Can someone please help me. I am currently HEAD as of about 5 days ago
(stable was giving me all sort of problems, upgraded per other users
suggestions) on an Intel mainboard using a mix of Cisco 7960/40 SIP and
7910 SCCP. Can someone please explain what the following means? When
this
I fought with my ata186 until I decided to start dorking with the
settings. I found no outbound faxes could be sent (fax handshake never could
complete) until I set the AudioMode 0x00050005.
Basically this sets the ATA for fax mode which is documented on:
hello
any comments
hello
Thanks for replying. i know duration and billsec.
but i am getting wrong billsec.
for example in one call
billsecduration
48 55
and actually in this call phone rings 10 seconds.
and accual duration on my cell phone is 35
Hi,
Look at
Hi there, I'm experiencing an echo problem and dammed If I can sort it out.
We're running Asterisk on Fedora Core 3 64bit, installed as per
http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3.
These are the specs of the Machine
1 x AMD A64/3500+ CPU: Desktop Athlon64 Retail w/fan SKT
1 x
In article [EMAIL PROTECTED],
Vamsi Pottangi [EMAIL PROTECTED] wrote:
I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3.
I don't have have any zaptel cards, so trying to use ztdummy.
/dev/zap is successfuly created... but I see some problems while
starting asterisk
I am a user of Teliax and voipjet. I find voipjet to be very reliable
and good for outgoing, very low lag, etc. Teliax is good too, but I am
finding high lag rates, to the point where there is a half-second delay.
I ended up just ordering a pots line for incoming (since I am going to
be doing
Hello All,
I am looking for a sip phone that is capable of automatic nat. The
Cisco ata186 for example works fine for natting with iconnecthere, but
as for asterisk, both my 7960 and polycom ip600 require you to set the
nat ip on the tftp.
Does anyone know a good phone (or ata) that can do this
I had been trying to get mwi working on a pingtel phone for some time,
with no success...
The solution was simple. When I made my voicemail.conf, I added the
boxes to the end of the file. The problem was, at the end was in a
different context, so mwi would not light. The solution, all I had to
Could you please give us some more detail as to what you did, in terms of
configuring the hint, and specifically what changes in the behavior of the
running server-phone interaction as a result?
You need to set the hint for the phone when the phone is being dialed like
this:
exten =
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
ChanIsAvail does not work with MGCP channels, as said in the wiki.
But other applications works simular, like Queue and Dial.
What's really the problem with ChanIsAvail?
Is it possible to use Queue and Dial to make a working ChanIsAvail?
I will take a
Thorben Jensen wrote:
Could you please give us some more detail as to what you did, in terms of
configuring the hint, and specifically what changes in the behavior of the
running server-phone interaction as a result?
You need to set the hint for the phone when the phone is being dialed like
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
It seems like Asterisk are having problems detecting DTMF digits when
using an Adit 600 channel bank via MGCP.
I've tried to turn on RFC 2833 on both Adit and Asterisk, but no
digits at all are working then.
Anyone experienced simular with Adit or
Jay Milk wrote:
You got your groups mixed up. Should be:
[default]
exten = _.,1,Dial(ZAP/g2/${EXTEN})
[outgoing]
exten = _.,1,Dial(ZAP/g1/${EXTEN})
Means that anything coming in to channel-group 1 (default context) will
be sent out through group 2, and vice versa.
Watch the console and be amazed
[EMAIL PROTECTED]:~# mpg123 -v
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
Uses code from various people. See 'README' for more!
THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
Hello all,
I was wondering if the DTMF were generated from the phone or from the
ATA? I have a cisco ATA 186.
Thanks
K.
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To
Mark Johnson a écrit :
Is there a way to get a download of asterisk from cvs-head as of like
3 weeks ago? Having some weird problems and most people say that alot
of these things have been introduced over the last few weeks.
cvs co -D 2005-02-15 asterisk
will give you the 15 february 2005
That is nice to hear. Congrats.
Wondering who could help me out with this unique zap channel problem of mine.
Thanks,
~Vamsi
On 5/7/05, Tim Connolly [EMAIL PROTECTED] wrote:
I've got three dual Xeon's running Redhat Enterprise 4 with 2.6.9
and CVS-HEAD from about a month ago. I didn't
Bulls Eye !!! Thanks for that Tony !
It worked. Initially I thought that default conf file would work like
my previous installations.
Thanks,
~Vamsi
On 5/7/05, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Vamsi Pottangi [EMAIL PROTECTED] wrote:
I'm trying to
Can anyone tell me if any IAX service provider
supply audible minutes left/account balance announcement?
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 06/05/2005
I started out happy as a clam with my new Broadvoice account and
asterisk machine. About 10 days ago things began to change. Inbound
calling has been down for 2 days. Beyond the We are currently
experiencing in-bound call issues with a carrier partner in some areas.
