You'll need to get the kernel source for 2.6.7.
apt-get install kernel-source.2.6.7
John, thanks for that.
Which version of Asterisk are you using? head or stable?
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On Sat, 30 Oct 2004 11:12:39 +0900 (JST), Isamar Maia
[EMAIL PROTECTED] wrote:
With so long distances, there is nothing better than G.729.
And why would G.729 be any better than iLBC or Speex.
In fact, if you have packet loss, G.729 absolutely sucks. iLBC is far
more forgiving and sound quality
Hi,
I am trying to dial out from asterisks to a h323 endpoint. But it's not allowing to
use g723. How can I get it to work in pass-thru mode.
Current scenerio is
H323 Ep1===Asterisk===Dial-Out=H323Ep2
This works with 711 but not with 723.
g723 is there on both the Ep. As
H323
Hi list!
I found an interesting wireless phone product. Tiptel will be selling a
base station for DECT phones that is VOIP capable. The base station comes
in two models, one with the SCCP the other with H.323 protocol support.
The interesting bit is that the base station is VOIP connected but
I have a peculiar problem.
I have installed asterisk
and also g729 (2 channels).
I have a Cisco7940 IP phone
with SIP installed (v6)
and a cisco router 2650xm
which has an isdn bri voice
interface that connects to
a legacy pbx system. Also
I installed a x-lite
to make some tests.
I have configured
Hi All!
I am trying to flash a fxo line from my sip phone during a call, in order to hold or
transfer the call. Could someone please tell me how to do this, if it is possible.
Everything in zapata.conf about transfer is enabled, and my incoming POTS line support
this (I have checked with a
Hi Remco,
I found it on www.tiptel.nl (the websites in other countries do not
mention this model) and the model is called tiptel DECT-Z 600 IP systeem
Yu can find it on www.tiptel.nl - producten - DECT Draadloze
telefoonsystemen - tiptel DECT-Z 600 IP systeem
Can you expand ... I'm hoping to use Asterisk to do SIP to MGCP
mediation ... what does it not work?
I don't know the particulars, because I've never used (or even looked at
MGCP). All I know is that whenever the issue comes up, people here say
that Asterisk does not know how to act as an MGCP
Hi,
On Sat, 2004-10-30 at 02:50, JB Hewit wrote:
Hi,
I'm looking at trying out an IP10S with Asterisk. I'll be recieving a
single unit next week to try out and see what she can do.
It seems to be comparable to a Snom190, but I don't seem to find much
detail online about it with Asterisk.
Michael Giagnocavo wrote:
I think his point is that for a commercial rollout (say, a VSP), IAX is not
practical for all clients right now. It's not strange to have a personal
preference that is technically better but not commercially viable. That's
not an insult, just how things are sometimes.
Exactly.
- Original Message -
From: Michael Giagnocavo [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Friday, October 29, 2004 10:56 PM
Subject: RE: [Asterisk-Users] Suggestion re: SIP/NAT/*
I think his point is that for a
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, October 30, 2004 6:18 AM
Subject: Re: [Asterisk-Users] Suggestion re: SIP/NAT/*
Michael Giagnocavo wrote:
I think his point is that
I have delt with their 3com offerings and yes if you are lucky enough to be
able to use this as a stepping stone solution then its a closed deal (on
3com system)
- Original Message -
From: Jim Van Meggelen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, October 30, 2004 1:00 AM
On Fri, Oct 29, 2004 at 09:50:47PM -0400, Nick Bachmann wrote:
Michael George wrote:
I'm having some trouble with DISA() in a call plan that worked before 1.0.
If
anyone has experience with it, I would appreciate some advice.
Perhaps you could post relavent sections of your dialplan...?
An update on my own question...There is some built in numbers in the Zap channel. I
think that *0 should do a hook flash but nothing happens. What have I missed?
[EMAIL PROTECTED] 2004-10-30 10:50:32
Hi All!
I am trying to flash a fxo line from my sip phone during a call, in order to hold or
Thanks. I'll have to see about that SIP functionality.
[EMAIL PROTECTED] wrote:
I have delt with their 3com offerings and yes if you are
lucky enough to be
able to use this as a stepping stone solution then its a closed deal
(on 3com system)
- Original Message -
From: Jim Van
Hi;
I have a Dialogic Card D160 SC and Trunk Board
TP100B installed on my PC, I configured them and was able to detect the Trunck
Board and the Dialogic Card, but I am not able to ping the TP100B? Is there any
one has idea about special settings to be done for TP100B to be able to ping
it?
Is asterisk capable
of sealing (some amount) of losses that occur on IP based channels before it
routes the Calls to a TDM channel (BRI, E1, etc.) to limit quality loss if IP
loss occurs ?
chris.
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Can someone please help me out here?
On Oct 29, 2004, at 10:37 AM, Roy Sigurd Karlsbakk wrote:
hi
I've written a small AGI thing to allow lots of stuff, including
diverts. If a call is placed to a diverted number, a new call is
initiated from * to that number. Simple. But then, to make billing
I'm trying to config a temp iax connection between two current * boxes.
