I have a TDM2400P with hardware echo cancel. We seem to have static on
some calls but not others and the receive audio appears 'choppy'.
Transmit side works fine and does not have any audio problems. I had to
turn up the RX gain to 18 or the receive audio volume is too low.
Can anyone shed
I am having issues with a TDM2400P. It appears when the ZAP channel dials
out, it randomly chops the first digit off of the number. I have tried
relaxdtmf=yes, turning up and down the txgain, turned off and on the echo
cancellation, generated new zaptel (with updated spinlock.h)...
I am at a
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500
Tim Jackson wrote:
Copied your sip.conf and changed the settings and I'm getting the exact
same error. I'm also running 1.3.4 of the SIP app for the IP500.
Someone has already pointed out that you might have ran
: [Asterisk-Users] Polycom IP500
How about your zapata.conf and zaptel.conf files? Were they updated for
the new card?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 12:09 AM
To: asterisk-users
= parkedcalls
include = trunklocal
include = trunktollfree
include = trunkld
include = trunkint
include = sip
YOURS
sip.conf:
[101]
type=friend
callerid=Tim Jackson - Home 101
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all
May
nat=yes? And if NAT is in use, what
is your network infrastructure?
Also, what is Polycom's SIP firmware version? (mine is 1.3.4 from
October 2004).
And of course: what is Asterisk and zaptel version? What is your
zapata.conf (just curious)?
Andrei
Tim Jackson wrote:
Earlier tonight I moved
TDM400's use the wcfxs module to drive both FXO and FXS ports on them.
I have an IBM xSeries 305 (1U P4 2.4ghz 1gb of RAM) server and I just
picked up a TDM04B today, and I am getting the exact same problem.
When I make calls to/from the TDM04B card I get this really really
staticky sound.
I dug around and found my newest UpdateXpress cd from IBM and ran it on
this box and updated the BIOS and my problem went away. *shrugs*
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Swan
Sent: Wednesday, January 05, 2005 7:35 PM
To:
=Tim Jackson - Home 101
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all
OR
[101]
type=peer
callerid=Tim Jackson 101
secret=itsasekret
host=dynamic
dtmfmode=rfc2833
nat=yes
mailbox=101
canreinvite=no
context=default
disallow=all
allow=ulaw
Has anyone gotten the Swissvoice IP110 to work w/ * ?
-Tim
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Walker
Sent: Friday, December 10, 2004 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re[2]: [Asterisk-Users]
, in my opinion. Are they so
big, they do not even care?
Sincerely,
Andrei
Tim Jackson wrote:
Any idea if 1.34 fixes the problems with the phones being up for long
periods of time and weird call problems (I cannot hear remote caller,
but they can hear me) ?
-Tim
-Original Message-
From
the default gateway on all of my phones to a router that knows about all
of my subnets, instead of my internet gateway.
Ty
-Original Message-
From: Tim Jackson [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 02, 2004 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes
in sip.conf in order to get the phone to work.
Sincerely,
Andrei
Tim Jackson wrote:
Theres no NAT going on here. Just 1 router in between, phones reside on
10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT
Any idea if 1.34 fixes the problems with the phones being up for long
periods of time and weird call problems (I cannot hear remote caller,
but they can hear me) ?
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Wednesday, December
Ive had the same problem. I posted
to the list earlier about the problem, and from what I can tell, its a Polycom
issue (not a network issue as stated in the other post). It happens after the
phones have been on for about 2-3 days from what I can tell. My solution to
this was to use a
I havent found any recent information on this, but
can Asterisk act as a MGCP UserAgent?
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile
___
Asterisk-Users
I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent
only.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels
- Original Message -
From: Tim Jackson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 23, 2004 2:48 PM
Subject
The ILEC here is using VocalData for doing VoIP Centrex systems etc. My
sales engineer preached SIP to me when he was talking about it, but I
actually got a hold of an engineer today, and he told me they are using
MGCP only for now. He seemed really interested in *, they are bringing
out some demo
. No errors on the console, using g.711u. Any ideas?
