AIPHONE makes all that stuff, I would not try to reinvent that.
Vincent wrote:
Hello
I assume I'm not the first one to think about this: Is it possible to
connect an intercom and/or door bell to Asterisk, so that I can get an
e-mail that someone rang my place while I was out?
Even better:
a call file and do this:
- exten = 5551212,1,System(/bin/cp newcall.call
/var/spool/asterisk/outgoing)
- exten = 5551212,2,hangup
Just my .02
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason
I have need of a very simple callback function - when any call is made
to a special SIP DID, the call is not answered but Asterisk then calls a
pre-determined number - no need for CallerID to capture the calling
number. Does anyone have a simple script to do this?
Chris
--
This message has
tried on of
these - but they look good (and can contain a camera as well if needed).
HTH
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris
Mason (Lists)
Sent: 17 March 2009 13:23
To: Asterisk Users
Has anyone found a good wayt o do a gate intercom using Asterisk? I am
looking at a Xorcom PBX with programmable contact, so I have no issue
with opening the gate, but the interface at the gate is a bit tricky. I
thought about a weather proof housing containing a phone but it seems a
bit
You are configuring Asterisk to LISTEN on 5062 , if you want it to talk
to another server on 5062, then configure that server's config stanza
accordingly.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
I have a client with 30 extensions, all Polycom 501 phones, an Asterisk
1.2.30.2 installation, and trunking over SIP to TelIAX. Everything works
fine except where they need to use DTMF to navigate IVRs such as
Dell.com. The tones are not recognized at all.
My sip.conf lists for each extension:
Ruchir wrote:
Have you set dtmf mode rfc2833 or avt in your phone?
No, I have not changed anything in the phone. The sip.cfg setting is the
default:
DTMF tone.dtmf.level=-15 tone.dtmf.onTime=50
tone.dtmf.offTime=50 tone.dtmf.chassis.masking=0
tone.dtmf.stim.pac.offHookOnly=0
Jonathan Disher wrote:
He has two buildings (the office,
and the shop proper), separated by about 3-400 yards.
Your inter-building distance exceeds ethernet over copper limits, you
will need a fiber link.
paging intercom (to
page employees, etc) on a dedicated extension -
Easy to
I would look at AIPHONE for product that are built to do this and work
perfectly.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
-- Bandwidth and Colocation Provided by
Mike wrote:
do a decent
job of providing QoS on the upstream, which is where you (usually) need it
anyways.
QOS can only be on outgoing, you can't set the priority of a packet
after you receive it. The only other solution would be the cooperation
of the ISP to provide QOS upstream of you.
Royce Souther wrote:
Are there any tests that can be done to pinpoint the problem?
Swap out the card - that usually fixes anything you have control over.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
bilal ghayyad wrote:
OK that worked, but how can we resolve it without need
to type the command manually, as the destination might
change its IP address without our notice, so the
question is:
How can the host be updated periodically (like
externrefresh settings), but need it for host, any
Use a linksys speakerphone behind a metal late and a Push to Connect
button wired as the hook switch - bat phone connection. Mount the mic
and speaker on holes in the plate and the guts glued to the plate.
Simple and cheap and they have to buy from you.
--
This message has been scanned for
Chris Bagnall wrote:
The 4 line limitation has never been a problem for the vast majority of
people.
I can't imagine what an office worker would do with four line
appearances. I use a 6 line Polycom but I register different line
appearances to different customer PBX's that I am working
How would we be able to determine the reasonable cost for an unlimited
plan for an unspecified business? If the business was General Electric,
I would bet they would consider $1M/month very reasonable for unlimited
service. A plan for a corner shop might be reasonable at $19.95/month,
typical for
It might be simpler to have the person record the message then attach it
to an extension. The audience calls the number/extension and listens to
the broadcast.
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International: (305) 704-7249 Fax: (815)301-9759
Yahoo
Kenneth T. Van Wie II wrote:
...
Did you mean to include an answer or a question of some type?
--
Chris Mason
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
Sign up now for
Joao Pereira wrote:
But still, the user can choose not to answer the phone.
I want to force the users to accept the calls.
You would also need a phone that did not have Do Not Disturb. I don't
think you would find that.
I think you could get a software company like Zoiper to remove the
Mike Clark wrote:
Yes, the Asterisk boxes were on private addresses. The Polycoms are also
behind a NAT. Yes, I tried using externip in sip.conf and this allowed
registration, and calls to be placed, but no audio. Unfortunately,
Polycom does not support STUN.
