Displaying 20 results from an estimated 47 matches for "mschulte".
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schulte
2004 Dec 15
2
chan_sccp compile problem w/ CVS head?
Any ideas? I edited the Makefile as instructed, ty.
Now compiling .... sccp_channel.c 279 lines
sccp_channel.c: In function `sccp_channel_send_callinfo':
sccp_channel.c:48: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no
2004 Aug 24
7
SMP Performance
We're looking at implementing Asterisk in our department in the near
future, we're looking at anywhere from 15-25 extensions. The machine we
were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/
1GB of ram. I've heard bad things about running Asterisk on SMP
machines? Would we be running into any performance issues with this
machine?
Tim Jackson
Network Engineer
2005 Jan 12
7
Operator Panels?
Ok, we're trying to use Asterisk as a PBX in our office. Our original
plan was to use a Cisco 7960 with a 7914 attached. Short story is, no
one updated chan_sccp in a long time and the 7914 is questionable at
best anyway from what I've heard. We couldn't ever get chan_sccp to
compile, I went to an older version of Asterisk and that broke some of
our SIP devices. We tried using a couple
2004 Aug 20
6
Sipura endpoints
Anyone have experience with Sipura's? Anyone know if they offer a
warranty? Would like opinions on these, good or flame.
We bought *one* to test with and it died, can't even get a
response from Sipura "support". Could anyone recommend another device to
replace these? Prefer 1 or 2 port design.
Ty :-)
2005 Mar 15
6
Realtime config
Having problems getting realtime working, I'm trying to use odbc for all
of this. I've got Fedora 3 and have been fighting with odbc for a day
now. I think I got it working correctly, however I can't seem to get the
realtime portion working. In asterisk 'odbc show' shows it connected, I
see it on my (odbc) mysql server connected and all, it connects and just
idles. So, without
2004 Dec 30
6
Nagios and Asterisk
Does anyone have some decent Nagios scripts out there that do more than
monitor the proc itself? Rather than reinvite the wheel, figured I'd
ask. I already saw the one on the wiki.
Matt
2005 Mar 24
14
Realtime mysql problem?
All, I get this whenever trying to dial to a peer when the peer
registered to another server. I'm basically trying to use realtime to
check for the peer and dial it.
Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f",
"IAX2/brak-test/107") in new stack
Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM
sip_users WHERE name =
2004 Dec 16
0
FW: Cisco 7960 (SIP) hold problems
...d problems
Hi Matt,
Seems you have gone one step further than me,
Can you possibly let em see your sip configs for the cisco 7960 and also
the
configs in the phone. I am having trouble getting it to talk to asterisk
Thanks
Paul
----- Original Message -----
From: "Matt Schulte" <mschulte@netlogic.net>
To: <asterisk-users@lists.digium.com>
Sent: Thursday, December 16, 2004 3:01 PM
Subject: [Asterisk-Users] Cisco 7960 (SIP) hold problems
> Has anyone had problems with using hold on a 7960 SIP firmware? The
> problem is when the 7960 puts a call on hold and you take i...
2005 Mar 29
8
Dell 1750 & TDM400P - Power
Has anyone come up with a way to get power to a TDM400P card installed
in a Dell PowerEdge 1750?
Thanks,
Adam
The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this
2005 Feb 03
5
Cisco 7960G phone crashes during SIP upgrade
Hello,
I've recently received a Cisco 7960G phone with the factory default
SCCP firmware on it.
As we're using SIP on our network, the first things i've done was to
upgrade but unfortunately the phone just restarted. By looking on the
TFTP logs and tcpump output, i've seen that the phone crashed and
restarted just after downloading the OS79XX.TXT file, without
requesting the image
2004 Aug 10
0
T400P modprobe/ztcfg failure
Another nibbage question:
modprobe tor2 -v
install /sbin/modprobe --ignore-install tor2 && /sbin/ztcfg
insmod /lib/modules/2.6.5-1.358smp/misc/tor2.ko
ZT_SPANCONFIG failed on span 1: No such device or address (6)
FATAL: Error running install command for tor2
dmesg states that the module is registered however. Am I doing something
seriously wrong?
