I am looking to build a small PBX for an office that has 3 incoming analog
lines and less than 10 extensions.
For the Asterisk server I am going to use a small form factor PC with no-PCI
slots so the FXO interface needs to be either FXO-SIP or USB. Can anyone
make suggestions?
I am
I am in the process of trying to terminate a PRI into a new * server. The
server has an old T100P T1/PRI card in it. I have compiled the following on
Centos 5.4.
dahdi-linux-complete-2.2.1+2.2.1
libpri-1.4.10.2
asterisk-1.4.29
Everything seems to have compiled fine. DAHDI reports Found a
time. If I had to guess I
would say it is a configuration error.
Thank you and have a nice day,
Anthony Francis
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Merkel
(Mail Lists)
Sent: Thursday, January 21, 2010 1:41 PM
service.
On Tue, Jan 20, 2009 at 9:32 AM, mail-lists mail-li...@peachnet.com wrote:
Hello,
We're having some issues with CallerID and I thought someone here might
be able to shed some light as none of our carriers seem to know what I'm
talking about.
The issues is this:
A client of ours uses
Hello,
We're having some issues with CallerID and I thought someone here might
be able to shed some light as none of our carriers seem to know what I'm
talking about.
The issues is this:
A client of ours uses an after-hours voicemail service as mandated by
their corporate office. We have a
about callerid or ani,
they instead use the DID that the call comes in on to decide how to
answer the call.
Get a different voicemail/answering service.
On Tue, Jan 20, 2009 at 9:32 AM, mail-lists mail-li...@peachnet.com wrote:
Hello,
We're having some issues with CallerID and I thought someone
Steve,
Hijacking this post here - How 'good' is freeswitch currently. I'm
looking for some sort of SIP proxy and have looked into openser and ser.
Freeswitch seems to have more functionality than these and it seems a
lot easier to configure. I particularly like the xml config files, etc.
Our
Alex Balashov wrote:
mail-lists wrote:
Steve,
Hijacking this post here - How 'good' is freeswitch currently. I'm
looking for some sort of SIP proxy and have looked into openser and ser.
Freeswitch seems to have more functionality than these and it seems a
lot easier to configure. I
I can't see why not. You should easily have enough power for asterisk.
You can probably also run it as your firewall in a home environment
thanks to the dual RJ45's
I don't know whether or not you can use the built in RJ11 to interface
with your POTS line though - maybe someone else could
Hm. $300 in the US and the UK disty is selling them for just short of
£240, so they can go stuff themselves, low-power or not. (I buy 1GHz
systems with 1GB of RAM, running at 15W for half that. No drive though)
Gordon,
If you don't mind my asking: What do you get for $150.00 ?
Woah,
How weird. I JUST bought this off of ebay 2 minutes ago. The exact one.
This will be my first time playing with PoE. I have all cisco phones
here but I'll let you know how it goes.
This will be my first major asterisk experiment and I'm trying to
choose a PoE switch for 15-24 phones. I
Same here - Gafachi has been great. Decent rates, very stable and great
voice quality.
I use Gafachi.com http://Gafachi.com and have good quality with no
minimum requirements. Try them at www.gafachi.com http://www.gafachi.com
On 12/16/07, *Benjamin Jacob* [EMAIL PROTECTED]
mailto:[EMAIL
Chris Bennett wrote:
Hi All,
I am seeking input from anyone who may have seen a similar
configuration and dealt with similar issues to what I'm experiencing.
Configuration:
- 2 sites (site A and B)
- Asterisk 1.2.23 on each site (Trixbox)
- Internet 512/512 symmetric at each site,
Does anyone know how I could integrate my Asterisk setup with Outlook so
that when I click on a phone number is my outlook address book it will
dial the number and ring my SIP phone so that I can just pick it up? I
am interested in this integration for WinXP with Outlook 2003 and
Anciso, Roy wrote:
Hello List,
Does anyone have access to the soft key configuration files for the
Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
didn’t find much up there.
As far as I know (and I might be very wrong), you can't change the soft
key configuration of
Turbo Fredriksson wrote:
We have now got our new PRI line (10 channels, 100 numbers) connected
and everything is working except the outgoing caller ID. Whatever
SIP phone I'm using, the CID that's shown is the very first number...
