.
Any other suggestions?
This is a relatively low volume system. Usually running less than 1 or 2
concurrent calls. Would turning on debugging logs to a file cause a
problem?
Many thanks,
Ejay Hire
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ejay Hire
like
asterisk is seeing a hangup from the analog end.
I have attached my zaptel.conf and zapata.conf. What additional information
can I provide to make this an intelligent question?
Many Thanks,
Ejay Hire
Zapata.conf
; Zapata telephony interface
[trunkgroups]
[channels]
musiconhold=de
Happy Holidays!
Sourceforge provides free hosting for open source projects. That is where I
would put it if I were me.
For licensing.. I use the BSD license for my creations, but version 2 of
the GPL is "stronger" in my opinion.
Good luck,
Ejay Hire
-Original Message-
Fr
Hi.
How do I cause voicemails that land in one mailbox to be delivered to
another?
I.e. I have a incoming call extension that rings all the phones. If it
times out, the caller drops into the general mailbox. I would like messages
dropped in the "general mailbox" to fall into another users mailb
I second the vote for ldapadmin.
You can extend it with custom templates for your asterisk specific
attributes.
-ejay
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of marvin horst
Sent: Tuesday, November 28, 2006 6:52 AM
To: Asterisk Users Mailing List - Non-Commercial
Yes! Rxgain and Txgain values still have to be set appropriately on a pri.
Follow up quesstion, does fxotune work and do anything useful on a pri?
-ejay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Friday, November 03, 2006 1:52 PM
To: Aste
Hello.
In extensions.conf; in the context that is dialed by your internal
extensions, add this line.
include=>parkedcalls
This will include the extensions created by the extensions module, and
create your extensions 9006-9009.
Good luck,
Ejay Hire
F
This is incorrect. The data is still packetized and passed through IP which
provides the same echo cancellation and distortion issues as a call that
passed through an FXO/FXS card.
Ejay Hire
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
Sent
o web interface. I didn't like any of the default
ringers on the Snom phones, but the users really liked the LED call
appearance lights compared to the 7960 LCD. I have no complaints about the
Linksys phones.
Ejay Hire
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTE
I have a couple of useful bits that could be tacked on to this..
1. Telcos required to offer the ability to set the outbound caller id.
2. Telcos required to offer the ability to write to the CNAM database, in
near-real or short time.
3. Telcos required to forward the ANI you provide to the 911 wi
I think this went to the wrong list. This is asterisk
support, not samba.
Having said that, I'd be happy to take a look if you want
me to ssh in. I have 6 samba boxes in as many states.
Ejay Hire
[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hello.
I had the same problem, and was able to fix it as
follows.
1. Run fxotune
2. Call your XO rep and get a milliwatt test line
number
3. set the gain in the zaptel.conf incoming with the
milliwatt test line
4. loop a call through the pbx and set the outgoing
gain.
With these se
I concur. We spent a lot of time trying to get USLEC
to take caller id NAME information from us to no avail. They recieved and
dropped it.
-ejay
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Thursday, October 12, 2006 2:59
PMTo: asterisk-users@l
No, not really. It makes the dialplan complex, to accommodate all it's
options. The asterisk sample configs (make sample-configs?) are a much
better place to learn from, once you understand the basic concepts.
Personally, I really could have used a 2 page asterisk quick reference sheet
with a sa
I couldn't make it do what I wanted, ring 15 phones for 15 seconds, and then
go into a voicemail box. The dial plan looked right in the website, but the
calls never went to the mailbox.
-ejay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Ramsey
S
-= 1967 extensions (2838 priorities) in 285 contexts. =-
Shared services PBX with a dozen or so customers.
-ejay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy
Sent: Tuesday, October 10, 2006 3:17 PM
To: asterisk-users@lists.digium.com
Subj
Hello. It would appear that the voicemail module is
not loaded. If this is a new install, did you install the sample config
files? Specifically voicemail.conf.
-Ejay
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of raviprakash
sunkaraSent: Tuesday, October 10, 2006 6:07
Hi. A "cross-over" cable won't work, the isdn network provides signalling
and adressing functions.
When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg,
around $1k used from ebay.
-ejay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf O
t, not twisted) was run 35 feet around the edges of
the room, under space heaters, from the NID to the closet, butt-spliced and
hooked to the 66block, and then to the PBX. Replacing this cable with a
category 5 cable run under the house (I didn't have any Cat3) fixed it.
Ejay Hire
-
Hello all, and good morning
In my /var/log/asterisk directory I have 492,018 log files, most of which
are empty.
event_log.XXX queue_log.XXX messages.XXX where XXX is an integer.
