On Sat, 28 Aug 2004, Michael George wrote:
So even with X11 eliminated the sound is still bad to Digium. I tried
another's 1700 number, and it sounded the same, so it's not something unique
to digium and me.
Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work
On Sat, 28 Aug 2004, Andrew Kohlsmith wrote:
Please note that it seems impossible to disable jitter buffer between 20040806
CVS HEAD endpoints. The jitterbuffer numbers in iax2 show channels look
live. The numbers look right (jitbuf 0ms) between 20040806 and RC1
(Nufone). I haven't
I'm using the firefly third-party softphone. However, the same thing happened
when I used IAXphone 2.0.
On 29 Aug 2004 at 7:13, you wrote:
On Sat, 2004-08-28 at 14:01 -0700, Muiz Motani wrote:
How do I go about disallowing transfers when I am running an IAX soft
phone. Is that setting
On Sat, 2004-08-28 at 23:36 -0700, Muiz Motani wrote:
I'm using the firefly third-party softphone. However, the same thing happened
when I used IAXphone 2.0.
I can't offer any real solution because I was only testing the
connection with Firefly, but I got exactly the same symptoms whenever
Dave Cotton wrote:
On Sat, 2004-08-28 at 23:36 -0700, Muiz Motani wrote:
I'm using the firefly third-party softphone. However, the same thing happened
when I used IAXphone 2.0.
I can't offer any real solution because I was only testing the
connection with Firefly, but I got exactly the same
Has anybody tried to integrate amobile phonevia blutooth in
asterisk PBX?
I believe the most things needed are just existing in open source. I found
a "kbthandfree" (http://docs.kde.org/en/HEAD/kdeextragear-3/kdebluetooth/components.handsfree.html)wich
allows to control amobile phonevia an
Jeremy
Don't bother looking for a PSU with a lower current rating, The IAXy
like any electrical device is designed to run at a particular voltage
and consumes a certain amount of power (W) at that voltage.
In simple terms this means that if the voltage is constant and the
design parameters of
I am not sure about vonage but if you go with an IAX provider you can have
multiple simultaneous calls to your DID.
- Original Message -
From: Michael Di Martino [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, August 28, 2004 6:46 PM
Subject: [Asterisk-Users] Newbie
I am
Hi all,
Can you guys recomend a good terminiation partner in Holland ?
/Mike
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Asterisk-Users mailing list
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To UNSUBSCRIBE or update options visit:
So how do I get IAXTel to set qualify=no for me?
On 29 Aug 2004 at 17:46, you wrote:
I believe this was discussed awhile ago, the solution was to set
qualify=no - if anyone knows why it's happening, I'm happily fix it in
firefly
--
While I'm waiting to hear from Digium,
Has anyone ever experienced this?
The IAXy now refuses to ask for an ip. I've plugged it into three
different routers with the same result: the RJ11 small LED lights and
the internal green one flashes, but no ip. Doesn't seem to be asking
DHCP for one.
RE: [Asterisk-Users] Are there any graphic designers on
(Bthis list?
(B
(BMark Paterson wrote:
(B
(BIs there an Asterisk Assistant for linux or windows?
(B
(BIt wouldn't make sense to create Asterisk Assistants for
(BWindoze (it'd probably be called Asterisk Wizards, btw)
(Bbecause
Muiz Motani wrote
(B
(B How do I go about disallowing transfers when I am
(B running an IAX soft phone.
(B
(BYou'll have to ask the author(s) of the softphone. I can
(Bonly tell you about my observations with peering Asterisk
(Bservers which suggest that disabling transfer might solve
On Sun, Aug 29, 2004 at 12:20:49AM -0600, Rich Adamson wrote:
Seems the opencall.org site has basically been unavailable for days/weeks.
Is there another location to obtain the current code?
http://sremington.zapto.org/downloads/asterisk/spandsp/
Just one week ago Seth Remington did send this
Roy Eddleston wrote:
Jeremy
Don't bother looking for a PSU with a lower current rating, The IAXy
like any electrical device is designed to run at a particular voltage
and consumes a certain amount of power (W) at that voltage.
