I’m setting up a new PBX in the Google cloud running FreePBX 14.0.1.36 /
Asterisk 14.7.5. Most calls are fine, but when calling an AT landline
that is busy, ringback tone is heard instead of the expected busy
signal. An example of a failing number is +1 408 269 1999 (a test number
that is
? (The
transcoding issue also affects the old system, but I gave up debugging
it and just disabled g722 on the phones.)
Any advice will be gratefully appreciated.
Thanks,
Stewart
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I am having problems getting a Nicherons TDM400P wildcard clone to dial out.
Everything appears to be configured correctly, but although I see call
progress, it never seems to actually pick up the phone.
(The following is a test of 911 emergency, where I substitute 811 [repair
service] as the
I am having problems getting a Nicherons TDM400P wildcard clone to dial out.
Everything appears to be configured correctly, but although I see call
progress, it never seems to actually pick up the phone.
(The following is a test of 911 emergency, where I substitute 811 [repair
service] as the
â
ââââ â
ââ
Wildcard TE121 Card 0 F10=Back
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astew...@notre1.com
: Andrew Stewart astew...@notre1.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 10, 2010 9:33:45 AM
Subject: [asterisk-users] PRI D-channel bouncing
I need some help getting a system running for one of my company's
plants. I
for the source of this problem?
-aws
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astew...@notre1.com
(205) 585-2980 - cell
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On Wed, Sep 9, 2009 at 8:59 AM, Alex Balashovabalas...@evaristesys.com wrote:
Andrew Stewart wrote:
We are using using what Cisco's Port Address Translation, so that all
SIP traffic is done through %EXTERNIP%. To any outside box, it should
look like the asterisk server is actually
On Wed, Sep 9, 2009 at 10:45 AM, Andrew Stewart astew...@notre1.com wrote:
On Wed, Sep 9, 2009 at 8:59 AM, Alex Balashovabalas...@evaristesys.com
wrote:
Andrew Stewart wrote:
We are using using what Cisco's Port Address Translation, so that all
SIP traffic is done through %EXTERNIP
We are using using what Cisco's Port Address Translation, so that all
SIP traffic is done through %EXTERNIP%. To any outside box, it should
look like the asterisk server is actually on %EXTERNIP%.
My SIP packet gets sent to the ITSP with a Call-ID:
2fd557964ca936b1d72f1328c...@%externip% ,
well with g729 as well :)
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the receiver up manually. Some
traditions die hard.
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/proc/timer_list
Clock Event Device: hpet
set_next_event: hpet_legacy_next_event
set_mode: hpet_legacy_set_mode
HPET showing up as not working means a kernel rebuild.
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to get it working. If none of the tests I
described reveal it then it is not included in your kernel and you need to
build a new one which includes it.
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pitiful offerings) that do not
observe RFC2369 headers?
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On Mon, 7 Apr 2008 11:35:43 -0400, Jay R. Ashworth [EMAIL PROTECTED]
wrote:
Question is: does Mailman *set* it?
Yes.
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On Thu, 3 Apr 2008 09:39:55 + (UTC), [EMAIL PROTECTED] (Tony
Mountifield) wrote:
nothing was shown in the main pane. So there is definitely something
wrong with IE compatibility.
s/ compatibility//
There. I fixed your post :)
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Godwin Stewart - Horwich IT services
:
64-bit/100MHz*
Could you please clarify *WHICH* of the above listed *PCI slots* are
suitable for use with your *High Density T1 cards*.
None of the above listed are PCI slots.
PCI != PCI Express
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based solution?
/me raises hand.
This said, if I did acquire sufficient knowledge of the system to be able
to sell Asterisk-based solutions, I would probably do just that.
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else see where I'm coming from on this.
My VoIP provider doesn't do emergency calls either. Who cares? If the
need arises there are 3 cellphones and the land line here as well as the N95
on which I can place an emergency call.
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, and
they've paid for the call anyway.
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*not* to form a mail body and how
*not* to deal with junk mail.
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are too p*ssed to realize what
they're listening to or watching anyway :)
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. General help that you'll be able to refine WRT the
specifics of your setup is available here:
http://www.google.com/search?q=asterisk+%22no+audio%22
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On Mon, 10 Mar 2008 16:22:45 +0530, sandeep [EMAIL PROTECTED]
wrote:
Can any body tell how to enable call forward facility in INDAI
for an asterisk IPPBX.
Why would it be different in India from anywhere else?