We are aware of the
But that only works when SIP/201 receives a call, right?
What if SIP/201 is making a dialout call, does it show as busy in the
phone's keypad?
Julian J. M.
On 5/7/05, Thorben Jensen [EMAIL PROTECTED] wrote:
Could you please give us some more detail as to what you did, in terms of
Yes it does.
Armin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julian J. M.
Sent: Saturday, May 07, 2005 1:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: HINT
But that only works when
On May 7, 2005 12:22 am, JD wrote:
Who's happy with their voip service using asterisk?
I am. Nufone. For the past 18 months. Totally happy.
Where do you get reliable DIDs?
I have a PRI I get my DIDs on. I have not yet found a VOIP provider with DIDs
available in a WIDE area with reliable
On May 7, 2005 05:16 am, Eric Wieling aka ManxPower wrote:
Watch the console and be amazed when _. matches extension h, which is
called when the far side of the call hangs up. You get two calls to the
same number by only dialing once! Stop being lazy and at least use _X.
as your pattern.
Andrew Kohlsmith wrote:
On May 7, 2005 05:16 am, Eric Wieling aka ManxPower wrote:
Watch the console and be amazed when _. matches extension h, which is
called when the far side of the call hangs up. You get two calls to the
same number by only dialing once! Stop being lazy and at least use _X.
Terje Elde wrote:
snip
[m197]
type=friend
username=m197
secret=
qualify=200
nat=yes
host=dynamic
canreinvite=no
context=from-sip
qualify=200, when the server is in the US, and my phone is in Norway,
might not have been the best idea.
Problem solved.
Terje
On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote:
In much of the world the PSTN dialplan is not that simple. Yes, a more
specific dialplan than my _X. exmaple is a good idea, but the USA has a
VERY simple PSTN dialplan and is NOT like most of the world.
Perhaps I am naive but I don't
I'd like to known what I have to do to upgrade
the firmware into a IAXy device.
It does it automagically when it connect to Asterisk if a newer
version is available.
Look in /var/lib/asterisk/firmware/iax and you will see iaxy.bin.
hth
___
You need to put that in whether SIP/201 is recieving or making a call.
This only work for SIP/201 - you will need to do the same for every phone
you have.
thorben
Julian J. M. [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
But that only works when SIP/201 receives a call,
I've had Broadvoice for over a year now, and although their outages are
really annoying, the fact that their service costs $20/month unlimited is
what keeps me with them..
I have 2 Inbound #'s through them (same account), one in GA (678-253) and
one in CT (203-935), and overall their inbound has
We offer termination in:
Miami,USA u$s 0.019
Buenos Aires,ARGENTINA u$s 0.019
Fortaleza,BRAZIL u$s 0.029
Check our rates in CHILE Santiago, PARAGUAY
Asuncion, URUGUAY Montevideo, Punta del Este, BRAZIL Rio de Janeiro, San Pablo,
Goiania, Puerto Alegre.
DID's u$s 5.50 each in all ours
Greetings All,
I have a number of projects in the works at the moment and for one of
them, I need to locate an inexpensive and reliable service that can
provide small-office virtual services:
1. FAX to Email
2. Toll Free number with voicemail boxes for Tech Support, Billing
Inquiries, Customer
Ive reviewed the wiki and other resources and
havent been able to locate a tool which would allow an end user to make
changes to their service. The features and end-user might want to change
is fairly limited (call fwd, number of rings, etc). This might
require real-time. Thanks in
I've been using www.maxemail.com for quite awhile and they provide great
service.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, May 07, 2005 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Thanks,
I'll look into that one as well.
I've been using www.maxemail.com for quite awhile and they provide great
service.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, May 07, 2005 8:52 AM
To: Asterisk
I tend to agree about the in-house being the 'stable part'. Like
anything else on the internet, if you don't have control of all parts
(trunks and phones and dialplans), there are bound to be issues with
uptime, and how your equipment responds to 'their' downtime. It reminds
me of the headaches
Can you post a full dialplan example...
Also, will this only work for certain phones and atas also?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thorben Jensen
|Sent: Sábado, 07 de Mayo de 2005 09:45 a.m.
|To: asterisk-users@lists.digium.com
Andrew Kohlsmith wrote:
On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote:
In much of the world the PSTN dialplan is not that simple. Yes, a more
specific dialplan than my _X. exmaple is a good idea, but the USA has a
VERY simple PSTN dialplan and is NOT like most of the world.
Perhaps I
Andrew Kohlsmith wrote:
On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote:
In much of the world the PSTN dialplan is not that simple. Yes, a more
specific dialplan than my _X. exmaple is a good idea, but the USA has a
VERY simple PSTN dialplan and is NOT like most of the world.