My comment was based on opinions of others that have posted to this
Dear all
I am using ISDN modem by OTE in Athens, the modem has 2 FXS interface which
is connected to X100P cards. The problem is we are not getting the caller id.
And also console print blank or garbage character in incoming call during
execution of NoOP,${CALLERIDNUM}. See the sample
For asterisk I am using more than one sip providers.
The provider in Holland would like to have the international calls like
00 31 20 1234567 but the provider in the US likes it like 0011 31 20 1234567
Can I make a rule in asterisk so that I can dail 00 31 20 1234567 and
asterisk dails 0011 31
Mike,
Sometime ago there was a message from rits they can help you out.
I think it is one of the biggest Voip companies in Holland
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas Sikkema
Sent: woensdag 18 augustus 2004 6:17
To: [EMAIL PROTECTED]
Subject: [OT] RE:
On Sunday 29 August 2004 02:06, [EMAIL PROTECTED] wrote:
On Sat, 28 Aug 2004, Andrew Kohlsmith wrote:
Please note that it seems impossible to disable jitter buffer between
20040806 CVS HEAD endpoints. The jitterbuffer numbers in iax2 show
channels look live. The numbers look right (jitbuf
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote:
If you think that the jitter buffer isn't working right and should fix
this, then please capture debug from the buffer and send over to me.
To do that, in /etc/asterisk/logger.conf edit the debug line to be:
debug =
On Aug 29, 2004, at 7:51 AM, [EMAIL PROTECTED] wrote:
Has anybody tried to integrate a mobile phone via blutooth in asterisk
PBX?
I believe the most things needed are just existing in open source. I
found a kbthandfree
(http://docs.kde.org/en/HEAD/kdeextragear-3/kdebluetooth/
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote:
If you think that the jitter buffer isn't working right and should fix
this, then please capture debug from the buffer and send over to me.
I notice that the timing measurements are still showing wild values at
times - here is a
Those wild times especially occur before any audio is sent. (e.g. while
ringing or pre ringing).
At 17:10 29/08/2004, you wrote:
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote:
If you think that the jitter buffer isn't working right and should fix
this, then please capture debug
Sure, this one works. You need a dringX definitions of the distinctive
rings. Put in each one the output you get in the log for the call
pattern when the phone gets answered.
[channels]
switchtype=national
signalling=fxs_ks
usedistinctiveringdetection=yes
usecallerid=yes
hidecallerid=no
On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote:
On Saturday 28 August 2004 23:01, Michael George wrote:
It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not
running and the Framebuffer has been turned off in /boot/grum/menu.lst. I
have disabled all the
At 17:10 29/08/2004, you wrote:
I notice that the timing measurements are still showing wild values at
times - here is a partial grab of an iax2 show channels:
Lag Jitter JitBuf Format
00020ms 6291456ms ms ALAW
00012ms 6291440ms ms ALAW
00017ms 0004ms ms ALAW
On Sun, Aug 29, 2004 at 07:59:20AM +0200, [EMAIL PROTECTED] wrote:
On Sat, 28 Aug 2004, Michael George wrote:
So even with X11 eliminated the sound is still bad to Digium. I tried
another's 1700 number, and it sounded the same, so it's not something unique
to digium and me.
Would
[EMAIL PROTECTED] wrote:
Why doesn't asterisk clock to the 1000 interrupts per second instead
of the incoming audio? Were there no interrupts available when it
started? Even if you had no card you could use the ztdummy module
and even though that might be off by a bit, surely it'd sound
Hi,
Is there a way to detect if the caller will be
entering an agentless queue? Id like to be able to redirect any caller
who tried to join a queue with no logged in agents, to be redirected to the
groups voicemail. Is this possible? I know I could create a menu and an
announcement for
Joseph Shi wrote:
Does anyone know if there are any reseller for the book VoIP
Telephony with Asterisk in Hong Kong/Asia region? I'm interested in
purchasing the book but the shipping charge to Hong Kong is expensive.
Thanks.
