Before or after you compiled zaptel and asterisk? It needs to be installed
before you build everything else.
after :-(
thanks for your reply!
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Hi,
I've been using Asterisk (now version 1.2.4) for quite a while, and I'm
trying to switch from POTS lines to a VOIP termination service. I've
had this problem with a few services I've tried, so the problem must be
on my end. Here's what I see:
When dialing out, I hear rings, and the call
Worth saying that the Aastra 9133i with the 1.3.1 firmware is a pretty
good phone. I used to run GXP-2000's, still have 10 new in a box and
another 20 in demo/test circulation, but I also run a few dozen 9133i,
480i and 9112i phones and I think Aastra are getting their now. Biggest
problem I had
OK, well the audio option was the last one I required for now.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gareth
Owen
Sent: 19 February 2006 16:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
I'm trying to get working a spa3000 with asterisk. My problem is I
cant get wroking the incomming calls (I installed the lastest
firmware). My problem is asterisk is rejecting the authentication from
the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I
think I placed the username
I'm a newbye myself so beware!
(1) http://www.voip-info.org is your friend. I've got most of my info off
that site and it's a good place to start.
(2) Download a Softphone like XLite (you'll also find info on softphones on
voip-info) and start experimenting on site. When you'll be able to
Yes, you need to remove the 'System' part.
You should only have:
exten =
s,n,MixMonitor(${CALLDIR}${CALLFILENAME}.wav||touch/tmp/test${UNIQUEID})
Garth
Alex Barnes wrote:
Has anyone had any success using the MixMonitor() plus command as
nothing I have tried works.
I am using 1.2.1 I
Thankx MC,
This is the solution.
Ive tried it and it works perfect.
But Ive got a question.
I want to set a variable with the command
SetVar
I place the following text file in the
directory /var/spool/asterisk/outgoing/
Channel:
Zap/g1/0655871460
MaxRetries:
0
RetryTime:
30
On Mon, 20 Feb 2006, Lee Archer wrote:
Worth saying that the Aastra 9133i with the 1.3.1 firmware is a pretty
good phone. I used to run GXP-2000's, still have 10 new in a box and
another 20 in demo/test circulation, but I also run a few dozen 9133i,
480i and 9112i phones and I think Aastra are
On Mon, 2006-02-20 at 01:22 -0800, [EMAIL PROTECTED] wrote:
I would still like to know what they were smoking when they put two
_10 meg_ ethernet ports on the linksys 942.
Probably the let's not cannibalize the 79xx series pipe. Wouldn't
surprise me if the Ethernet chip is capable of doing
On Mon, 20 Feb 2006, Nathan Alberti wrote:
Is there a reason the variable ${DNIS} does not get set with incoming calls
via chan_capi ?
I don't know any channel setting DNIS. What are you expecting with that
variable?
Is it related to the MSN=X in capi.conf ?
No. msn= is obsolete and does
I have Data Sheet for 942 from Linksys web site. It says this on page 4
(close to bottom):
Physical Interfaces:
2 100baseT RJ-45 Ethernet Ports (IEEE 802.3)
And that was one of the reasons I was considering 942. Do you think the data
sheet may be wrong?
- Original Message -
From:
trixter aka Bret McDanel wrote:
On Sun, 2006-02-19 at 21:05 -0500, Doug Lytle wrote:
trixter aka Bret McDanel wrote:
since you have had a little time to play with this, was this the
problem?
Haven't had a chance yet, will look at it when I get into work this morning.
Doug
--
Hi,
Is it possible to use the hint priority to allow call parking slots to
be monitored on (for example) Snom indicator lamps? How do you refer to
the slots (i.e., what is the channel) in the hint?
- Mike
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Where can I get the tar.gz sources of libnewt?
Reg,
Anthony Azzopardi.
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On Mon, 20 Feb 2006, Soner Tari wrote:
I have Data Sheet for 942 from Linksys web site. It says this on page 4
(close to bottom):
Physical Interfaces:
2 100baseT RJ-45 Ethernet Ports (IEEE 802.3)
And that was one of the reasons I was considering 942. Do you think the data
sheet may be wrong?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You can not define groups in sip.conf
But there are, as you hint, other ways to solve the problem, like using
queues or implementing it in dialplan logic.