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman
I have a Plantronics M12 amplifier and a bunch of interchangeable
headsets. I haven't found anything that this won't work on yet.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Sunday, November 21, 2004 5:45 PM
To: Shaun Ewing;
canreinvite=no ?
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tracy R
Reed
Sent: Friday, November 19, 2004 6:56 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Routing between different
is
available to answer at this time
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
Its working here, some issues tho. All outbound calls have no CID.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Komito
Sent: Saturday, November 13, 2004 1:16 PM
To: Doug Shubert
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
It's no issue to use more than one nic.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, November 11, 2004 7:29 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Multiple NIC's on * box?
Can * support a box with
I've applied the patch (after scanning over the file). No issues with *.
BV still works, too.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan
Wilkins
Sent: Wednesday, November 10, 2004 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial
Nov 10 14:52:56 NOTICE[20579]: chan_sip.c:4041 sip_reregister:--
Re-registrm
-- Responding to challenge, registration to domain/host name
sip.broadvoicem
Nov 10 14:52:56 NOTICE[20579]: chan_sip.c:6821 handle_response: Outbound
Regist)
I got this after applying the patch. I'm guessing
From my experience the Tyan Tiger MPX is a great board. I've never used
it with *, but I have been using it as a high volume samba server for
over a year and its never even hicupped.
16:24:30 up 197 days, 20:45, 2 users, load average: 0.94, 0.92, 0.89
-Tim
-Original Message-
From:
Instead of Wait use Background and play silence:
exten = s,3,Background(custom/menu)
exten = s,4,Background(loligo/silence/10)
-Tim
On Sat, 2004-11-06 at 13:03 -0700, Damon Estep wrote:
My incoming auto attendant plays a prompt, waits for 5 seconds, and the
plays the prompt again giving the
Not to say that all Govt's are like this, but we employ a LOT of open source. It comes
down to a money issue. But besides that, we still use Windows, and YES it does have
its place. I don't think that this thread belongs on this list.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
I can't speak for the US Govt, but I speak for a local government. We use open source
everywhere. My departments PBX is Asterisk, Fileservers, Webservers, we use Linux
everywhere. In my dealings with the State of TX they are adopting open source for some
very mission critical applications. If
'Zap/pseudo-874077465'
== Spawn extension (default, 800, 3) exited non-zero on 'SIP/101-74c0'
[meetme-int]
exten = 800,1,Answer
exten = 800,2,Wait(1)
exten = 800,3,MeetMe(|Dx)
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile
, 2004 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MeetMe
Tim Jackson wrote:
I've got a problem with MeetMe. I dial the extension that dynamically
creates the new conf, but it just hangs up on me after telling me I'm
the only person
Use something like ProFTPD or something that is supported under their
manual (These are better FTP daemons anyway).
The default username/pass is PlcmSpIp btw.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob Knight
Sent: Tuesday, October 26,
We just got setup with Broadvoice
yesterday for LD. This isnt something I REALLY need (No local numbers
avail so we just got a Houston number),
but Im just curious. I can make outbound calls to Broadvoice
and they work great, but I cant do inbound. I have bvs voicemail turned off so all I
I'm having the same issue, and I'm not behind NAT.
Maybe this is a BV issue?
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry
Evans
Sent: Saturday, October 23, 2004 4:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk and
I'm using 3 X100Ps with no problems in an old IBM machine.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent
Franks
Sent: Tuesday, October 19, 2004 11:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sipura or X100P Option
Hello,
Our
Anybody ever tried doing voice over Sprint/Verizon 1xRTT
cell service? 10-15KB/sec downloads/uploads with 400-1200MS latency is what I
usually see on my service.
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile
The horse has been dead for a long while. Please stop beating it.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Saturday, October 16, 2004 4:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Best bang for the buck out there are Polycom SoundPoint IP phones. We
use IP500s.