Joseph
Just wait for the simplified Chinese version to appear in
Clayton Smith wrote:
Hi, i'm trying to send some songs over via asterisk, so i'm trying to
get the very best quality possible
i've been using gsm, using sox with a rate of 8000, single channel,
resampled q1, and got some good results, but i'm wondering if there
is at all a better way
I'm
On 30 Aug 2004 at 0:26, Steve Underwood wrote:
[EMAIL PROTECTED] wrote:
Why doesn't asterisk clock to the 1000 interrupts per second instead
of the incoming audio? Were there no interrupts available when it
started? Even if you had no card you could use the ztdummy module
and even though
I feel this is in error some place. If I call a sip
device that is not registered or not connected at the time. Asterisk will send
that call to voicemail to busy not unavailable. Is there a way to correct
this?
Ariel Batista Kasi International - Computer
NetworkingPh: 305-574-6721Fx:
I feel this is in error some place. If I call a sip device that is not
registered or not connected at the time. Asterisk will send that call to
voicemail to busy not unavailable. Is there a way to correct this?
That's the way its always been. Lots of folks believe its not the 'correct'
give a piece of you extensions.conf where it is configured
On Sun, 29 Aug 2004 13:46:37 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
I feel this is in error some place. If I call a sip device that is not
registered or not connected at the time. Asterisk will send that call to
voicemail to
Hey guys,
Im interested in hearing about servers (and thier hardware specs) that
successfully run both asterisk and samba for an office of maybe about
12 extensions (SIP) and about 12 workstations. Im hopeing to not only
replace a traditional PBXs with Asterisk/Linux but to provide a solution
to
I'm using 2 Dell Poweredge 2650 servers with a Wildcard TE410P in each
and a custom linux installation. Works great and even picks up the dual
xeons as quad processors.
Duane Cox
- Original Message -
From: Steve Maroney [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 29,
On Sunday 29 August 2004 15:20, Steve Maroney wrote:
Im interested in hearing about servers (and thier hardware specs) that
successfully run both asterisk and samba for an office of maybe about
12 extensions (SIP) and about 12 workstations. Im hopeing to not only
replace a traditional PBXs
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote:
Also, is are logs of problem conversations already in progress any use to you?
You nailed down the dead audio after 65535ms problem but every now and
again (very very rare) we will have a conversation where the incoming audio
goes totally
On Sun, 29 Aug 2004, joachim wrote:
Those wild times especially occur before any audio is sent. (e.g. while
ringing or pre ringing).
Yeah - because the sender does weird things to the timestamps it
generates. This is the problem that needs to be resolved; the jitter
buffer just shows
On Sunday 29 August 2004 15:52, [EMAIL PROTECTED] wrote:
The jitter buffer makes all its decisions about dejittering based on the
timestamps of incoming frames. There a fundamental expectation that the
sending side is correctly stamping each frame - 20msec, 40msec etc etc.
Right, this makes
Hi,
I thought I'd repost this to the -users list for some background on the
jitter buffer and its workings and remaining issue.s
I'll also pu a little executive summary here at the top:
Where a channel is native bridged to another iax2 channel:
1) Lag is not measured and will usually
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote:
Hmm... I think next CVS update I'm gonna add a bit of code in chan_iax2 that
tries to verify that timestamps aren't getting sent incorrectly. Fun fun
fun. :-)
Its not that the generation is broken. Its that various optimisations and
things
Anyone using Python to write their AGI applications? I find very little
info on it. The wiki has a link to http://sourceforge.net/projects/pyst
but it seems like a dead project. I posted some questions to their mailing
list a week ago and have not seen a reply or other posting. Is there some
other
On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote:
On Saturday 28 August 2004 23:01, Michael George wrote:
It has nothing to do with IAX or GSM. Stop blaming them. My upstream is half
duplex as well (pretty much anyone on DSL or cable is on a half duplex
connection whether
Robert,
Thanks for the reply. I tried that initially and it did not work. To verify
I went back and tried again. It answers and still no sound is heard. From
the CLI I can see it answer and ask for conf-getconfno three times before
executing the hangup... But no sound. Yet if I point the DID
I posteda
problem earlier thinking it was due to a lack of sound card.Several
members stated that you do not need a sound card to play audio to a PRI
channel. I did some further testing and discovered that there is a problem
with call progress tones or signalingon my PRI. Ithink that
the
This is my PRI Debug
info for those interested in this problem:
PMDBRIDGE*CLI Protocol Discriminator:
Q.931 (8) len=39 Call Ref: len= 2 (reference 115/0x73)
(Originator) Message type: SETUP (5) [04 03 90 90 a2]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability:
On Sunday 29 August 2004 16:07, [EMAIL PROTECTED] wrote:
Where a channel is native bridged to another iax2 channel:
1) Lag is not measured and will usually show 0ms. Any other number is an
old measurement from the start of the call
2) The jitter buffer on this machine is not used.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Duane
Sent: 29 August 2004 11:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXy Power in Australia?