Your problem is not Linux-HA,
Steve Totaro wrote:
You should have no problems. Make sure you put surge protection and
ground your POTS lines. It is a small investment. I have had SEVERAL
FXO modules die or behave strangely after thunderstorms. I cannot prove
it was a surge, but logic would indicate that was the
Watch the mpg123 process, it can take 99% of available cycles and slow
everything down.
I would disable http, cups, smb and any other non-vital process, reboot
and see if things are better.
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International: (305)
Lee Jenkins wrote:
I'd say that Micro is the MS of Restaurant POS. We replace their
systems regularly ;)
I'm curious what with?
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International: (305) 704-7249 Fax: (815)301-9759
Yahoo IM only: [EMAIL PROTECTED]
Teliax has been great, VoipJet seems to be working vry hard to give
outstanding reliability and fault notifications so I would recommend
them as a backup. I like to have two account and failover for termination.
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
***No-Borders Numbers http://reliaclear.com/en-usa/noborders.html
*View Current Service Areas in: Canada
http://reliaclear.com/en-usa/noborders.html#Canada | United States
http://reliaclear.com/en-usa/noborders.html#United_States
I guess it is no borders as long as you only mean the US and
I love the smell of lemonade in the morning
--
Chris Mason
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
It would be better to let MySQL handle that - use the built-in
replication facilities. It's easy to setup.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been
Everything you need is here
http://www.sugarforge.org/softwaremap/trove_list.php?form_cat=345
I use VoiceRD and it workes great.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
If you use a computer witha webcam and use Eyebeam for the softphone set
to autoanswer, you should be able to do this.
However, I would tend to use a browser based security camera such as the
DLink camera which would be a simpler setup.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int:
http://www.dlink.com/products/?pid=295
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
Kevin DeGraaf wrote:
What am I missing? Thanks.
Nothing. Keep your money in your wallet. Your users will never need or
understand using more than 2 different calls appearances atone time.
Even I, with several PBXs to mointor, I never us all the call
appearances I have on a 601.
--
Stephen Bosch wrote:
The fax-to-e-mail services charge as much as the telco does for a
business line, sometimes more (at least, the ones I can deal with in
this area). Better to set-up hylafax, IMHO.
http://www.maxemail.com/fax/fax-lite.html
$24/annum.
--
Chris Mason
(264) 497-5670 Fax:
I have a remote user with Eyebeam on a laptop. Internet connectivity
seems good, there is no packet loss to that location from the PBX.
Everytime the user starts eyebeam, the application tries to register.
Asterisk accepts the registration but the reply never gets to the client
application, so
The only thing I'd probably lose is the ability to do faxes! So I am going
to investigate that further first!
Havn't doen that in years - an online fax service sends me my faxes by
email and I sent out faxes through them, not that I ever do that.
--
Chris Mason
(264) 497-5670 Fax: (264)
Per Jessen wrote:
Perhaps something along the lines of unauthorised tampering with a
telecomms installation?
More likely conspiracy to aid terrorists by destroying the
infrastructure.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK
Tielin Xu wrote:
Hi list:
Has anyone done to set up two servers in different remote offices
through VPN
in order to get the VoIP communication?
OpenVPN will work fine, you do want to have plenty of bandwidth and
processing to make it work though.
--
Chris Mason
(264) 497-5670 Fax: (264)
You must be relying on DNS for the server address resolution. Use the ip
instead.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
Check sip.cfg to see if the voip server is set up as a dns name.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
dangerous content by
If your phone is getting its parameters by DHCP from a linux server, add
the NTP server option to that server:
in /etc/dhcpd.conf
option time-servers 192.168.0.3;
If your phone is getting an NTP server setting by DHCP server, you can't
override that from any setting. I came across
Something like
exten = s,1,SetVar(ALERT_INFO=something)
--
Chris Mason
(264) 497-5670 http://www.snapanumber.com/ Fax: (264) 497-8463
http://www.snapanumber.com/
Int: (305) 704-7249 http://www.snapanumber.com/ Fax: (815)301-9759
http://www.snapanumber.com/ UK 44.207.183.0271
It has one, you just can't see it as easily in photos. It is to the
right top corner of the display, top edge of the phone.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This
bilal ghayyad wrote:
I heared that polycom needs adaptor for the power as
it does not provide standard PoE, also I do not know
this.