Do you have any example how to do that?
--
Tomislav Parcina
[EMAIL
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
aHR0cDovL3d3dy52b2lwLWluZm8ub3JnL3Rpa2ktaW5kZXgucGhwP3BhZ2U9QXN0ZXJpc2srY21k
K1ZvaWNlTWFpbAoKSXQncyBhbGwgaW4gdGhlcmUKCk9uIDIvMTMvMDYsIFRvbWlzbGF2IFBhcuhp
bmEgPHRwYXJjaW5hQGxhbWEuaHI+IHdyb3RlOgo+Cj4gSGkgbGlzdCEKPgo+IEhvdyB0byBzZW5k
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Hi Patrick,
I'm new to Linux, so can you please tell me how
I have used version 0.0.2 every version from pre8 bar pre23 with 1.0.x and
pre23, 25 with 1.2.2 and 1.2.4. My libtiff is 3.5.7 with asterisk 1.0.x and
libtiff 3.7.1-6 with asterisk 1.2.2 and 1.2.4
I am of the personal opinion through experience that txfax talking to rxfax
does not work, and
well, does this gateway support SIP?? and does it generate its own CDR? could you send the devices brocure/tech spec.??
thanks
On 2/19/06, Sam Tam [EMAIL PROTECTED] wrote:
Why not get 30 GSM Gateway from us at £60 each and then get an asterisk or some voip gateway like A800 and then link it
On 2/20/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:
Hi,
Is it possible to use the hint priority to allow call parking slots to
be monitored on (for example) Snom indicator lamps? How do you refer to
the slots (i.e., what is the channel) in the hint?
You're looking for the
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I think it's a bit of a known fault - the attended transfer function
does not work from the queue system. It would be nice if it did, though.
Hi Paul!
Is there any explanation about this? Is that something that will change?
--
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You'll have to use uattended transfers for CCs.
l.
I have read Paul's mail. Is this bug or feature?
--
Tomislav Parcina
[EMAIL PROTECTED]
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Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
aHR0cDovL3d3dy52b2lwLWluZm8ub3JnL3Rpa2ktaW5kZXgucGhwP3BhZ2U9QXN0ZXJpc2srY21k
K1ZvaWNlTWFpbAoKSXQncyBhbGwgaW4gdGhlcmUKCk9uIDIvMTMvMDYsIFRvbWlzbGF2IFBhcuhp
John Morris wrote:
Hi,
I've been using Asterisk (now version 1.2.4) for quite a while, and I'm
trying to switch from POTS lines to a VOIP termination service. I've
had this problem with a few services I've tried, so the problem must be
on my end. Here's what I see:
When dialing out, I hear
On 19 Feb 2006, at 14:54, Dumpolid Exeplish wrote:
Hi everyone,
Can anyone give me suggestions on any equipment that can connect
from VOIP to a GSM gateway (channelbank that can load up to 30 sim
cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked
at 2z's Stargate (which
Hi folks,
Over the weekend I finally decided to upgrade one of our Asterisk
systems from 1.0.9 to 1.2.4
I had no significant problems and all is well in general - as usual
Asterisk rules!
However, I did run into two small issues. Can anyone help me solve them
please? The first one
Hello, Mark!
At 06:33 AM 02/20/2006, you wrote:
Please forgive the question, but what is the rationale behind using Solaris
over Linux as an asterisk hosting platform?
Because of a few reasons, actually:
(1) The remote hardware management options available for the X2100
work better (or only,
Thank you very much.I will contact Sprint, Magma, and Unlimitel about their service.For those who want to participate in Ottawa Asterisk Users Group, please send me email off-list at oss_richard at rogers dot comso I can update you and share ideas on activities.I can ask
Dumpolid Exeplish wrote:
Hi everyone,
Can anyone give me suggestions on any equipment that can connect from
VOIP to a GSM gateway (channelbank that can load up to 30 sim cards and
make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate
(which has a VOIP card) but the device
yusuf wrote:
Dumpolid Exeplish wrote:
Hi everyone,
Can anyone give me suggestions on any equipment that can connect from
VOIP to a GSM gateway (channelbank that can load up to 30 sim cards
and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's
Stargate (which has a VOIP card)
Ah! There you go - I knew Chuck Norris had something to do with it...