One is behind a firewall, the other uses a registered IP.
I config'ed the * box behind the firewall to 'register' with the one that
has a registered IP. The registration is occuring and the CLI indicates:
-- Registered
On October 29, 2004 11:49 pm, Chris A. Icide wrote:
Only in the X100P format, and only 2 of them
I have to ask -- why are you running such high-end equipment for a craptastic
FXO device? Don't you find other issues that going to a TDM4xxP or even a
T1+channel bank would fix? I mean I ran an
On October 30, 2004 02:38 am, Benjamin on Asterisk Mailing Lists wrote:
And why would G.729 be any better than iLBC or Speex.
lower conversion latency? less bandwidth?
In fact, if you have packet loss, G.729 absolutely sucks. iLBC is far
more forgiving and sound quality is just as good.
Rich Adamson wrote:
I'm trying to config a temp iax connection between two current * boxes.
One is behind a firewall, the other uses a registered IP.
I config'ed the * box behind the firewall to 'register' with the one that
has a registered IP. The registration is occuring and the CLI indicates:
I have a dual xeon server with 2 MB of ram running latest CVS *
all calls are SIP and mu-law is the default codec for all connections
my cpu power has not ever reach above 30% of its load. everything works fine.
the problem is an echo on caller side of a call. but this is not an
always event,
it
Rich Adamson wrote:
I'm trying to config a temp iax connection between two current * boxes.
One is behind a firewall, the other uses a registered IP.
I config'ed the * box behind the firewall to 'register' with the one that
has a registered IP. The registration is occuring and the CLI
Rich Adamson wrote:
Rich Adamson wrote:
I'm trying to config a temp iax connection between two current * boxes.
One is behind a firewall, the other uses a registered IP.
I config'ed the * box behind the firewall to 'register' with the one that
has a registered IP. The registration is occuring and
we use RHAS and whitebox and are quite happy with it on heavy loaded
boxes. Dunno about analog stuff tho since we don't use it :)
On Sat, 30 Oct 2004 09:57:27 -0400, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
On October 29, 2004 11:49 pm, Chris A. Icide wrote:
Only in the X100P format, and
On Sat, 30 Oct 2004 09:57:27 -0400, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
On October 29, 2004 11:49 pm, Chris A. Icide wrote:
Only in the X100P format, and only 2 of them
I have to ask -- why are you running such high-end equipment for a craptastic
FXO device?
The X100P is not only an
Stewart Nelson wrote:
MCGP is a master-slave protocol. The master is referred to as a
Call Agent, a Media Gateway Controller, or just a softswitch.
This is the role that Asterisk can play. The slave is a Media
Gateway, an MGCP phone, an MGCP ATA, or just an endpoint.
Asterisk cannot presently
Roy,
On Sat, 2004-10-30 at 15:31, Roy Sigurd Karlsbakk wrote:
I've written a small AGI thing to allow lots of stuff, including
diverts. If a call is placed to a diverted number, a new call is
initiated from * to that number. Simple. But then, to make billing
sane, I need to change the
On October 30, 2004 10:37 am, Benjamin on Asterisk Mailing Lists wrote:
The X100P is not only an FXO device. Many folks use it as a Zaptel
timing source only.
Yes you are of course correct; but that then raises the question of why he
wants two in there.
-A.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Herrick
Sent: Friday, October 29, 2004 3:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom IP 500 Config Files - searching
All,
Has
On Fri, 29 Oct 2004 20:13:02 +0200, Wilson Pickett wrote:
I've used them for calls terminating in the US with good results. I
happened to put through a call to Romania today and it seemed the
person was hearing me very much lagged behind. The actual asterisk IAX
figure given was like 80 ms which
On Sat, 30 Oct 2004 09:37:00 +0200 (CEST), Remco Barende wrote:
Hi list!
I found an interesting wireless phone product. Tiptel will be selling a
base station for DECT phones that is VOIP capable. The base station comes
in two models, one with the SCCP the other with H.323 protocol support.
hi
the SIP doc on the wiki said that DTMF inband only worked on G.711. If
I use G.726 and RFC2833 DTMF, will asterisk re-generate the DTMF at the
Zap interface?
roy
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hi list,
anyone have any success getting asterisk to pass message waiting
indicator to a grandstream with SER in the middle as a SIP proxy? I
recently implemented SER between asterisk and my SIP clients and it's
significantly more stable (no more dropped clients) but i haven't been
able to figure
Hello,
I have just bought an IN1002 phone from Integrated
Networks in Chine ( www.integratednetworks.com.cn).
This phone is based on the 50 MHz PA1688 processor like many others and I find I
have quitea lot of delay/latency/echo with the phone when using SIP. I
have noticed this when
On Sat, 30 Oct 2004 10:59:22 -0400, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
On October 30, 2004 10:37 am, Benjamin on Asterisk Mailing Lists wrote:
The X100P is not only an FXO device. Many folks use it as a Zaptel
timing source only.