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Asterisk is updated, for now I've backed up the original auth-thankyou.gsm
and symlinked silence/1.gsm to auth-thankyou.gsm.
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to pass the silent option down the chain
to this function, but it's going to mean a major overhaul.
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from each application that uses that common
denominator. However, I also agree with your statement that the 's' option
should silence *everything* with the exception, perhaps, of the beep before
recording starts.
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Godwin Stewart - Horwich IT services
.
How can I suppress playback of auth-thankyou at the end or get VoiceMail()
to play back a different file?
Thanks in advance,
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to remote check
the voicemailbox).
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Or better yet, download the PDF of the Asterisk: The Future of Telephony
(aka the starfish book): http://asteriskdocs.org/
Glenn Cobb wrote:
Go here
www.voip-info.org
and read alot. Almost everything you need to know (or a link to it) can
be found through there.
Seriously, its
,Voicemail(u2070)
exten = 2070,n,HangUp()
exten = 6368,1,Answer
exten = 6368,n,Ringing
exten = 6368,n,Dial(SIP/Linksys02,20)
exten = 6368,n,Voicemail(u6368)
exten = 6368,n,HangUp()
---
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it.
Rob
Andrew Stewart wrote:
I'm trying to setup my first Asterisk setup on a CentOS 5 installation
on VMWare Workstation 6. Got two Linksys SPA941s working fine. But
X-Lite softphones can't answer phone calls, and when one of them calls
on of the Linksys phones they connect
to specify the URL of a new firmware image. I'm
guessing that in some cases it looks at the file contents to decide
whether it's configuration data or firmware, so it works anyway. See
http://www.sipura.com/Documents/faq/Section_2.html#11
--Stewart
I will be out of the office until Monday, May 14. Please contact OWD at
800-337-3839 and ask for Client Services if you need immediate assistance.
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no luck (and your network speed and jitter permits), perhaps alaw
codec with inband tones will work.
--Stewart
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, or
waveform.
If no:
Use SIP Debug (or networking tools) to look at the SIP received by
the PAP2; confirm that Asterisk is sending valid CID info.
--Stewart
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I've spent several days now trying to get my TDM400 card to work. I'm
running TrixBox 1.1 (at least to start). I've tried an old PII 233mhz
with 256 MB, and a modern Dell Dimension 8400 (P4 3.0 ghz, 1 GM RAM). On
both machines I have a series of installation headaches, some of which
seem to
) or something reasonable. If still no luck,
you may have a bad SLIC; try the other port. If all else fails, use a
scope to see what ringing voltage, if any, is present. It's hard to
believe that this could be a * problem.
--Stewart
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, but would be willing to contribute some code.
Another possibility is a standalone gateway program that acts as SIP
server and MGCP UA.
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, compare with *. Same codec?
Same packetization? If not, adjust * to match. If they're the same,
please provide info on your Internet connection upload speed, codec,
type of client phones, etc.
--Stewart
We are using [EMAIL PROTECTED] 1.5 and SIP trunks to communicate to the PSTN
network
best
for both your budget and your growth.
Best Regards,
Jason Stewart
On 11/01/06 15:06 -0600, Jim Freeze wrote:
Hi
I am setting up a phone system for a small office.
The office will have 5-8 phones and a fax line.
There are 4 hunt lines coming into the office.
We have made no hardware
is going wrong,
use Ethereal (on both sides of the router if necessary) to see what
is happening.
--Stewart
Hi all ,
I have tried configuring Asterisk at home to make calls outside our Lan
WITHOUT any success (Setting up your router/firewall so your remote SIP
phones can communicate
. Unfortunately, I'm not aware of anyone
that has implemented it yet. If you undertake such a project, IMO you
should do it in Asterisk, or as a separate process that can run on the
same machine as Asterisk, because many more people would use it and
contribute to its development.
--Stewart
respond 200.
Then, * should send an RQNT to the MG, which should respond 200.
When you pick up the phone, the MG should send NTFY, and *,
after its 200 response, should send another RQNT that causes
a dial tone.
Use Ethereal to check for the above, or verbose mode in *.
--Stewart
that the problem could be fixed by pointing the
phones to a doctored DNS server that returned 11.22.33.44
for provider.com. However, we don't want the added
complexity and unreliability of such a solution.
Any suggestions will be gratefully appreciated.
Thanks,
Stewart
.
Thanks,
Stewart
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.
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like to be able to connect to
MGCP providers.
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. If the latter, just put
a longer timeout in your Dial statement; 180 seconds should be enough.