;-)
Mark
-Original Message-
From: Alexander Burke [mailto:[EMAIL PROTECTED]
Sent: Monday, 20 February 2006 11:17 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Asterisk on Solaris 10 (AMD
Doug Lytle wrote:
trixter aka Bret McDanel wrote:
since you have had a little time to play with this, was this the
problem?
Haven't had a chance yet, will look at it when I get into work this
morning.
This works correctly now.
Doug
___
I'm trying to get working a spa3000 with asterisk. My problem is I
cant get wroking the incomming calls (I installed the lastest
firmware). My problem is asterisk is rejecting the authentication from
the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I
think I placed the
2006/2/20, Rich Adamson [EMAIL PROTECTED]:
In the [linea2] section, you want the fxs port to be able to place
calls as well as receive calls, therefore use type=friend. The same
with section [line2]. Note that I also changed the [PSTN_2] to [line2]
to match the username= line. Watch the
On Tue, Feb 21, 2006 at 12:17:43AM +1100, Mark Edwards wrote:
At 06:33 AM 02/20/2006, you wrote:
Please forgive the question, but what is the rationale behind using
Solaris
over Linux as an asterisk hosting platform?
Solaris is also a supported OS (well if you pay for it). It's also 64
bit
I usually do the same for IVRs, but I always make sure not to use
itself as the increment, and I use a tempvar instead, like this:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(100);this is where we start the loop
exten = s,100,Set(TCOUNT=${COUNT})
exten = s,101,Noop(${COUNT})
exten =
Hi folks,
need some help on queue behaviour.
What Im trying to do is accepting a call from
pstn, put it into a queue, while callee is waiting contact some numbers till
one responds, then bridge the two calls.
What I cant manage is jump to next dialplan
command soon after callee enters
Hello everybody,
I have this problem where I can't get a ring tone when
SIP devices dial an IAX2 route. I get the ring tone
using IAX2 devices to dial the same route. The call
completes normally in both cases...
Facts:
- Asterisk 1.0.9
- The Dial command is within an AGI.
- ATA (grandstream)
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You have to use AgentCallbackLogin for that.
If a phone logs in that way, it's reachable as Agent/200
You can also use AgentCallbackLogin to logout the agent.
You don't have to worry about an agent that forgets to
logout on phone X
2006/2/20, Rich Adamson [EMAIL PROTECTED]:
In the [linea2] section, you want the fxs port to be able to place
calls as well as receive calls, therefore use type=friend. The same
with section [line2]. Note that I also changed the [PSTN_2] to [line2]
to match the username= line. Watch the
I have 2 Polycom SP 500's attached to my system. Both are behind NATs,
but both seem to work fine, for the most part.
A few weeks ago, I started to notice that I get an error message from
one of them:
Feb 20 08:54:58 NOTICE[10663]: chan_sip.c:7691 handle_request:
Registration from 'sip:[EMAIL
I have that set up, but I cannot get some of the phones to change the
hint State. The SNOM phone show State:InUse, but Swissvoice phones show
State:Idle even when on a call.
I use 'show hints' to see this.
Kind Regards
Garth
Colin Anderson wrote:
Breeze to set up, too. To monitor and
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
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Tuesday, 21 February 2006 1:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] spa3000
I don't mess with configuring these, the wizard
I think this would help you.
http://packages.debian.org/unstable/perl/libnewt-perl
Anthony Azzopardi wrote:
Where can I get the tar.gz sources of libnewt?
Reg,
Anthony Azzopardi.
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On 20/02/2006, at 12:08 PM, Andrew Furey wrote:
On 2/20/06, Nathan Alberti [EMAIL PROTECTED] wrote:
Is there a reason the variable ${DNIS} does not get set with incoming
calls via chan_capi ?