Perhaps I
Andrew Kohlsmith wrote:
On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote:
In much of the world the PSTN dialplan is not that simple. Yes, a more
specific dialplan than my _X. exmaple is a good idea, but the USA has a
VERY simple PSTN dialplan and is NOT like most of the world.
Perhaps I
For example,
I want to give a phone to my brother, who is going to europe. His ICH
softphone is fine there. Both the poly and cisco though require you to
setup for nat. He would not be able to set this up though, so I want to
just give him a preconfig'ed phone and plug and go...
My
1. FAX to Email
Check out TrustFax (http://tinyurl.com/8png8). $10/year for a toll-free fax
number and $0.10 per page in/out.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
___
Isn't amazing what has happened in the last five or six years with the
Internet. There is no design flaw with IPv4. It was created back when
you were in diapers and with todays pda's having more power than the
systems back then. An industry protocol that is going strong 30 or more
years is
Ok, at the bottom of my h323.conf file on my 1st server I have this:
; -
[test]
type=user
host=209.237.227.185
context=termination-test
incominglimit=10
accountcode=005
; -
Using an Asterisk at the other IP, I have this:
exten =
I just installed Asterisk @home 1.0 and its up and running, i added an
extension with the web interface, but now when I try to connect with a
sip client (x-lite) it just times out. here is the log from x-lite
below. Is there any way to view a log on the asterisk side to see whats
going on?
Ok my first question is I have seen messages about a patch for asterisk so
that I can do auto answer on these phones. I found the message in the
archives but I do not have that message as an email still, so I do not have
the attachment. Can anyone tell me where to get it? Also on this phone how
What is the purpose of the beeping?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield
Sent: Saturday, May 07, 2005 12:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] two questions about the Sipura 841?
Ok my first question
My WIP-5000 phone is working well with my Asterisk box now, except for DTMF.
All DTMF key presses come across as clipped or just clicks on the remote side.
I had this problem with my Sipura ATA as well, but fixed that by playing with
the settings on the Sipura device.
I've tried dtmfmode=inband
Jim,
My (3) WIP -5000 phones work just fine with DTMF.
I setup the user.ini file to the following and of course the same in the
sip.conf.
Hope this helps. The OpenSip is also in the browser config for the phone...
[OpenSip]
*T1 = 500
*T2 = 4000
; DTMFType - 0 RTP
; DTMFType - 1 INFO
; DTMFType - 2
Hello All,
Does anyone know how to reduce the incoming ringtime on the polycoms?
What I mean is, When I have an incoming call, my 7960 and pingtel ring
immediately, but the polycom seems to delay 2 seconds before ringing...
Any ideas?
Greg
___
The beeping is to tell you that the remote end has hungup, im sorry I don't
know the technical term for it but it happens on your regular home phone if
the other end was to hang up and you did not hang up your receiver. the web
interface calls it the Reorder.
Thanks
Joel
-Original
I currently have 2 Cisco 7960's and 2 ATA 186's connected to asterisk. The 7960's work just fine for call waiting, but the ATA's dont. I cant seem to get the ATA's to use the call waiting feature, the calls just go straight to voicemail instead of prompting with the usual tone.
Please help
Jim:
I modified your script to first look up Google and then look up 411.com.
It's better for me, because 411.com has Canadian listings too. I still left
Google in because it's much faster and if it has information, I'd rather use
that. I removed the area code thing because it's no use to me. I
BTW This does not do most business name lookups from 411.com correctly.
Maybe someone who actually knows Perl can do that :)
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
-Original Message-
From: [EMAIL PROTECTED]
Michael D Schelin wrote:
snip
Again in the the last few years VoIP has come a long way as the PSTN
has had over 100 years to perfect theirs. If we did not have to
interface with the PSTN don't you think we would be better off? They
didn't have to interface with anybody else.
Well, if one
Anyone have call waiting working on the ATA-186 connected to Asterisk?
Other VoIP phones seem to work, but I can not get the ATAs to allow call
waiting.
Christopher M Iarocci
Network Admin
JD Posillico
631-249-1872 X244
___
Asterisk-Users mailing list
I think it has to do with your CallFeatures.
Callfeatures: 0x
I have a screen shot of my converters config if you want it, it supports
call waiting. I had to turn it off on one of my customers converters once,
I had to change the last 2 digits or something to turn off call waiting.