I don't know if the same is true for you but we had to call our
I was wondering how everyone here is giving users (say via the BLF on a
Polycom, or the sidecart/buddies) the ability to see how many channels they
have in their group and how many are in use. Since so many users are used to
seeing Line 1, 2, 3 etc on a key system I have been trying to think
I am having some really strange problems calling from 2 asterisk boxes
of mine. One is version 1.2.22 the other 1.2.18. The problem is
identical on both boxes.
When I try to call certain numbers (8006375410, for instance) the call
rings and rings and rings. Eventually the receiving end will pick
Razza wrote:
On 27/09/2007, Eric B. [EMAIL PROTECTED] wrote:
For starters, what is the difference btwn the 1.2 and 1.4 branches of
Asterisk? I can't seem to find a document that describes the changes.
Anyone?
Not much/Lots
Depends what you're looking for. Important considerations for us in
Tzafrir Cohen wrote:
On Mon, Sep 24, 2007 at 02:04:05AM +0100, Ex Vito wrote:
This might sound lika a small issu, but here it goes: I'm a long time
unix user and my shell history usage and editing is configured to use
vi keybindings; it's something that's already built into my fingers
How can I know that the traffic went directly between
the endpoints and did not go via the asterisk?
I'm sure there are many ways to do this
one way would be to do rtp debug on the cli and watch for media packets
another would be to do tcpdump on the command line and watch for packets
Hello,
We're in the process of moving to a PRI circuit for our asterisk switch.
Can anyone point me in the right direction as far as PRI Cards are
concerned?
Thanks!
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Jared Smith wrote:
On Thu, 2007-07-19 at 11:28 -0400, mail-lists wrote:
We're in the process of moving to a PRI circuit for our asterisk switch.
Can anyone point me in the right direction as far as PRI Cards are
concerned?
You haven't given us many details on your setup, but I'll take
Jared Smith wrote:
On Thu, 2007-07-19 at 12:11 -0400, mail-lists wrote:
thanks for your reply - Our setup isn't complicated at all - just a PRI
coming into an asterisk box. Maybe you could answer another question for
me - what disadvantages does a PRI have from a channelized T1? or vice
Hello,
We're running into a problem I thought some of the enlightened people on
this list might be able to help with. Our VoIP stuff has grown to the
point where it makes sense to get a PRI (we've been doing things purely
voip till now).
The problem we're running into is this:
We have
Noah Miller wrote:
The 430's have two line appearances. I'm trying to get the second line
registered to a different extension but for some reason it's not
allowing me to do this. The first line will register fine but the second
line never seems to register no matter how I swap the device ID's
Hello all,
I have some polycom 430's which I'm trying to get to work with asterisk.
I have them working for the most port other than one little issue.
The 430's have two line appearances. I'm trying to get the second line
registered to a different extension but for some reason it's not
I probably shouldn't be hijacking this thread but it seems that there's
some people paying attention here that know what they're talking about.
We've recently acquired a cisco IAD 2400 router with 2MFT-T1 VWIC card
in it. Doing some cursory reading It seems that this card can be
interfaced
[EMAIL PROTECTED] wrote:
Hi.
Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the
call established but no sound heard on either end.
What is the best/correct way to try and see what codecs Asterisk is using on
each end of the call as it passes through Asterisk
Anton Krall wrote:
I have been looking around for examples or code on making a click to call
application for web sites... has anybody had any luck on this topic? Is
there any open source code out ther that could do this?
What we have done in the past is created url's like this :
shadowym wrote:
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
What's the bottom line with recent updates on 1.2.x? Is it production ready
for fax? By
Ken,
I have similar problems every now and then on one of my asterisk boxes.
I'm also running CentOS4 on that box.
I've found that doing a sip reload when in that state results in
something along : Last reload not yet finished (can't remember the exact
wording)
We're using cisco 7960's
ax.
The downside of rx_fax is that you need to compile it into asterisk.
The downside of iaxmodem is that (to my knowledge) you can't easilly
implement an auto-answer/detect fax/voice/ auto attendant/voicemail
system. The channel must be dedicated to faxing, and that's that. This
may or may
The downside of rx_fax is that you need to compile it into asterisk.