Is it related to the MSN=X in capi.conf ?
Just a guess, are you thinking of ${DNID} instead?
Phil Blundell wrote:
I suspect that the
datapath through our regular network switches is probably close enough
to lossless for this purpose as well.
You could be surprised. I know that I have been suprised by how easy it
is for a UDP packet to get dropped or lost. I've had a few customers
Anyone know if asterisk supports q931 85 in the uk?
Thanks
Bails
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Hi there
We are wanting to build our own SIP soft phone using VB6. What is a good
component to use for this? We have done research and have only found very
expensive ones offered by VaxVoip or Radvision. Anyone know of a good
component that does the basics that doesn't cost two arms and both
On Mon, Feb 20, 2006 at 03:15:52PM +, bails wrote:
Anyone know if asterisk supports q931 85 in the uk?
Nope, it only supports Q.931 110 (which is EuroISDN). 85 is UK ISDN,
most providers can set the line to 110, but you may have to ask for it.
Marconi System X switches (as used by BT, THUS
Adam Robins wrote:
Hi Adam
After many days of playing with the new jitterbuffer and trunking options for IAX2, I
have finally received almost acceptable quality. I am receiving 5-8 complaints a day of
calls breaking up from both the customer and agent sides. What I have
discovered is
Page 3 of the same data sheet reads:
Optional 5 volt DC Universal (100-240 Volt) Switching Power Adaptor
And, Package Contents section on the same page reads:
Important Note: Power Supply is Ordered Separately
-- Models: PA100-NA, PA100-EU, PA100-UK, PA100-AU
This explains the PoE issue, I
Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can
make calls from one h.323 device to the world using sip trunks :)
I can call to sip devices from the h.323 one. Now I want to make calls
from sip to h.323 but it does not work. Maybe one of us have a
configuration example to do
Hello,
I didnt exactly find what the
problem was but I built a new Asterisk server, copied the conf files over from
the original server and now the phones work fine.
Thanks, Dan
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Peters
Sent: Friday,
Anyone integrated Landmark (formerly Southwestern Bell) Digital Key System
phones into an Asterisk installation? The phones are model DKS930 and the
main CPU for the system is a DKS1224. I'm hoping to reuse some of the
phones with a new Asterisk install I'm building.
Thanks!
John
Cornell's
You have to make all of your manual changes in the
_custom.conf files. [EMAIL PROTECTED] overwrites the
xxx.conf files -I think this happens every time you restart the
app.
Log files are usually in /var/log/asterisk and you can see
them in the maintenance screen on AMP
From: yrving
Hi,
Can you post your working config, I'm wasting my time to config h323-sip
Thanks
Guillermo Salas M wrote:
Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can
make calls from one h.323 device to the world using sip trunks :)
I can call to sip devices from the h.323 one.
On Mon, Feb 20, 2006 at 08:56:41AM -0500, C F wrote:
I usually do the same for IVRs, but I always make sure not to use
itself as the increment, and I use a tempvar instead, like this:
Why?
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(100);this is where we start the loop
exten =
Hi,
I got this message on my Asterisk messages file and
after it Asterisk went down...
2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl:
ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting
TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:+
1^2006-02-18 08:13:55
Telecom Ottawa?
Large, Ultra fast pipe with direct connections to TDM providers (Which
may be at 151 Front St. in Toronto) but they should work for what you
want.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
OSSSent: February 19, 2006 12:04 AMTo:
It was trying to perform looping in the dialplan that made me seriously look at
AGI. Gee, I wonder what's easier.
This:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
exten = loop,1,GotoIf($[${COUNT} 5]?next,1);exit if more than 5 esle start
again
exten =
Why not get an asterisk and install
software like a2billing on it.
It has CDR and things like that
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid Exeplish
Sent: Monday, February 20, 2006
7:33 PM
To: Asterisk
Users Mailing List - Non-Commercial
Dov Bigio wrote:
Hi,
I got this message on my Asterisk
messages file and after it Asterisk went down...
2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or
TOK_COMPL or TOK_LP or TOKEN;
Douglas Garstang wrote:
It was trying to perform looping in the dialplan that made me seriously look at
AGI. Gee, I wonder what's easier.
This:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
exten = loop,1,GotoIf($[${COUNT} 5]?next,1);exit if more than 5
Dov Bigio wrote:
Hi,
I got this message on my Asterisk messages file and
after it Asterisk went down...
2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN;
Hi,
Why not try http://www.voipjet.com
Been with them and found them quite good... Try it out
Dan#
On 20/02/06, Joseph Tanner [EMAIL PROTECTED] wrote:
I have voicepulse connect too. I had occassional problems with
incoming calls, but not many and not recently. Have had more problems
Good to know about that Loopstart thing --- helped me quickly solve my
problem of the phones not ringing :-)
thank you for the input
Chris
- Original Message -
From: Leo Ann Boon [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Hello,
I have about 10 Asterisk PBX in production with Bristuff-0.2.0-RC8q
(asterisk 1.0.10) and I want to use Bristuff-0.3 now for the new PBX
I am going to set up.
With Bristuff-0.2.0-RC8q the ISDN lines are working fine, but the new
version of Asterisk add some nice features.
All these
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Saturday, February 18, 2006 2:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] problem with
outgoingcallsUnabletocreatechannelof type 'ZAP' (cause
I do not think so by reading the documentation
however I have changed the settings and still get the same error when starting
Asterisk
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
- Message from [EMAIL PROTECTED] on Mon, 20 Feb 2006 07:08:05
+0100 -
One thing to keep in mind with PoE is that you can simply use an injector at the phone location. At least with the 480i you can easily order the phone with the power injector.
Richard
On 2/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Mon, 20 Feb 2006, Soner Tari wrote: I have Data Sheet
I was using G729 with Asterisk 1.07. With the new ability to do packet
loss correction with ILBC, I felt I'd give it a try. The new PLC does
not work with G729. I don't use Speex because my softphone does not
support it.
This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569
Hi!
is there any documentation or simple example around for app_sms.so
to get started with it and do two simple tasks:
1. send a message to an sms-capable phone connected to an ATA
2. receive a message from an sms-capable phone and so something
simple with it, even just write it to
At 03:01 AM 02/20/2006, you wrote:
also, rather disturbingly the linksys press release[1] implies the
942 is PoE only (like the aastra 480i), no external power supply.
Well, the 480i CT comes with a wall wart if you don't want to use POE
and their web site shows the optional PS available. I
Issue resolved, Thanks Digium!
The module slot closest to the bracket
that contains the connector is the last slot. I assumed that the
first slot would be at the bracket.
Silly user error, never assume.
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
o.
What I'm trying to do is accepting a call from pstn, put it
into a queue, while callee is waiting contact some numbers
till one responds, then bridge the two calls.
What I can't manage is jump to next dialplan command soon
after callee enters the queue in order to call other numbers.
I've
On Mon, 20 Feb 2006, Richard Amerman wrote:
One thing to keep in mind with PoE is that you can simply use an injector at
the phone location. At least with the 480i you can easily order the phone
with the power injector.
Aastra does not really make it clear that the 480i is poe _only_. A lot of
As I recall from various firmware versions on the spa3k,
incoming pstn calls are forwarded to asterisk meaning the
incoming call is answered and then forwarded. Later versions
did something a little different.
I can definitely confirm that the SPA3000 here at home forwards the call to
Greetings all,
I'm trying to improve the codec selection on a few of the asterisk boxes we
have to keep the g729 licences free for calls from ATAs that don't support
anything apart from g711 and g729. GSM seems to offer noticably inferior
call quality (at least when using a softphone + decent
Does anyone know how to setup a linear type of queue strategy? By that
I mean that agents will be tried in a particular order and the call will
be routed to them unless they are on the phone or not logged in.
I want a 3rd party app to be able to re-arrange this order on the fly
based on sales
I would use an agi and the local channel with SQL running the logic from
an AGI.
Anybody setup something similar? Any pointers or products
already out there open source or not?
I have done this before.