You need a special cable to use a 501 with a POE switch, that's all.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax:
Once I updated asterisk it went away.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed
The 501 are extremely sensitive to power fluctuations and will reboot as
a result of a power transient even though every other piece of equipment
is fine.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM:
I found the solution to having calls from SugarCRM auto-answer on the
extension I route it to. I am documenting here for users who may want to
do the same thing.
Setup
I have a second extension on my phone (500) which the calls are routed
to, but there is no reason this wold not work on the
Sune Kloppenborg Jeppesen wrote:
They seem nice but with a Via EDEN 300Mhz CPU are they any more powerfule than
the Soekris net4801 with a 266Mhz CPU?
I get the version with a 1GHz CPU.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK
I have the polycom auto-answering calls to the extension I am using by
using this in the dialplan
exten = 830,1,SIPAddHeader(Alert-Info: RANR)
exten = 830,n,Dial(SIP/830,25,t)
However, I want the feature for the SugarCRM contact feature, which uses
the manager interface to place the call to
Here's how I do it.
Buy complete fanless system flash card ready unit with four ethernet
interfaces:
http://www.ibt.ca/v2/items/fwa7204/index.html
It is very small, in an aluminium extrusion case, very robust.
Install Astlinux on 128 MB flash card - http://www.asterisk.org/
Voila! - flash
Brian Capouch wrote:
What kind of money are those things? There doesn't seem to be any
price information on the website you linked to.
About $350.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM:
Lacy Moore - Aspendora wrote:
Has anyone tried this?
This is how I do serious installations. I use a cheap ADSL connection
for browsing traffic and a dedicated feed for voice. Dual hjome the
machine, run shorewall setup for two ISPs, and write the rules to route
accordingly.
--
Chris
I think you can do this with outlook. Use the Third Lane dialer product,
set your extension to that of the conference, then initiate the calls.
It will call the extension then the party and connect the two.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax:
[EMAIL PROTECTED] wrote:
I joined VirtualPhoneLine.Com service and am really enjoying the use of it.
I am pretty certain this constitutes fraudulent and *misrepresentative*
http://www.google.com/search?hl=ensa=Xoi=spellresnum=0ct=resultcd=1q=advertising+misrepresentativespell=1
advertising.
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so normal calls do not get auto-answered. However, I have
Andrey Solovjov wrote:
This usually happens if one of the log files in /var/log/asterisk is
more than 2Gb...
I had deleted all the log a week previously, so that's is not likely.
I think we have a bug. I built two systems on the same hardware, the
only difference was one had a Sangoma
I have a persistent problem with a PBX I commissioned recently. After a
few days it goes into a spasm, creating thousand of log files and giving
the message below on the CLI.
Dell PE 1600 with Sangoma A200.
pbtpbx*CLI show version
Asterisk 1.2.14 built by root @ pbtpbx.local on a i686 running
Do you rotate Asterisk's logs with the logger or with logrotate?
Don't know. I have not modified the installation.
What do you have on logger.conf ?
[general]
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)
; see
Tzafrir Cohen wrote:
Do you rotate Asterisk's logs with the logger or with logrotate?
I have never addressed this before and never seen this problem before.
The issue is causing thousand of log files to be written to the
/var/log/asterisk directory, so many that I have to use find to
Older models, 500 and 600, are 12V, newer 601s are 24v
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
dangerous content by
Mochamad Susantok wrote:
I have already use smokeping, and great for measure latency and packet
loss, but not voip packet especialy, or you has been modified smokeping ?
I have not modified it. I take it that if the network has considerable
latency, so will VOIP. It has been my experience
I like smokeping, as it gives a good sense of the quality of the route
over time.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
Forum wrote:
Does anyone on this list know of a reputable T1/PRI provider in St.
Lucia? If so, what monthly costs am I looking at? I do know that
Cable and Wireless are the biggest Telco.
I think you will find they are the only telco and the cost will be enormous.
--
Chris Mason
(264)
Jeremy McNamara wrote:
as most people here know, I yell at stupid people.
Be honest, Jeremy, you yell at everyone!
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message
Martin Joseph wrote:
I found a terminator called sellvoip.net, whose website is
crap(currently), but whose route from my server is very clean and short.
My calls all sound perfect now. I keep teliax and nufone configured
as backups, and they both largely work well, but not as well as my
Wilson Pickett wrote:
How do I program the dialplan in extensions.conf to:
(a) try multiple provider to make an outgoing call based on current
latency between my * box and the different providers ?