But
On Sat, 7 May 2005, Andrew Kohlsmith wrote:
Perhaps I am naive but I don't think that diaplans would be that much more
complex if people matched more accurately at all. Granted most of my calling
is north american, but there's some south america and germany in there as
well, along with a
if you could set me up with your config, that would be great.
thanx
Chris
[EMAIL PROTECTED] 5/7/2005 6:52 PM
I think it has to do with your CallFeatures.Callfeatures: 0xI have a screen shot of my converters config if you want it, it supportscall waiting. I had to turn it off on one of
Any special settings on * or your nat firewalls?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Nabeel Jafferali
|Sent: Sábado, 07 de Mayo de 2005 01:07 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE:
On 5/6/05, Ariel Batista [EMAIL PROTECTED] wrote:
I just setup one Dell SC420 with just one SATA drive and 512mg Ram ($
404.00) with 2 TDM04B in it that is 8 FXO ports. And a second system for
another customer with 3 TDM 2 TDM40B 8 FXS ports and one TDM01B for 4 FXO's.
Both systems are
On 5/6/05, Anton Krall [EMAIL PROTECTED] wrote:
Will this only work on polycoms? Do you need to be on an active call to send
text?
As far as I know polycoms are the only phones that support it, but
there might be others.
No you don't really need to be on an active call to send text messages
To be more specific to my point -- Using the internet today,
with the demands of streaming real-time applications, which require a
level of QoS wasn't originally designed in to IPv4. With a wide array
of mods, patches and additions, there is 'some' support for
prioritization. We would be
Hi all - sorry if
what I'm asking is FAQ by now - I only have 2789 digest messages that I've not
read yet...
The local phone
company (Bell South) has gotten completely out of hand with their rates, and
with them suing anyone who wants to compete against them... So, I'm
thinking very hard
Any special settings on * or your nat firewalls?
Nope.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
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On 5/6/05, Steve Rawlings [EMAIL PROTECTED] wrote:
I'm using a TE405p with all four spans enabled, two configured as pri_cpe
and two as pri_net, the asterisk is sitting between our ISDN (UK BT
EuroISDN30) and our phone system. We have 200 DDI numbers on the ISDN's and
I need to give one of
On May 7, 2005 11:04 pm, John Stegenga wrote:
Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive
ring - for a reasonable fee...
Please do a google search for broadvoice problems site:lists.digium.com and
reconsider your choice of VOIP provider.
That reasonable fee
Is your * open on the internet? No firewalls? And on the nat firealls no
need to open any ports or do port forwarding to your natted phone?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Nabeel Jafferali
|Sent: Sábado, 07 de Mayo de 2005 10:09
My thoughts:
1) I would not run asterisk on a laptop, or on Windows (if you can get
it to turn properly via emulation like Vmware).
2) A 586 *might* be enough to handle this low call volume with no transcoding.
3) I know nothing about a Azatel 2 port adapter, but you could acquire
a Sipura 2000
Folks, from googling, I see that the dtmfmode
parameter is not valid in the [general] context.
My problem is that my overseas DID through Libretel
seems to want to come into the [general] context!
And, having done that, I get my welcome message, but
then the DID does not accept the DTMF when I
Are these things possible?
1) Set the local Asterisk jitterbuffer size, but only for a particular
connection. I'd like to force Asterisk to use a particularly large
buffer in certain cases. Should I expect this to work?
[general]
jitterbuffer=no
register = username:[EMAIL PROTECTED]
If Asterisk is on a public IP, nat=yes in sip.conf takes care of all the
required magic. No port forwarding needed anywhere, no special NAT
settings needed on the phone.
Anton Krall wrote:
Is your * open on the internet? No firewalls? And on the nat firealls no
need to open any ports or do
http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants+USA
On 5/6/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I've been lurking here for about a month and I've been putting together
our companies planned migration to a new office and a new phone system.
Could anybody
And if asterisk is behind nat doing prot forwarding? Say you just forwarded
udp 4569 5060 5004 1-2000?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Eric Wieling aka ManxPower
|Sent: Sábado, 07 de Mayo de 2005 11:02 p.m.
|To: Asterisk Users
And if asterisk is behind nat doing prot forwarding? Say you just
forwarded
udp 4569 5060 5004 1-2000?
You'd just need to set externip correctly, assuming you have a static public
IP.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD:
Guys.
How do you configure asterisk to recognize distingtive ringing using x100p
cards? Can this be done and how?
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To UNSUBSCRIBE or
From somewhere in Cyberspace to all
points on the compass its the On Computers Radio Show.
Sunday's show will be all tech talk with the
team. Peter Kastner will start off the show by talking about this week's hot
news stories then the gang will continue on with tech talk. During the
Then you need to use externip= localnet=, portforward 5060 and whatever
ports you are using for RTP. Check rtp.conf. I don't recall if
rtp.conf controls incoming or outgoing RTP packets. You have to
portforward whatever ports the incoming RTP is.
This has been discussed to death on the
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