Hi,
We recently tried installing Asterisk for a small office. We figured the
safest way to go would be to buy from someone who sold equipment
specifically for Asterisk and to use a consultant that they
recommended. However ... it didn't turn out so great. Sound quality is
terrible -- the
Eric Wieling aka ManxPower wrote:
Sean Rima wrote:
Eric Wieling aka ManxPower wrote:
Sean Rima wrote:
Does anyone have any experience of these, I have been offered one
and am
thinking of adding sticking it onto the back of my Asterisk box and
just
ignore the WAN port if possible, It would
Hello
I have problem with transfer call if using ACD
When i using ACD with agent and queue setting, i cannot monitor call and transfer call. this's my setting
- i have 2 IAX phone (phone number as 201, 202), agent.conf
agent = 1001,4321,member 1agent = 1002,4321,member 2agent = 1003,4321,Tin
then,
Hello
how to calculator billing exactly when IAX accept the call, my configure
customer -- telco --- asterisk -- ACD -- IAX
at time, for example: 11:00 i dial to asterisk
11:01 asterisk answer channel and dial to IAX phone (11:02)
ring 20 second (at 11:22).
when IAX answer call (11:22) and talk 10
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Can Asterisk(*) be installed on this and if so are the setup
instructions any different?
I have a client that's asking.
Specs: 533Mhz, 512MB RAM, 10GB HDD, Cobalt 4 OS
- --
=
Joshua Abbott, Support Technician
Jenny -
I'm glad I'm not the only one! I just installed Asterisk on Friday and I
spent all day trying to de-gremlin my system. I'm glad I'm doing it for
myself and we haven't switched from our legacy system yet, but I have a
potential client that wants to see how well I can implement this system
harry gaillac wrote:
Hello,
I patched asterisk cvs head sources with
http://juraj.bednar.sk/work/software/asterisk/messaging/
and presnce patch without success.
asterisk send 405 method not allowed to sender.
I use polycom ip300.
THat is a response to the polycom's PUBLISH request, a
harry gaillac wrote:
Hello,
I patched asterisk cvs head sources with
http://juraj.bednar.sk/work/software/asterisk/messaging/
and presnce patch without success.
asterisk send 405 method not allowed to sender.
I use polycom ip300.
THat is a response to the polycom's PUBLISH request, a
Hi,
Klemens Kasemaa schrieb:
hi
PSTN -- [Teles ISDN / Asterisk] -- SIP client
When call is made through ISDN, no matter if taken from PSTN or
Asterisk side, person in PSTN side can hear perfectly but in Asterisk
side I only hear a very scrambled or very low quality voice, words
repeated
jennyw wrote:
Hi,
We recently tried installing Asterisk for a small office. We figured the
safest way to go would be to buy from someone who sold equipment
specifically for Asterisk and to use a consultant that they
recommended. However ... it didn't turn out so great. Sound quality is
hello
i m getting follwing messages in asterisk-1.0.9 what
is the reason calls are not going out. can u pls tel
me how to solve this
Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new:
Unable to allocate socket: Too many open files
Aug 20 13:06:09 WARNING[7706]: channel.c:311
Hi,
I got my TDM11B and am trying to get it to work on my PC.
However, I'm having difficulty getting the wcfxs driver to
load. I've Googled this problem, and while there are others
who have ran into the same problem, none of the solutions
work for me.
I would very much appreciate it if you
Jennyw,
I have setup about 8 Asterisk systems with The TDM400p boards in them. Yes
allot of them had at the beginning some echo and other things. But I have
been able to work and get them fixed.
1) Make sure your motherboard is able to assign it's own IRQ for the board.
This is one of the
On Sunday 21 August 2005 02:25, Joshua Abbott wrote:
Can Asterisk(*) be installed on this and if so are the setup
instructions any different?
I have a client that's asking.
Specs: 533Mhz, 512MB RAM, 10GB HDD, Cobalt 4 OS
As long as you don't expect to run more than a couple of calls at a
On Thursday 18 August 2005 14:55, chawki hammoud wrote:
Hi:
Please advice me of a voip provider with reasonable
reseller program. I was using voipjet and it has a lot
of problems.
Did anyone experienced asteriskout.com service? They
have good prices.
What you may want to do is to
People, please cut down the original post in your replies.
It's wasting space, bandwidth and time.
--
List Manager
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To UNSUBSCRIBE or update
I would like to know how to install asterisk 1.0.9 with zaphfc working
on a SuSE 9.2. I tried this:
- The rpms with SuSE 9.2 are asterisk 1.0.6
- bristuff works, except for zaphfc, which doesn't compile.
- The official asterisk download file doesn't contain isdn bri support
Any ideas?