I removed them all and restarted asterisk a few days ago, but they came
back.
This morning I turned off event and que
I use the Linksys SPA-942-na's with great success.
For a good POE switch, the linksys swr224p fits the bill too.
-ejay
bilal ghayyad wrote:
Hi List;
I am looking to use an good IP Phone working with
Asterisk and work based on PoE (so it takes the power
via the ethernet cable, no need to conne
The procedure you want is called the nimdy/nindy procedure, and the
authoratative source is the "Book of nindy." If I remember right, you
can find it and the current software releases on ftp://ftp.ascend.com
I used these units at an ISP for a long time, and they are okay. If
they are install
Traditional overhead paging systems are a little more
complicated than they first appear. It's not just speakers
and a centralized amplifier. They would have too much cable
loss if done that way. Instead, they use a centralized
power source, and amplifiers at each speaker unit.. The one
I just
Hello.
To answer your question, root is a restricted account. It
is too powerful to trust a telnet connection. So, you
telnet (preferably SSH) in as a normal user, and then type
`su -` and enter the root password. Su (short for
SuperUser?) allows you to become root. The - specifies to
load all
I second this. Our Primary (not the one we bought) system
is radiator, with mysql for the backend. I've never found
something it couldn't do.
-e
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
Of
> Saliel Figueira Filho
> Sent: Tuesday, August 03,
Thanks for the vote of confidence guys. We just bought an
ISP that uses rodopi exclusively for Accounting and Billing.
...sigh...
-e
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
Of
> [EMAIL PROTECTED]
> Sent: Tuesday, August 03, 2004 12:15 PM
Hi all.
We've been working on this with PRI's from USLEC connecting
to a Cisco AS5300 as the gateway. We are sending the CNAM
information, but they can't read it dynamically. We are
stuck (for now) with manually sending in database update
requests.
Having said all that, we're looking at pulling
, so what do we know?
:) ). If you don't get a yellow light with the BNC's
hanging then the smartjack probably has an internal loopback
set and you need to give your telco a call to clear it.
Good Luck,
Ejay Hire
ISDN-Net Network Engineer
> -Original Message-
> From: [EMAIL PR
Hi. Asterisk doesn't currently support fax pass through as
far as I know. W/o fax pass through the faxes don't work
well at all.
-e
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
Of
> Lach Dunlop
> Sent: Thursday, March 04, 2004 11:49 AM
> To: [
Hmm. Red button FACTRESET# works for me on 2.16 and 3.0
doesn't work for you?
Ejay Hire
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
Of Matt
> Sent: Thursday, March 04, 2004 8:30 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [A
The original NBX100 phones spoke a proprietary voice-over-l2
ethernet protocol, but would upgrade to ip connectivity with
a liscense key on the NBX PBX box. There was an optional
software package that would let the NBX talk to an h323
gateway but it ran on nt and was rather klunky.
These origi
Hi, I can answer part of the Caller ID question.
> Also, at least in the testing I've done, the text portion
of
> the CLID string is ignored by the telco. They only look
at the number and
> generate the text based on what is in their database. IE;
If I tell my
> asterisk server to set my call
there is a bug in the v3.0 code for the
186's. If you use the TOS bit's to mark SIP for QOS,
downgrade back to 2.1.6. In 3.0, the ata sets TOS to 0x0,
and ignores the TOS: configuration field.
Hope that helps,
Ejay Hire
ISDN-Net Network Engineer
...Providing VoIP services to Tenne
I have this problem intermittently, and doing an asterisk
-r showed "too many retries." hunting around with
ethereal found a bad hub.
-e
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
Of
> Tomica Crnek
> Sent: Tuesday, February 10, 2004 9:23
Hi.
Low Temp Hot glue is what I use on my robots.
Stay away from silicone (conductive) and rtv (peels traces
off cheap pcb's)
-Ejay
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
Of
> Greg Kedrovsky
> Sent: Wednesday, February 04, 2004 8:18 AM
If you've got a Linux workstation, www.vovida.org offers
sipset, a free softphone.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
Of Tim Sailer
> Sent: Tuesday, February 03, 2004 4:10 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] voip phones
When troubleshooting, I'll often
"tcpdump -s 0 -w filename.cap -p host (ipaddressofphone)"
To capture the entire contents of all packets from or to
ipaddressofphone non-promiscuously to filename.cap. Since
my workstation is Win*, I have to sz to move the capture
over to my desktop and then open
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