I have my zaptel.conf configured as
2004 Aug 17
1
Faxing over ulaw
Are there any considerations to take in account when faxing from analog
to SIP using ULAW? The problem we're having is faxes are only making it
halfway, getting cut off. Neither fax machine seems to report an error.
Pretty diagram:
FXS --> SIP --> PSTN Provider --> FAX
^ULAW
2004 Aug 27
0
OT re: sip change?
Kind of off topic but I know CVS is the "prefered" way of upgrading,
however are there such things as "stable" CVS upgrades? It seems a lot
of the CVS's have a lot of devel bugs in this that I would be scared to
put even near production. Just IMHO. :-)
Matt
-----Original Message-----
From: Rich Adamson [mailto:radamson@routers.com]
Sent: Friday, August 27, 2004 9:15 AM
2004 Oct 22
1
IAXy echo avoidance/cancellation
Ok, I searched the lists and found no definitive answer. I'm assuming
the IAXy has some primitive form of echo cancel, is there anyway to
adjust this? Or any ideas on what to do instead. Here's the setup, this
will not be a typical setup for our company however, well whatever.
Anyway it looks like this:
IAXy --> Netgear router --> Charter --> Level3 --> Asterisk --> VOIP or
2004 Dec 06
0
Passing SIP digest auth to dialplan
This maybe a simple question however I can't find a way to do this, I'm
wanting to EITHER:
Pass SIP digest authentication via dialplan (extensions.conf)
OR
Make Asterisk realize that the incoming peer in sip.conf doesn't have to
authenticate.
The reason I have this is because I'm connecting through SER and using
aliases. The reason I'm using aliases is because we're
2004 Dec 08
1
ftmp header
All,
We are using a SIP provider that is expecting 0-15 response for
fmtp. Our CVS Head asterisk server is sending 0-16, I looked up an rfc
and it stated:
RTP Payloads for Telephone Signal Events
RFC 2833
Henning Schulzrinne, Scott Petrack.
May 2000
Implementation notes:
* Implementations can support events 0 through 15 (DTMF) by
simply ignoring the packets, but MUST
2004 Dec 09
0
Can asterisk accept cleartext auth (uri user:pass) via SIP
Does anyone know if Asterisk can accept cleartext auth (SIP), as in it
recv's a call destined to:
1234:blah@har.har.com
The problem I'm having is simply for faxing, normal calls come in as
g729 and of course we need ULAW for faxes.
sip.conf snippet
[sipfarm]
insecure=very
host=blahblah.netlogic.net
type=peer
context=sip-out
username=+18165551212
secret=blah
canreinvite=no
disallow=all
2004 Dec 15
2
Cisco 7960 SIP + 7914
I found a few mentions of the 7914 being used with Asterisk, these all
covered SCCP/skinny though. Does anyone know if the 7914 can even be
used with SIP? If so, any pointers? Is it a services thing? Anyone get
the operator (line/extension status) to work with it. Thanks for the
help, Cisco doesn't even mention ANYTHING about SIP + the 7914.
Matt
2004 Dec 22
0
TE410P to a Rhino CB-24 channel bank
Has anyone had any success with the Rhino CB-24? I can't get mine to
work, I tried all the obvious settings. The cb-24 gets stuck at init ESF
framing, as if it's not seeing the t1 card at all. It does get a t1
carrier (detecting voltage??)
Help!
Thanks..
Everything appears to look good on the server, I did:
[zaptel.conf]
span=1,0,0,esf,b8zs
fxogs=1-24
[zapata.conf]
signalling=fxo_gs
2005 Jan 03
2
SIP Jitter buffer(control?)
I'm assuming asterisk does not have a SIP jitter buffer in place? Any
ideas on how to help with this going over a data T1 where VoIP is shared
with regular traffic? Problem is when people are downloading the voice
is jittery, even lossy.
Matt