Lars
On Sunday 21 August 2005 02:24, Nguyen Trung Tin wrote:
Hello
how to calculator billing exactly when IAX accept the call, my configure
customer -- telco --- asterisk -- ACD -- IAX
The phone company does not bill for talk time but for use time. You used the
phone network for 33 seconds.
Kamran Ahmad wrote:
i m getting follwing messages in asterisk-1.0.9 what
is the reason calls are not going out. can u pls tel
me how to solve this
Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new:
Unable to allocate socket: Too many open files
Aug 20 13:06:09 WARNING[7706]: channel.c:311
hi
PSTN -- [Teles ISDN / Asterisk] -- SIP client
When call is made through ISDN, no matter if taken from PSTN or
Asterisk side, person in PSTN side can hear perfectly but in Asterisk
side I only hear a very scrambled or very low quality voice, words
repeated several times. Same is
I like Broadvoice but there are others.
Do you want SIP or IAX termination? Business or residential?
Mark
Joshua Abbott wrote:
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Hello
I currently have internet service through MediaCom (Cable Internet)
and need to find a VOIP provider that is
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SIP and/or IAX would be nice. Residential.
Mark Phillips wrote:
I like Broadvoice but there are others.
Do you want SIP or IAX termination? Business or residential?
Mark
Joshua Abbott wrote:
Hello
I currently have internet service
On the clicking front, it could still be packet loss. I recently (just
this week in fact) solved a clicking problem my client was having.
turned out to be interference from his wireless network bridge. Both it
and his Cisco ATA186 were sitting next to each other. I put about 3 feet
of space
What about asterisk chan_sip and IM +presence !!!
Harry
--- Olle E. Johansson [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
Hello,
I patched asterisk cvs head sources with
http://juraj.bednar.sk/work/software/asterisk/messaging/
and presnce patch without success.
On Sunday 21 August 2005 01:05, jennyw wrote:
up too high). The reseller and the consultant both say that the most
likely cause for this is using Digium cards w/ analog phone lines.
Apparently, they say, sound quality can be pretty bad.
The reseller/consultant aren't worth the money you paid
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Hello,
What are some common reasons why a phone would report not registered
even when the extension has been setup through Asterisk(*) AND phone
username/password is correct?
Joshua
- --
=
Joshua Abbott, Support
Hi Paul:
Thanks very much for the suggestion. I don't understand why this just
began to happnen. I never
had problems before. Your suggestion has shed a lot of light on the
problem. Because wcfxs disagrees about the version of the symbols
listed below, I get the following unknown
Hi Rich:
Does the new distro tree issue explain the unresolved symbol references
noted in my last post? I don't know if SuSE's 9.3 network autoupdate
would have changed the tree structure. The unresolved symbols are (from
dmesg):
zt_receive
zt_qevent_lock
zt_ec_chunk
zt_transmit
The easiest thing is probably to get a card that is more widely supported.
Any cheap pci HFC-S card will do, they are sold for anthything between 9
and 15 eur.
With an hfc-s card you can then use bristuff or chan_capi
On Sun, 21 Aug 2005, Klemens Kasemaa wrote:
hi
PSTN -- [Teles ISDN /
Sure, this can work. I have an Asterisk install in my home office
running about the same hardware. Our is a newer mini-itx system booting
Astlinux from a compact flash card.
Just don't bother with any form of transcoding, especially G.729a...you
don't have the cpu power for more than 2 channels
I too have had trouble with FXO interfaces. I tried the Sipura SPA-3000
FXS/FXO device , X101p cards and a TDM11B card. None were satisfactory
for my small office with 6 extensions and 3 lines.
My longer term workaround was something that I setup just to bridge a
perioud when I was taking down a
I got my TDM11B and am trying to get it to work on my PC.
However, I'm having difficulty getting the wcfxs driver to
load. I've Googled this problem, and while there are others
who have ran into the same problem, none of the solutions
work for me.
I would very much appreciate it if you
Rich Adamson wrote:
2. I think the driver you want is wctdm (not wcfxs). I don't use
the fxs modules, but the fxo modules use wctdm. Seems to me there
was a change some time ago where the fxs modules are now supported
from with wctdm. I'm not 100% sure though.
In 1.0.x it's called wcfxs in
www.teliax.com has treated me very well for about six months, and have
lots of choices for DID (and 800) numbers.
www.nufone.com handles calls very nicely and provides 800 numbers. But,
their web site leaves a little to be desired, and they only use paypal.
-BEGIN
Hi,
I just stopped in at Best Buy here in CT, USA. I found an interesting
offering from Vtech there. It states it's a VOIP wireless phone system made
for the Vonage service.
Here it is at their website:
http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm
For $149, it comes with 2
Hi Rich:
Does the new distro tree issue explain the unresolved symbol references
noted in my last post? I don't know if SuSE's 9.3 network autoupdate
would have changed the tree structure. The unresolved symbols are (from
dmesg):
zt_receive
zt_qevent_lock
zt_ec_chunk
I never heard anything on the AMP list, so figured maybe someone here might
be able to help me sort this one out..