Roy Eddleston wrote:
Jeremy
Hi all! I am intrested in the following scheme
My mobile phone - SMS to SOMEONE - Redirect to FWD number - FDW
redirect to my *
There are companies like calluk.com that provide DIDs for free, but
they do not support SMS.
In http://www.voip-info.org/wiki-Asterisk+cmd+Sms they say
Works to ETSI
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote:
On Sunday 29 August 2004 15:20, Steve Maroney wrote:
Im interested in hearing about servers (and thier hardware specs) that
successfully run both asterisk and samba for an office of maybe about
12 extensions (SIP) and about 12 workstations. Im
I am using Festival to synthesize some menu Interaction with a caller
and am having a problem.
What I am working on is a remote callback where I can remotely call in
to an extension, and enter a callback number (or use the CALLERID info)
and a second outbound dialing number to connect to.
I am wondering if it is possible for an extension that is served by a
zaptel device to revert to dial tone once a call disconnects.
For instance, if I make a call to another extension, talk with them, and
THEY hang up, can I then be presented with a new dial tone rather than a
congestion tone?
Hi
(B
(BI am trying to build an Asterisk-1.0-RC2 server with
(BZaptel support and two X100P cards.
(B
(BI was wondering which Zaptel release I should check out to
(Bgo together with 1.0-RC2. I tried the CVS from 18 August
(Bwhich worked fine on another machine with the exact same
(BOS and
Hi all! I am intrested in the following scheme
My mobile phone - SMS to SOMETHING - Redirect to FWD number - FDW
redirect to my * - My * doing smtg
There are companies like calluk.com that provide DIDs for free, but
they do not support SMS.
In http://www.voip-info.org/wiki-Asterisk+cmd+Sms they
if you have anyone questions about your service you can contact us at the
support 978-418-7300
James Jones
Broadvoice Technical Support
From: [EMAIL PROTECTED] on behalf of Ben Wern
Sent: Sat 8/28/2004 4:34 PM
To: Asterisk Users
Subject: [Asterisk-Users]
you are correct. if you don't use sip.broadvoice.com it mess up you sip uri
so the server will reject it. Also you should enable srvlookup it will help
things run better.
James Jones
Broadvoice Technical Support
From: [EMAIL PROTECTED] on behalf of Ed Brady
Hello,
We keep having a really bad static problem on phone calls completed
using a Adtran TA750 and T100P card. The phones are Polycom IP 500
phones and, it occurs across all phones. Not just one.
Everything appears to be on it's own interrupt. I noticed the last time
we did this, we rewired
Is timestamp information calculated purely from the relative timestamps of
each frame of the current incoming stream or is there some degree of RTC
synchronization expected between the two endpoints?
Similarly, are jitter calculations made seperately for each discrete channel
(ie. the IAX level)
Hi all,
I am trying to use a "Siemens optiPoint 300"
IPPhone (H.323 only) with Asterisk (1.0-RC2).
So far I have been using the H.323 channel
included in the tarball (Nufone ?).
I encountered a strange behaviour when I try
to make a call from the IPPhone to my Asterisk box :
=
From my observation, if a call cannot be successfully placed then execution
goes to n+101. So for example if a phone is busy then the call can't be
placed (channel can't be created) and you jump tp n+101 which is typically
voicemail busy. In the case of a phone being offline then the call cannot
Is it possible to make cmd_dial() bridge the audio going out to the network
back to the calling party as soon as dial() starts? Put another way, is it
possible to have the caller hear the outside dialtone and subsequent DTMF
digits? I notice that there is an option 'r' to dial(), thus:
r:
Hello,
How many different codecs support
Asterisk?