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start. However, it would
also start on busy signals, rejected calls, etc. Sorry,
I don't know if there is a way to have it start only
when call progress is received.
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shed some light, I'd greatly appreciate it.
Stewart
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Hello Everyone,
Please accept my appologies - I've been reading through the handbook
and the online documentation / mailing list archives and can't quite
get my own answer to these inquiries... The biggest mystery is
how the existing handsets are connected to a new machine running
Asterisk.
, on INVITE, or both?
For all clients, all of the same type, or just one device?
How often?
Does the client reissue the request, and does it then succeed?
--Stewart
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Asterisk
?
Thanks in advance,
Stewart
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to do this? If you just register the two
units with Free World Dialup or similar, it should work ok with NAT
and dynamic IP, and the config will be provided for you.
--Stewart
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On 22/07/05 02:49 +0900, Kuniyoshi Murata wrote:
Hi,
Now, I think I want to disable Asterisk's access to console audio device
based on the logic above. How can I do that?
Make sure the following is in your modules.conf file:
noload = chan_alsa.so
noload = chan_oss.so
of
hardware are you using for FXO?
Jason Stewart
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HTTP uses TCP. Too much overhead. Add SSL on to that and you have a
huge amount of overhead. The end result would be poor and choppy sound
quality.
Jason
On 21/07/05 21:58 +0200, Rob Scott wrote:
For work environments where you only get HTTP or HTTPS access, what is
the feasibility of doing
On 18/07/05 17:06 -0700, Michael D Schelin wrote:
I was waiting for everyone to reply so here is mine.. Check out the
Mediatrix web site. There are no downloads or lists of resellers who might
have this provisioning software that is normally included with purchase.
You may be
to factory settings.
It is possible to disable the factory reset, and conceivable that
the previous owner did that. However, if he did, the SNMP community
string was probably also changed, and the CD that you are complaining
about wouldn't do you any good.
--Stewart
community
string was probably also changed, and the CD that you are complaining
about wouldn't do you any good.
What happens when you try to access the unit? Can you at least ping it?
--Stewart
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On 07/06/05 11:30 -0400, Matt wrote:
Hi,
Has anyone used the SS7 link from Digium? If so, how did it work for
you? Any issues? Anything to be aware of? Do I just need a T1 card
like the PRI card I have now from Digium?
Hi Matt,
There are some links to user reports on the wiki:
Hello,
Firstly sorry for covering old ground - I'm new to this. . . .
I've read that you have to be careful when configuring SIP phone extensions
so that an incoming call can't be connected to the trunk.
Anyone have some info on how this can happen and how to stop it?
Next,
Can anyone tell me
services do that automatically.
If your desire for TCP is not related to firewalls
or packet loss, I'd be interested in hearing about
your application.
--Stewart
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Version 3.1.1 software for MGCP, 02-JUL-2004
When I clicked that link, the license agreement came up.
I did not proceed, but it seems likely to work.
--Stewart
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/tablebuild.pl/ata186
and entering your credentials.
If you still see no files listed, it appears that Cisco has (perhaps
inadvertently) downgraded your account. Open a case with them.
--Stewart
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during Progress. Your phone or ATA may have an option
send RTP during Progress or something similar.
Good luck,
Stewart
Log snippet below:
-- Executing Dial(SIP/116-3e81,
SIP/18887467426 at sip.broadvoice.com|45) in new stack
-- Called 18887467426 at sip.broadvoice.com
-- SIP
to
check all calls. Indeed, the change that I would request might break
operation with some other provider or device.
Is it worth posting such a vague bug report? Unfortunately, I know
absolutely nothing about the internals of Asterisk.
Thanks,
Stewart
software, even on high-end
packages that charge an arm and a leg for maintenance.
Many thanks to Mark, Kevin, and the Asterisk team.
--Stewart
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Richard,
Yep, got that config'd in there:
1001 = 1001,Andy Stewart,[EMAIL PROTECTED]
1002 = 1002,Lorri Barnett,[EMAIL PROTECTED]
1003 = 1003,Andy Stewart - Home,[EMAIL PROTECTED]
1004 = 1004,Andy Stewart - HardPhone,[EMAIL PROTECTED]
1005 = 1005,Lorri Barnett - HardPhone,[EMAIL PROTECTED
Richard,
I feel a little stupid now. Our spam filter (GWAVA) was blocking the
emails because I had WAV files in the block list. One of those things
that doesn't occur to you until you've had a little bit of sleep.
Thanx for the help!