The downside of iaxmodem is that (to my knowledge) you can't easilly
implement an auto-answer/detect fax/voice/ auto attendant/voicemail
system. The channel must be dedicated to faxing, and that's that. This
may or may
will never use.
On 5/4/07, mail-lists [EMAIL PROTECTED] wrote:
I get the following error when trying to compile zaptel on CentOS 5
kernel 2.6.18-8.1.3.el5
CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o
/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â
/root/asterisk
Forrest Beck wrote:
The problem is that your kernel is newer than the xbus-core.c file is
looking for. See:
http://forums.digium.com/viewtopic.php?t=15317sid=7beaf6bfed1550f4a8676427283800c5
thank you for pointing me in the right direction with this - the answer
is write there in
Forrest Beck wrote:
The problem is that your kernel is newer than the xbus-core.c file is
looking for. See:
http://forums.digium.com/viewtopic.php?t=15317sid=7beaf6bfed1550f4a8676427283800c5
thank you for pointing me in the right direction with this - the answer
is write there in
Hello,
I'm trying to compile asterisk from source (1.2.18). Faxing is fairly
critical for us, so in the past we've used spandsps app_rxfax and
app_txfax to support faxing in asterisk. Unfortunately I can't find
these applications on soft-switch.org anymore and even so I have a
feeling they
Stefan Wintermeyer wrote:
Steve,
Am 04.05.2007 um 14:44 schrieb mail-lists:
I'm trying to compile asterisk from source (1.2.18). Faxing is fairly
critical for us, so in the past we've used spandsps app_rxfax and
app_txfax to support faxing in asterisk. Unfortunately I can't find
I get the following error when trying to compile zaptel on CentOS 5
kernel 2.6.18-8.1.3.el5
CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o
/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â
/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: â has no
Salvatore Giudice wrote:
You can go directly to 5.2 and then move on to 7.x and 8.6. 5.2 allows you
to upgrade to the newer firmware releases that have an app loader, which
Cisco added in later releases. Beware that some cisco non-sip loads can not
generate the proper firmware filename to
Boot it off flash and have it load an initrd.gz into RAM. Everything
will
run entirely from RAM - no writes to the flash at all! I can get
everything inside a 48MB flash drive, but I use 64MB ones which gives
me
space to store configs, etc.. (of-course, I make it sound so simple ;-)
but
Hello,
I'm trying to put together a low cost - low powers PBX appliance for several
customers. I have purchased a couple of the soekris net4801 boards and have
asterisk up and running on them fine but they just don't quite cut it in the
processing power department. I've been able to get about 10
Hey these look pretty good - I was going to build my own but if they're
~$350 like you say it's probably not worth it.
I haven't played around with Astlinux at all. I'm assuming it doesn't
install freepbx does it?
I don't really need (or want) the gui - but clients will. I'm assuming all
your
On 3/10/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Sat, 10 Mar 2007, Mail Lists wrote:
I've had a look around and I think I have settled on one of the VIA EPIA
fanless boards. Does anyone have any experience with these running
asterisk
as far as performance and reliability
[test]
disallow=all
allow=gsm ;GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
Are you sure that the xlite phone can handle gsm??
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
a) to what extent Asterisk can manage everything necessary to allow
machines A and B to communicate if they were SIP phones. Is it
possible to go for a setup with the firewalls/NAT devices as shown
If the asterisk machine isn't NATed you shouldn't have a problem at all.
If you're using
If your connections are VoIP, the first area to look at for quality is
network jitter/congestion/drops.
I'm mostly worried about drops. A little bit of garbling I can deal with
but a dropped call is just VERY bad. Especially when it happens again
and again. Does anyone know any methods
Hello,
Can anyone recommend the 'best' kernel and zaptel versions to use with
asterisk?
we're currently running trixbox and are having numerous call quality
issues(disconnects, echo, garbled speech) and I'm considering wiping the
asterisk box and installing a virgin copy of centos,
Yes,
But try calling your cell from one of your phones - Does the cell start
ringing the moment you hear ringing on the SIP phone? or does it ring a
half ring/full ring later?