Where can I find more detail
information?
I read Digium sell G.729 codec license. Is it
support all differentformatted of G.729 codecs or just one?
Regards,
Balgaa
___
Asterisk-Users mailing
Hi Steve
3COM 4400 PWR 802.3af switches (there's about 6 on E-Bay at the moment) and then use
3cnjvoip-cpod to convert the 802.3af feed to be used with the Cisco 79xx phones (it's
specifically designed to do this). I'm using this setup (also with 3cnj205 wall
switch for traffic
I have been unable to get the asterisk voicemail to work reliably with
broadvoice.
-Original Message-
From: James Jones [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 29, 2004 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Broadvoice
Good Evening
I found your post about this problem. Did you ever find a fix for it? I'm
experiancing the same
problem.
Thanks.
Quoting Steve Creel [EMAIL PROTECTED]:
I have two Adtran 750's connecting our analog phones to asterisk. On
occasion, I get a channel that gets stuck off
Steve Underwood Wrote:
Joseph Shi wrote:
Does anyone know if there are any reseller for the book VoIP
Telephony with Asterisk in Hong Kong/Asia region? I'm interested in
purchasing the book but the shipping charge to Hong Kong is expensive.
Thanks.
Joseph
Just wait for the simplified
[EMAIL PROTECTED] wrote:
On 30 Aug 2004 at 0:26, Steve Underwood wrote:
[EMAIL PROTECTED] wrote:
Why doesn't asterisk clock to the 1000 interrupts per second instead
of the incoming audio? Were there no interrupts available when it
started? Even if you had no card you could use the
I have some problems with my extensions.conf. When a call from pstn comes
in, the call gets put into the [from-fxo] context. From there the caller
is able to dial sip extensions that are included from the [sip-extenions]
context.
When a sip extension is dialed and connected, and then at some
I am still having unacceptable echo on my X101P and twidling with the
rx/tx gain levels and echo settings appears to have no discernable
effect.
Some questions for those who may have more significant electrical
engineering background than I.
1. This impedance match thing ... will it affect this
On 30 Aug 2004 at 10:38, Steve Underwood wrote:
[EMAIL PROTECTED] wrote:
On 30 Aug 2004 at 0:26, Steve Underwood wrote:
[EMAIL PROTECTED] wrote:
Why doesn't asterisk clock to the 1000 interrupts per second
instead of the incoming audio? Were there no interrupts available
We're using the patch and it's working alright aside from the MOH suspension
issue. I've got a C guy in our office I could put on the problem if anyone
can tell me in general what needs to happen. (I tried to figure it out
myself but haven't worked in C in nearly 6 years...)
-Corey
Hi,
Were looking at options for logging agents into the system programmatically
via Perl/PHP and I was wondering if anyone else is doing this and if so,
how. We're using AgentCallbackLogin now but would like to set up a web
interface instead. I've been looking at Asterisk::Manager and didn't
Hi,
I've been trying to get my zaptel x100p cards working for the past
week now. this is what I've done:
installed asterisk:
make clean
make linux 26 (for fedora core 2)
make install
installed zaptel:
make clean
make
make install
did a modprobe zaptel, and wcfxo
got this in
On Sun, 29 Aug 2004, Johannes van Hulst wrote:
For asterisk I am using more than one sip providers.
The provider in Holland would like to have the international calls like
00 31 20 1234567 but the provider in the US likes it like 0011 31 20 1234567
Can I make a rule in asterisk so that I can
Hi,
I am fairly new to asterisk. I am currently testing my first setup.
I've been able to debug most of the problems to make asterisk work
with my hardware setup until this time.
Currently I have the following issue:
Voicemail is running but when I test to leave a voicemail thru my
incoming
On Mon, 30 Aug 2004, Imran Akbar wrote:
Hi,
I've been trying to get my zaptel x100p cards working for the past
week now. this is what I've done:
installed asterisk:
make clean
make linux 26 (for fedora core 2)
make install
installed zaptel:
make clean
make
make install
did a
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