Thanks,
Steve Totaro
___
- Original Message -
From: Tomislav Parčina [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 20, 2006 10:46 PM
Subject: [Asterisk-Users] Re: Call centre - * hang's up
In article [EMAIL PROTECTED],
Losing an audio packet here or there wouldn't normally be so bad
for fax. Normally I would expect the fax protocol, especially ECM
protocol, to be able to recover from it. However, Asterisk seems
to not work in an ideal fashion for this purpose. Whenever
Asterisk encounters a lost
On Tue, 21 Feb 2006, [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] says...
I think it's a bit of a known fault - the attended transfer function
does not work from the queue system. It would be nice if it did, though.
Hi Paul!
Is there any explanation about this? Is that something that will
On Mon, 20 Feb 2006, Soner Tari wrote:
For 100bit issue, I tend to believe in the data sheet, but I would also like
to hear a first-hand verification. (But I guess we have to wait, because
voipsupply accepts pre-sale orders for now, they don't ship them yet.)
The SPA-942 is $179.95, I would
Jerry Jones wrote:
Turning ECM seems to cause most of my issues with FAX. Most newer
machines have this on by default. However if there is any packet
loss, then when ECM tries to resend and there is additional loss,
then it gets in a loop and everything just fails. Whereas with ECM
off,
Can anyone recommend a good voip provider? Thanks
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At 11:21 AM 02/20/2006, you wrote:
Aastra does not really make it clear that the 480i is poe _only_. A
lot of people are very suprised when I explain to them that the 480i
is poe only.
I thought them made it really clear it was POE only and I was really
surprised when I found the wall wart
Hi everybody,
I sent an e-mail this morning regarding SIP / IAX2
with no ring-back, I now succeeded to pin-point the
problem, here it is, if I dial a provider directly from
extensions.conf I get ring-back, if I dial from an AGI
script I don't get the ring-back but it calls anyway.
I use 1.0.9.
Hello,Digium uses the Dell PE 2850 for their testing. This site says that 3.3V PCI slot. http://www.voip-info.org/wiki/view/Asterisk+hardwareWe are planning on purchasing a Dell PE 2850 and putting a TE205P card on it. However, the needs a 5V PCI slot. Does Dell PE 2850 has a 5V PCI
Hi, i'm having problems with broadvoice incoming calls. I can perfectly place calls but my Asterisk Box is having problems when registering with the SIP Proxy. Sometimes it register and the call gets into asterisk, but without sound (seems to be NAT problems) and sometimes its not possible for
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Bagnall
Sent: Monday, February 20, 2006 11:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] g729 quality at GSM bitrates
I'm trying to improve
I am having an issue where outbound external calls. Calls made using
an analog line (connected to an FXS) route correctly out the trunk
(connected to an FXO). However, when I make a similar outbound call
using a SIP phone the analog phone connected to the FXS rings. I was
having this
Where are you located? That makes a big difference!
PaulH
Melbourne, Australia
- Original Message -
From: CyberSource [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2006 7:37 AM
Subject: [Asterisk-Users] good voip
Can anyone recommend a good voip
I have now set the resyncthreshold to -1, to turn it off. I have also
set the maxjitterbuffer to 2000.
I still received 10 complaints of choppy calls today on Asterisk 1.2.4
versus only 1 complaint on Asterisk 1.07.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
will try that,
thanks bob !
jl2006/2/16, Bob Goddard [EMAIL PROTECTED]:
On Thursday 16 Feb 2006 22:20, Jean-Louis curty wrote: hi, My question is may be a bit out of scope but I don't know where to turn, I have a 1760 with a ccme 24 user licences 1 bri card.
I want to configure a bri card in a
[EMAIL PROTECTED] wrote:
Where are you located? That makes a big difference!
PaulH
Melbourne, Australia
- Original Message -
From: CyberSource [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2006 7:37 AM
Subject: [Asterisk-Users] good voip
Can
hello-
i'm having trouble completing a connection between an older skinny
phone (12sp+) and a soft sip phone (x-lite).
the skinny phone appears to successfully register:
-- Starting Skinny session from 192.168.1.50
Device SEP00D0BA03AB66 is attempting to register
-- Device 'office'
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