A
*
Date: Fri, 25 Mar 2005
Have Asterisk us at running fine, but have run into a small snag. It's
not emailing the voicemails to the users.
I have attach=yes set, sendmail is configured and works from from the
commandline (sent an email to myself).
Unless I'm wrong, or missing something, asterisk is configured by
default
...]
Thanks,
Stewart
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to upgrade [EMAIL PROTECTED] from the CVS?
If not, could someone please suggest where to start looking at the code?
Thanks,
Stewart
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have good luck connecting to US GSM?
Thanks,
Stewart
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. For huge enterprise databases I use PostgreSQL.
Regards,
--
Jason Stewart | Tel: 616-532-2300
Systems Administrator/ | Fax: 616-532-3461
Programmer | Email: [EMAIL PROTECTED]
Right to Life of Michigan | Web: http://www.rtl.org
/asteriskathome/asteriskathome-0.6.iso
any one know the password for this one?
Hi Phil,
http://asteriskathome.sourceforge.net/install_doc.html
says that the password is password. Don't know for sure,
because I haven't installed it yet.
Good luck,
Stewart
On 10/02/05 15:10 +0100, Jean-Louis curty wrote:
so I stopped asterisk, type mail and got a strange mail saying that
user [EMAIL PROTECTED] could not be reached and body was like if it was
the result of commands ifconfig etc
unfortunally I don't have the message anymore but I went to the log
slave (Media Gateway) on one side, and speak IAX, SIP, or
H.323 on the other?
If this is not available, I would be willing to put some effort
into enhancing the * MGCP stack, to also speak the slave side of
the protocol. Are there other Free users that would be interested
in contributing?
--Stewart
fading out. With 5% loss, a good G.729 system sounds
like a mediocre cellular call, but G.711 sounds terrible.
There are systems that use G.711 when traffic is light, but
switch to compression codecs under heavy traffic to conserve
bandwidth. I don't know how/if this can be done in Asterisk.
--Stewart
Has anyone tried SunRocket with Asterisk?
http://www.sunrocket.com/
The $199/yr. plan seems like an excellent value,
and most reviews have been favorable.
However, I don't know if it is possible to obtain the SIP
credentials, so one can bypass their gizmo.
Thanks,
Stewart
to 50 Hz or down to 20 Hz.
--Stewart
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On 15/12/04 22:53 -0600, Kevin Curtis wrote:
I would recommend Uniden UIP200 phones. Great sound quality with inbuilt
phone book, call logs etc works great with asterisk. I recently purchased
from [1]www.qualvoip.com (they also provided me sample configuration files
for asterisk).
from the * box, but not the phones.
See if you can do ping -I 192.168.6.10 target IP
--Stewart
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the bank so in the event of a power and battery
failure I don't have to type in the configuration commands, just load
a file.
Is there a way to get a config from the Adit 600 and load it back in
again?
Thanks,
Jason Stewart
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192.168.5.10 .
Good luck,
Stewart
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. But whether you can
make phone calls through this system is a complex issue.
NAT traversal for SIP is often problematic, and many on
this list have had to set canreinvite=no.
Regards,
Stewart
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Hi Rodney,
I dont have a cisco acount yet
can some bady hel me with the
ata18x-v2-16-030401a-1.zip file.
You will need a PC running Windows.
1. Unzip it.
2. Read the text file ata186us.txt
3. Follow instructions in it :)
This will convert your ATA from MGCP/SCCP to H.323/SIP .
--Stewart
MGCP/SCCP to H.323/SIP .
--Stewart
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suggestions, too.
Thanks,
Stewart
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at about the same time. You can use their free trial offer to
see if the delay is bothersome.
--Stewart
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On 09/11/04 16:13 -0500, Matt Gibson wrote:
Hi Everybody,
I have a quick question regarding some old Dialogic hardware. We have an
old Artisoft PBX (http://www.artisoft.com/PBXPhoneSystems.html). In this
box are some older ISA Dialogic cards.
My question is, does anyone know if the
assigned by DHCP.
Make sure that your neighbor's kids won't be hacking into your system ;)
Good luck,
Stewart
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the functionality that you
desire. However, I don't know if SIP-MGCP calls can presently
be completed without Asterisk proxying the media stream, so you
may have performance issues. Perhaps someone else can address
that.
--Stewart
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.
Can you configure an H.323 phone to call * directly (without
a GK)? Also, try turning Fast Start on (or off). Likewise with
H.245 tunneling.
Good luck,
Stewart
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