Pros:
Pricetag (Cheaper than Cisco ~$180/phone)
Quality (Built really well)
Features (3 lines, XML Directory, DND, MWI, etc etc)
Fairly straight-forward provisioning (Once you get the hang of it)
Very very very
Break down and learn Debian, its more GNU than you. You'll never go
back. Also worth mentioning is Ubuntu Linux which is a Debian offshoot.
http://www.ubuntulinux.org/
http://www.debian.org
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James H.
Ditto. I'll provide a mirror as well.
-Tim
-Original Message-
From: William Suffill [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 23, 2004 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.0 released
If anyone who got
Got the 1.0 tarball up, anything else that needs to be mirrored?
http://mirrors.angelinacounty.net/asterisk/
ftp://mirrors.angelinacounty.net/asterisk/
-Tim
-Original Message-
From: Benjamin on Asterisk Mailing Lists
[mailto:[EMAIL PROTECTED]
Sent: Thursday, September 23, 2004 10:19
BTW, That machine is on 100mbit. Should be able to rape it pretty bad,
as long as you don't go over my 1600gigs/month.
-Tim
-Original Message-
From: Tim Jackson
Sent: Thursday, September 23, 2004 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
equipment, but their idea is going to
be costly. Is it wise to use Asterisk on something this big? I am not a
PBX/Voice guy, I just do IP up here right now. Any
tips, pointers, design guides, or advice to give?
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827 office
$ email me
direct and I will explain how you can save.
Brandon
- Original Message -
From: Tim Jackson
To: [EMAIL PROTECTED]
Sent: Friday, September 10, 2004 4:03 PM
Subject:
[Asterisk-Users] Organization wide
After our department went to using
Woops, wasn't supposed to go to the list ;)
-Tim
-Original Message-
From: Tim Jackson
Sent: Friday, September 10, 2004 5:16 PM
To: Brandon Patterson (peering); Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Organization wide
How can we save? H
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, September 10, 2004 5:40 PM
To: Tim Jackson
Subject: Re: [Asterisk-Users] Organization wide
Tim Jackson [EMAIL PROTECTED] writes:
After our department went to using *, I've had several inquiries about
doing VoIP for my entire
Well, 1GB is what it has now, I can up it to 4GB but I think that's over
kill ;). The coolest part about this machine is 3 PCI busses. 2 64bit
and 1 32bit. I'm assuming that this would make IP through the machine
quite a bit more robust. Since these machines can be had for $500-600
refurbed it is
A local vendor here carries IP500s for sub $200. Right now they are out
of stock, but he has more coming in. If you want his contact info msg me
off list.
-Tim
-Original Message-
From: Ty Purcell [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 2:45 PM
To: Asterisk Users
I'm using it on Debian Stable (Woody), works great, using it with the
backports.org 2.6.7-1-686-smp kernel, zaptel drivers compile fine with
their headers. I think it's all a matter of personal preference. I
prefer Debian, so I use it, use whatever you like best :)
-Tim
-Original
I recently dug into this, from what I've seen, the best bang for the
buck out there is going to be Polycom's. A local vendor has Polycom
IP500 phones for $174 shipped to me. IP500 would be comparable to a
7940G I'm assuming. I ran into the same problem with pricing, don't want
grandstreams, but
make a call between extension 101 and 1009 which are both
Xten Xlite SIP clients, I get that error and asterisk crashes. I'm
running the CVS from yesterday. Any ideas?
Here's the sip.conf 1009 is identical:
[101]
type=friend
callerid=Tim Jackson 100
host=dynamic
dtmfmode=rfc2833
nat
into any performance issues with this machine?
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827 office
(936)414-6723 mobile
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we be running into any
performance issues with this machine?
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827 office
(936)414-6723 mobile
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
What I have found are the best servers, are IBM xSeries machines. You
can obtain a Refurb xSeries 305 P4 2.4ghz for around $750-800. I have
never had the first problem out of these boxes. SuperMicro is also good,
but the IBMs tend to be a little smaller (a lot less deep) than
SuperMicro 1U
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