I was making some updates to my attendant config, which is really very
basic, and now incoming call processing stopped. Not sure exactly what the
heck happened, but figured
I just stopped in at Best Buy here in CT, USA. I found an interesting
offering from Vtech there. It states it's a VOIP wireless phone system made
for the Vonage service.
Here it is at their website:
http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm
For $149, it comes with 2
If this is a limitation of asterisk, where is it located? In the
chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN?
Best regards,
Arik
Peter Svensson [EMAIL PROTECTED] wrote:
On Sat, 20 Aug 2005, Nico Giefing wrote:
how many connection do you have from your asterisk to
I would like to know how to install asterisk 1.0.9 with zaphfc working
on a SuSE 9.2. I tried this:
- The rpms with SuSE 9.2 are asterisk 1.0.6
- bristuff works, except for zaphfc, which doesn't compile.
- The official asterisk download file doesn't contain isdn bri support
Any ideas?
Lars
Is this a CVS-HEAD that was released AFTER 8/13? I ask because we're using
that release and it's still deadlocking.
Thanks,
Sherwood McGowan
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Kevin P. Fleming
-Sent: Saturday, August 20, 2005 3:25 PM
Yes, and no. It would help to know what phones you're working with, what
you've got in sip.conf, and what you've got in extensions.conf.
-K
Hello,
What are some common reasons why a phone would report not registered
even when the extension has been setup through Asterisk(*) AND phone
I did a quick google search of the lists site and couldnt
find a definitive answer, so if its there, I apologize for asking again.
Starting about noon yesterday, I am no longer able to
send/receive calls via Broadvoice. When calling in, I get a fast busy, and when
calling out I get the
Thanks,
I just read all the literature on the Vtech website, and, I think you are
exactly correct!
Oh well, so much for cheap SIP wireless phones :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Sunday, August 21, 2005 2:51 PM
To:
On Sun, 21 Aug 2005, Arik Funke wrote:
If this is a limitation of asterisk, where is it located? In the
chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN?
It is inherent in the channel-based structure of Asterisk. An audio
channel is the basic measure used by applications
Title: Message
We want touse
the accountcode in the CDR for billing and tracking total usage.
We wanted to set the
accountcode for calls coming into our network so we know which of our users to
assign the usage to.
But then when we
receive an "on network" call we run into a problem with
In your opinion, is there any chance that this situation will change in
forseable future? Is anybody already working on such a pseudo-channel
structure required for decoupling asterisk channels from the physical
channels? If not, is there a significant interest in the asterisk
community to do
Hi list!
I'm trying to get PrivacyManager working but for some reason it always
thinks that CallerID is present (when it isn't). I get this on the
console:
== Primary D-Channel on span 1 up
-- Accepting voice call from '' to '0711234567' on channel 0/2, span 1
-- Executing
Hi Everyone:
Problem Solved. Thanks to Matt, Paul and Rich for their excellent help!
It is always appreciated.
Here's the solution, for thsoe interested:
SuSE distributes zaptel drivers and the auto update referenced puts
drivers that may not be compatible with the current release on
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul Belanger
Sent: Friday, August 19, 2005 4:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] any ISDN/PRI signaling experts out
there?
All these things are is just an ATA built into a plain vanilla analog
cordless phone system...ok, a 5.8 Ghz systemvanilla with sprinkles
;-)
Nothing special at all. You'd be better off with a Sipura device and a
Panasonic cordless phone.
Michael
On Sun, 21 Aug 2005 15:12:49 -0400, Andre
Anyone has the settings to connect a TE405 to Meridian-1 line side E1? I
saw T1 on the voip-info.org, but no E1. Is Nortel's E1 a variant of MFC/R2?
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Asterisk-Users mailing list
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There seems to be a random thread of BV issues this last week all
amounting to the same proble - no calls.
Do a sip debug peer sip.broadvoice.com and see what happens. I found
that BV were sending calls to my number and for some odd reason my *
server wasn't dumping them into the
Hello everybody. Recently I've been trying to limit the duration of some
calls for a simple application I'm writing. Unfortunately all of the
documented methods are failing and I'm not sure what else to try. Here
are some samples of what I've done:
1) The AbsoluteTimeout application.
-
I have been looking for the answer to this question for a
while. Google-ing and reading the archives of Asterisk-Users has not
enlightened me.
It seems that this question has been asked many times, and many times it
has gone unanswered.