I'm just curious because I do the same thing. (Very strict pattern
match) but I don NOT have the 'r' option set in
I've been getting a number of dropped calls today with the following
visible in the logs:
Feb 5 12:06:29 DEBUG[8340] channel.c: Bridge stops because we're zombie
or need a soft hangup: c0=SIP/peachnet-213243-b7830fd8,
c1=SIP/2142-08f4b6a0, flags: No,No,No,Yes
Feb 5 12:06:29 DEBUG[8340]
Mark Coccimiglio wrote:
Marty,
Where are you paying $1000 for a 1600 series Cisco? I can get you
20% off that price on any quantity (note: Sarcasam). Its not the
1990's anymore. You can get them on eBay ($50-150) for only slightly
more then the Linksys. The performance is rock solid.
Steve Totaro wrote:
Doug Lytle wrote:
Steve Totaro wrote:
Steve,
You neglet to mention:
Distro
Version of HylaFAX
Version of iaxmodem
Version of Asterisk
How you're connecting to the PSTN (From previous conversations,
I'm guessing PRI)
I can't say that I'm not experiencing
shadowym wrote:
Is there anyway to get CDR to show just the answered calls. Not by
exporting to a spreadsheet and editing. We have ring groups and queues and
CDR shows everything as calls received. Even if it's multiple extensions
ringing it shows them as multiple calls received. This seems
Also, I am not using a zaptel timer. Could this possibly be causing
problems with DTMF??
I really don't know for certain but here's what I experienced: When
calling out asterisk gives the option to allow called numbers to
transfer by hitting the '#' by putting 'T' (or 't'?) as an option in
sip extension and call somebody outside via my
sip provider, dtmf is recognized.
On 11/9/06, mail-lists [EMAIL PROTECTED] wrote:
Also, I am not using a zaptel timer. Could this possibly be causing
problems with DTMF??
I really don't know for certain but here's what I experienced: When
calling
Vicky wrote:
Anyone having problems with voxee since last few days or is it just me
? In peek hours i get LAGGED when i do a iax2 show peers or even 1000
ms latency . Most of time it is 20 ms or so but when i start sending
traffic to them latency increases to 1000 ms or even LAGGED ( also
Hello everyone,
This probably isn't the correct place to ask this but I thought I'd
check here first.
We're getting ready to roll out a hosted pbx solution on a very limited
trial basis (some company employees are going to get voip service at
home). Our main issue is of course bandwidth.
I concur,
We run about 50 7960's and they work quite well. Sound quality is pretty
good. Like Aaron said, unless you're running call manager you can't
program soft keys, etc. We're looking at going to a different phone
that would give us some more customization options. Also.. Cisco's come
Martin Joseph wrote:
On 2006-11-02 07:34:15 -0800, mail-lists [EMAIL PROTECTED] said:
snip
My question is this: How do huge voip companies like vonage handle
bandwidth. I'm pretty sure that they have to have sufficient
bandwidth available for X numbers of simultaneous calls, in other
words
Barry Fawthrop wrote:
Thanks Andrew
I have no plans to VoIP my Faxes to a VoIP provider
I just would like to send them from my desktop (which is windows) to
my PBX (which is AstLinux inside a net 4801)
The PBX connects to PSTN lines via a FXO Gateway (CG-410 in my case)
So really it's
Hello,
I've posted this at the trixbox and freepbx forums and haven't been able
to get an answer. I thought perhaps the guru's here might be able to
help me out :)
I'm having some issues with setting caller IDs. There are 2 problems
that I would like to solve.
1. I have a DID pointing to
Rene wrote:
Hi Arik,
I have Asterisk running as guest on my Debain Xen system and it works
fine. I used to work with an AVM Fritz!PCI ISDN card as well by compiling
the ISDN driver (Hisax) in the XEN-kernel. You have to be aware that the
host system does not use the ISDN PCI card by putting
I am trying to get an initial setup up and going
which I assume is a very common question here. My basic
questionsare the following:
Can I get Asterisk up and going without voice cards
using it with SoftPhones internally as a proof of concept. (just calling
extensions and leaving voice
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