To do this, you need a seperate application that would run something
like fping on all your
(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2 or provider 3 ?
The way to do this is to use a PSTN based DID provider such as Kall8 and
use a rollover list to route to your DID provider. If your VOIP provider
goes dead,
Douglas Garstang wrote:
Somewhat off topic...
I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying
to get all calls forwarded to Asterisk. However, (and this is hard to believe),
the docs say that 1-stage calling (I presume that means no PIN is required) is
not
Marcin Lukasik wrote:
Have you even _tried_ to create your dialplan?
And to make it worse, he copied this drivel to the Developers lists.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL
Kevin Smith wrote:
Any other thoughts as to what may have caused the phone to reboot?
the power supplies on these phones are very underrated and any power
fluctuation will cause them to reboot. I get it when we are on generator
and the A/C cuts in.
--
Chris Mason
(264) 497-5670 Fax:
I have no problem with paying Digium the $10 for G729 licenses, everyone
has to make money. It's the administration of the licenses that sucks. I
experiment with different hardware a lot, and make up demo machines to
install for customers with available hardware. I have to put G729
licenses on
I normally build Asterisk on Centos4 from source and have no major
problems, but I tried the [EMAIL PROTECTED] from the iso to make integration
of the FOP and FreePBX applications easier for the next machine I am
building. I have a Via C3 based system with a Digium T1 card in it and I
I am trouble finding a configuration that works for ChanIsAvail and SIP.
My two providers are Voxee and Teliax.
I have these lines in a macro
exten = s,n,ChanIsAvail(SIP/teliaxSIP/voxee)
exten = s,n,Cut(CH=AVAILCHAN,-,1)
exten = s,n,NoOp(AVAILCHAN= ${CH})
; Dial the available Channel
exten =
Joshua Colp wrote:
Please don't do product announcements on asterisk-users, that's what
asterisk-biz is for. Thanks!
Please don't send the whole announcement back to the list just to add a
line tot he bottom. Trim.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:
Rich Adamson wrote:
Well... outbound calls via Nufone still function, so I'd guess they
are scrambling to find another cost effective source for DID's (which
will likely not be the same DIDs they had due to probable portability
issues). That would also suggest 800 numbers can't be remapped
Kerry Garrison wrote:
I
am ready to pull my hair out. I cannot seem to get the Polycoms to read
the time properly. Regardless of the server they are pointed to our the
offset, i am getting the correct time, but 24 hours ahead. So for today
it is showing Friday April 28 but with the
One of my PBXs drops calls after 7 to 10 minutes. I cannot see any
reason for this. I upgraded to asterisk 1.2.7.1 last night, still no
improvement.
Calls are IAX2 to either teliax or voxee, doesn't seem to matter which.
Codec is G729.
Connecting over ADSL.
Load is only onw or two calls,
I hav been trying to debug this for a while and I am drawing a blank. I
have an Asterisk PBX with a Adtran 750 and a Digium T1 card.
Sip to Sip calls work great.
Incoming PSTN calls through the CB have one way to audio when sent to a
Sip phone on the local network. What could cause that?
Joao Pereira wrote:
Hello
I configured Asterisk to put CDRs in the database like it was
explained in:
www.voip-info.org/wiki/view/Asterisk+cdr+pgsql
What I want to know is how do the billing solutions (like
Asterisk2Billing) work with Asterisk.
The billing system just use the information
Tom Vile wrote:
I am open to suggestions as well.
On 4/18/06, Seth Remington [EMAIL PROTECTED] wrote:
Why is it a problem. I have 800 numbers through Teliax without any problems.
--
Chris Mason
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
jennyw wrote:
We've been reasonably happy with Polycom SoundPoint phones, but we
only have them installed on the LAN. I've read that they have problems
working across NAT. So ... I guess I have a few questions. First, is
there a way to get Polycoms to work well over NAT? If not, then are
I have been experiencing dropped calls on my iax2 connections between my
Asterisk server and my ITSP providers, I use Teliax and Voxee but it
seems to happen on both so I don't think it is the provider. I don't see
any packet loss at the time so I don't think it is poor Internet
connectivity,
mike webb wrote:
i wrote previous about a setup i thought might work with asterisk and
the tsu-600. no one replied, so i thought i would ask if anyone is
using a tsu-600 with asterisk and if so how do you have it setup ??
___
I have three working.