I have call waiting and three way calling on my
I already have a Sipura and a panasonic cordless, I was just hoping that I
could find a true SIP wireless phone at a reasonable price :-)
Oh well, one can drea laugh
Thanks,
- Andre
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
[EMAIL PROTECTED] wrote:
Hello everybody. Recently I've been trying to limit the duration of some
calls for a simple application I'm writing. Unfortunately all of the
documented methods are failing and I'm not sure what else to try. Here
are some samples of what I've done:
I believe this
Mark,
Thanks for the tips. After adding the exten = XX,1,BLAH, i am able to
received calls, however I still get the same error when dialing out, and now,
there is an additional error on the end. I am beginning to think this is a
Broadvoice issue and will try to contact them after
Hi
i just implemented asterisk and is such a grate solution...i am using
polycom 301 and 501 phoneson lan a iam using g.711 and i have a
16 port linksys switch...
the problem come when somebody inside the network is making a call to
other extension (in the same network) and
Nico,
Same problem for me, did you find a way to compile latest CVS-HEAD ?
And by the way, you're right realtime do need HEAD version.
Thanks
Laurent
At 00:04 20/08/2005 +0200, Nico Giefing wrote:
but the non head version is not working with realtime configuration?
hm, i think its a
Thanks, everyone, for your suggestions. I'm going to stop by the office
tomorrow to try some of these out.
Here's more info on the setup: We bought a brand new computer for this
-- I don't have the specifics right now, but will look that up in the
office tomorrow. We have two Digium cards --
Does your linksys support traffic shaping in any way? If so you should
mark your VoIP data as a high priority.
Alternatively, use a slimer codec like G729. I'm not sure what the
Polycom supports but It'll probably do 729. You'll have to buy some
licences for your * box too.
Mark
Ing. Marlo
Help for this use case:
I need detect a dtmf tone during conversation.
For example:
A call B
B answer A
A and B talk
B (while talk) press *111
and I defined an action en extensions.conf to
exten = *111,1,Application()
and while the application is executed A and B continues talk
I have CVS-HEAD as of yesterday and it's still not working for me. Maybe
I'll try updating again and post the results.
Thanks,
Tim
[EMAIL PROTECTED] wrote:
Hello everybody. Recently I've been trying to limit the duration of
some
calls for a simple application I'm writing. Unfortunately
Hello,
My asterisk currently will dial an outside number after I dial the
number and press send on the phone.
How can I get it setup so I have to press '0' for an outside line.
Kind regards
Michael Felder
IT Medic Australia Pty. Ltd.
P: 03 9557 2213
F: 03 9557 2214
M: 0419 568 217
E: [EMAIL
Hello,
Any use of SDP media attribute in conjunction with SIP /Asterisk ?
I would appreciate any insight!
George
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo!
I currently have one of the Hitachi WIP-5000 SIP phones. I've been
using it off and on for about four months. Just one as an initial test.
While it's pretty good it does have some minor issue, or they could be
issues related to my wlan access points. Whatever the case, it drops
off lan
Dear list,
I was installing Asterisk via the AMP method off the AMP website.
There is a portion in there where they want you to use perl-cpan to
install telnet.
The first time I installed I had no problem. I messed up and trashed
the box further down in the install.. This time I made a
Steve Maroney wrote:
Does anyone know of any plans to add an intercom/all-page feature in *?
The few SIP phones that have auto-answer capability would be better if
Asterisk could broadcast one leg of a channel to many legs at one time.
I'm looking for an answer to this problem also. I am
I got my TDM11B and am trying to get it to work on my PC.
However, I'm having difficulty getting the wcfxs driver to
load. I've Googled this problem, and while there are others
...
1. The TDM card has several different revisions (rev e through h,
I believe). If you have one of the later
I had a similar issue with sending a flash to the PSTN for call waiting. I
found that my dial plan in the extensions.conf file was not allowing me to
dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78,
*79, etc. Training the wife on how to actually use it was an entirely
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
My zapata.conf config: -
# Below setting is for E1
switchtype = national
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
My extension.conf
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Please price me on this. I need a service provider that does this.
Like the guy that mentioned collocation. Could this be done and for
how much?
Joshua
1) What type of phones do you plan to use (analog, SIP, Skinny,
H.323, MGCP)?
SIP
2) How many
I can sign that immediately.
I am not using asterix yet, but I am having VoIP phone behind IpCop and
never had a problem yet.
About the SOHO design you mention, the only limitation it has is that
you can only have a single green network (internally subnet), but you
can abuse the blue
Hi, folks,
Is it possible to connect a IAX softphone to a SIP softphone via Asterisk?
IAX client -- Asterisk -- SIP client
I tried that, and I was able to dial and talk to my IAX client from the
SIP client. But not the other way around, I couldn't dial the SIP
client from the IAX client. The
On 8/21/05, Scott Huang [EMAIL PROTECTED] wrote:
Is it possible to connect a IAX softphone to a SIP softphone via Asterisk?
IAX client -- Asterisk -- SIP client
I tried that, and I was able to dial and talk to my IAX client from the SIP
client. But not the other way around, I couldn't
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