.
Has anyone had any experience with software raid
and Asterisk? Also, if the software raid doesn't play, any
I used software raids and want to get away from
them. I really really like Microlite's BackupEdge tape backup
software. BackupEdge does not work with a software raid, only a hardware
raid.
The second kicker was that the Promise (or any
other) hardware IDE raids are considered software raid
started messing around with the Monitor cmd for call recording but noticed
that the cmd was 'injecting' a small noise every 8 sec or so. looked
through the wiki for a flag setting but the 'm' for using soxmix post call
is the only one noted. trying to build a call logger that just sits next to
Ed Devine wrote:
I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I
can experiment with. I've been wanting to use Redhat with software Raid
1 on an Asterisk server.
Has anyone had any experience with software raid and Asterisk? Also, if
the software raid doesn't play,
On Sat, 21 Aug 2004, Ed Devine wrote:
I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that
I can experiment with. I've been wanting to use Redhat with software
Raid 1 on an Asterisk server.
Has anyone had any experience with software raid and Asterisk? Also, if
the
On Sat, 21 Aug 2004, Lyle Giese wrote:
I used software raids and want to get away from them. I really really
like Microlite's BackupEdge tape backup software. BackupEdge does not
work with a software raid, only a hardware raid.
The second kicker was that the Promise (or any other)
Depending on the application the Grandstream is decent but for
prolonged use I've found it's better to not pinch the pennies and go
with something a bit more expensive but with less problems. For a
simple SOHO deployment I'm just putting a Sipura SPA 3000 on their
cordless base station
Lyle Giese [EMAIL PROTECTED] writes:
I used software raids and want to get away from them. I really really like
Microlite's BackupEdge tape backup software. BackupEdge does not work with a
software raid, only a hardware raid.
I have worked with software raid for several years, and to date
Hi,
I've sucessfully registered windows messenger to Asterisk and can talk to
other extensions cause they appear as being online.
But can I place a call to arbitrary SIP address (if I enter it to contacts
it appears as offline and don't get possibility to start voice or video
conversation) ?
Hi,
I'd like to have local extensions accessible through SIP uri (like
[EMAIL PROTECTED]), but at the same time for convenince to be also extension
with number (like 100) for more convenient dialing thought softphones that
support only numeric keys.
Can this be done ? Since I'm newbie, I'd
Hi,
I'm interested in ActXPhone active x web control (mentioned on
voip-info.org) but link is dead.
Has anyone cached files and is willing to share them ?
Thanks,
Robert.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Here's our setup:
8 analog lines all terminate in an Adtran TA750 channel bank we
bought. From there a crossover cable connects to the T100P card in our
Asterisk server.
(sorry, I'm more of a perl/java guy who got thrown into this, my phone
knowledge is pretty weak). I'm not too sure what
I understand that putting monitor-format in the queues.conf file will
start monitor recording of an active queue call. Is there a way to
automatically do the post call processing like the 'm' option like when
specifying the use of the monitor command?
Kevin
Monitor(wav,${CALLFILENAME},m)
Paul Concepcion wrote:
8 analog lines all terminate in an Adtran TA750 channel bank we
bought. From there a crossover cable connects to the T100P card in our
Asterisk server.
With this arrangement you cannot receive DNIS. There is no way for the
telco to transmit the dialed number to you.
We
On Saturday 21 August 2004 14:47, Ed Devine wrote:
Has anyone had any experience with software raid and Asterisk? Also, if the
software raid doesn't play, any recommendations for a hardware based IDE
Raid controller and suggestions on best practices for setting up the disk
partitions (2 X 40
On Saturday 21 August 2004 15:17, Lyle Giese wrote:
My experience with IDE in raid arrays is less than stellar and will be
trashing them as I rebuild servers. I have had several instances where one
drive fails and the entire array falls over as the kernel struggles to
recover from the loss of
Thanks for the information. Sorry about tossing that Caller ID tidbit,
a separate train of thought appeared while I was typing that last
reply.
On Sat, 21 Aug 2004 14:23:46 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
Paul Concepcion wrote:
8 analog lines all terminate in an Adtran TA750
well, that's our setup (8 analog lines - channel bank - t100P), so
it looks like DNIS is out of the question. We do have 8 phone numbers
though. Could we have a 1-800 number direct to each of those, then do
what you suggested with contexts? What would happen if two people
dialed 1-800-a if 1-800-a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Saturday 21 August 2004 05:22 pm, Andrew Kohlsmith wrote:
On Saturday 21 August 2004 14:47, Ed Devine wrote:
Has anyone had any experience with software raid and Asterisk? Also, if
the software raid doesn't play, any recommendations for a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 20 August 2004 02:25 pm, Eric Wieling wrote:
On Fri, 2004-08-20 at 12:52, Steve Szmidt wrote:
Does ANYONE have music on hold working across IAX2? Google does not
return anything on the subject. Except I did see on the release notes for
I cant get zaptel to make. I
get this error: make: *** [linux26]
Error 1. The previous
line is: Link /usr/src/linux-2.6 to
your kernel sources first! I am running Fedora Core2 Asterisk
compiles fine. I am using my SIP phones. I would like to get my TDM400p working.
I've got a nearly identical system running on Slackware 9.1 with a
Megaraid controller.
Works better than expected!
Niles
I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I
can experiment with. I've been wanting to use Redhat with software Raid 1
on an Asterisk server.
Look in /usr/src. You should see a directory
something similar to linux-2.6.1-1[as an
example].
If you DON'T have a directory (or link to a directory)
named linux-2.6, you should create one using the 'ln' command. In the case
mentioned above, the command would be:
ln -s
Hi
all,
Does anyone know if
it's possible to continue talking to one member of an initiated conference call
on Cisco 79xx ? In other words- disconnect one of the
parties.
Thanks.
Assaf
I have working SIP numbers with broadvoice, and just added a DID from
http://connect.voicepulse.com/ . The calls answer, but DTMF is not
recognized.
With iax2 debug active pressing DTMF does nothing. Zilch. Zero.
A friend tried a different IAX2 connection, and got the same results.
I see the
Ok I am past the compile. Now when I try
to modprobe I get FATAL: Error inserting zaptel
(/lib/modules/2.6.5-1.358/misc/zaptel.ko): Invalid module format
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Blakely
Sent: Saturday, August 21,
Brian,
Im new to this list, and ran across this post about a Uniden UIP200. Since its
been a few months now, I was wondering how it's turned out so far.
I am also looking to implement * for a PBX, and am in search of a good hardware
phone, that *hopefully* doesn't break the bank (HINT HINT ::
Michael Graves wrote:
A friend has asked me to get a new Zultys 4x5 working with his *
server. I've been over the web gui but don't see where all the
registration info gets entered in the phone. Is there someone on-list
who has Zultys phones and can advise me as to their setup? The
manufacturers
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