Title: Message
Curt Shaffer wrote:
As of now we
are probably looking in the
36 range. We would like to utilize this as a first step to migrating to
a VoIP
system.
To save cost. I would buy a couple of used Adtran 750's, they are cheap
and readily
I am copying the Master.csv file to another server and importing to
mysql. I am looking for a simple billing application that will produce a
bill for a give account code for a give period, based on a rate table.
Is this available?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Darren Wiebe wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Disclaimer: astpp is my software. :-)
It's quite easy to do this with astpp. Depending on exactly what you
want, there are a few ways to do it. Drop a note on the astpp forum
(www.astpp.org/forum) or on the astpp mailling list
Tele Cost Price Reducer wrote:
i would suggest Astribank-8 of XorCom. it is a dedicated
Asterisk compliant solution.
you can look at : www.xorcom.com.
Looks interesting, shame they don't have a FXO version.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305)
I have a Astlinux system that can't run mysql. I want to bill calls by
extension but all I have is the raw CDR files. Is there any software
that cal produce a simple call accounting from text CDR files?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax:
Lists wrote:
Has anyone successfully
hooked up ADT or Brinks home alarm
system to say a analog port or SIPURA through Asterisk?
I tried to to do with with another alarm system and gave up. The tones
are too high frequency to work.
--
Chris Mason
NetConcepts
(264)
Lists wrote:
I am hoping the alarm companies adopt quicker to the internet.
I don't see that happening. Internet reliability is not going to be
sufficient for alarms. PSTN lines, for all their issues, don't fail, and
alarm systems can sense the dial tone and alert if it is missing.
I would
Steve Murphy wrote:
Hello--
In the interest of Symmetry, does anyone else in the world see any need
for a device like the IAXy (or the SIP ones from other manufacturers,
like the ATA186), but one that presents an FXO interface instead, so it
can be connected not to phones, but the PSTN?
I have incoming PSTN lines on an Adtran 750 channel bank. Calls are
evaluated by an agi script based on callerid and forwarded to an
international DID through Voxee. There is an IVR at that number that
asked to user to enter a selection. When the user presses a key, my pbx
puts the call on
Erick Perez wrote:
How can I measure this latency all the way to the asterisk?
I have found two good ways to monitor routes for VOIP. Install mtr and
run mtr your.voipserver to find where you are seeing the latency, and
then install smokeping (not so easy to install) and you
FYI:
I am trying to build zaptel-1.2.4 against the recently updated kernel
version 2.6.9-34.EL on Centos 4.2. but I am getting errors and it will
not build. This is apparently due to a typo in a kernel header
spinlock.h although I have not successfully modified the kernel and
built zaptel
I have three Sipura 3000 FXO untis for incoming PSTN lines on a small
pbx. There is an IVR to select the extension. The DTMF tones are not
being sensed so the IVR does not work and incoming calls are not being
answered. I have listed my sip.conf entries.
Is there any solution to this?
I am trying to use ChanIsAvail to detect the best route for a call. I am
testing by dialing an extension that is then forwarded to the DID.
Normally it will be an incoming PSTN call that is forwarded.
When I try it, I get put on hold for a few seconds and miss the
beginning of the recorded
Jordan Novak wrote:
Warning: This message has had one or more attachments removed
Warning: (winmail.dat).
Warning: Please read the NetConcepts-Attachment-Warning.txt attachment(s) for
more information.
I have two issues...
First I am working with a hotel software vendor to include an automated
[EMAIL PROTECTED] wrote:
Time Warner provides an emta not an ATA and the technology is
different. You do not even need internet connection for that and runs
over their own private network through DOCSIS.
Who manufacturers that unit? Have you found a way to interface it to a PBX?
--
Chris
I am trying to put a Shorewall firewall in front of my PBX, all the
other port forwards work fine but forwarding port 4569 to the PBX is not
working, it is being logged as rejected even though there is a DNAT rule
in shorewall.
Anyone seen this and have a solution?
--
Chris Mason
--
This
I don't mess with configuring these, the wizard on voxilla.com does
everything except set the right context. Try using default for
everything to get it working then separate as needed.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK
Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network latency? If
not, how can I measure latency?
The most effective visual indication of network reliability in as far as
it relates to VOIP is an application called smokeping. The graphs show
latency as shadows
Have any providers started to offer T.38 yet? I am anxious to find a
solution for faxing.
--
Chris Mason
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation
1 - 100 of 336 matches
Mail list logo