Yes you are of course correct; but that then raises
I received my TDM11B as I assume everyone currently does with the green FXS
module nearest the bracket (slot 1), and the red FXO card in the far slot (slot 4).
After installing the card and the zaptel modules I believe I have the modules
loaded correctly, modprobe wcfxs returns:
On Sat, 30 Oct 2004 17:35:51 +0200, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
the SIP doc on the wiki said that DTMF inband only worked on G.711. If
I use G.726 and RFC2833 DTMF, will asterisk re-generate the DTMF at the
Zap interface?
I'm pretty sure it does. I am using SIP phones with
- Original Message -
From: Steve Prior [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, October 30, 2004 11:53 AM
Subject: [Asterisk-Users] confusing info from Digium and asteriskdoc
aboutsetup of TDM11B
I received my
If you are in AGI... make your own call log.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Florian
Overkamp
Sent: Saturday, October 30, 2004 10:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Modifying CDR
On Sat, 30 Oct 2004 12:18:20 -0400, Steve Totaro
[EMAIL PROTECTED] wrote:
Yes, it should be four unless you care to move the actual module on the card
to the second slot.
I have fixed this in CVS now. Should be propogated to the website in
a few minutes.
While we do try and test everything,
On Sat, 30 Oct 2004, Michael Graves wrote:
On Sat, 30 Oct 2004 09:37:00 +0200 (CEST), Remco Barende wrote:
Hi list!
I found an interesting wireless phone product. Tiptel will be selling a
base station for DECT phones that is VOIP capable. The base station comes
in two models, one with the SCCP the
As one of the people who introduced both DECT and CT3 into the
Australian enterprise market I'll take a crack at answering this. If
anyone thinks this information is worthwhile I'll add it to the Wiki.
DECT is a cordless phone solution. It can be sold as a stand alone
single handset - single base
On Friday 29 October 2004 05:09 pm, Paul Rodan wrote:
1-way audio problems. At least I think it's one way. We hear the remote
party breaking up. So it would be NuFone's ability to transmit, upload, or
our download bandwidth. We're not having bandwidth issues, we have 4 DS3's
at only about 70%
On Friday 29 October 2004 12:32 pm, Ronald Hartmann wrote:
Good Day list,
I have spent better part of the morning reading through the user group
messages and have found some people stating that they are able to get
the Transfer Button to work on the Snom 190/220
Yes, press the softbutton
On 06:57 AM 10/30/2004, Andrew Kohlsmith wrote:
On October 29, 2004 11:49 pm, Chris A. Icide wrote:
Only in the X100P format, and only 2 of them
I have to ask -- why are you running such high-end equipment for a
craptastic
FXO device? Don't you find other issues that going to a TDM4xxP or
I have had the no line available a couple of times. But so far the
call quality hasn't been bad, but I only make like 1 or 2 calls a day
and they are short.
On Sat, 30 Oct 2004 13:57:08 -0400, steve szmidt [EMAIL PROTECTED] wrote:
On Friday 29 October 2004 05:09 pm, Paul Rodan wrote:
1-way
I understand about the soft button and yes this does work, however I am
trying to figure out if the actual Button (the mechanical one on the
phone)
That says transfer is it possible to get this to work.
~ron
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I too am trying to get asterisk to connect to Lucent iMerge.
Any chance of somebody sending me a ethereal trace of succesful
connection to an iMerge?
My conversation with it very short. I sent a gatekeeperRequest. The
iMerge sends a gatekeeperReject. Two packets. That's it.
I'd love to see
On Sat, 30 Oct 2004, Remco Barende wrote:
Digital Enhanced Cordless Telephones. The system is more or less similar
to GSM (the mobile phone) that all the speech is transmitted digitally.
Also you can have (similar to GSM) unnoticeable switching from one base
station to another. Most
[EMAIL PROTECTED] wrote:
On Sat, 30 Oct 2004, Remco Barende wrote:
Digital Enhanced Cordless Telephones. The system is more or less similar
to GSM (the mobile phone) that all the speech is transmitted digitally.
Also you can have (similar to GSM) unnoticeable switching from one base
station
No those cards are used to connect wirelessly to a base station.
Basically like an 802.11x card connects to a base station.
What Remco means is that he wants someone to release software that would
sit on your pc. All a PCI card to connect to base stations and perform
those functions.
Remco,
We have had some problems registering the firefly with the Asterisk 1.0.2 it
seams that IAX version doesn't match? How to solve this?
Robert
-Ursprungligt meddelande-
Fran: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Pascal C. Kocher
Skickat: den 28 oktober 2004 11:25
Till: Asterisk
Hello Guys
Is Saturday and just to get my Actually poor Asterisk knowledge a
little more rich
How far is IAX to be a Industry Standard I mean like SIP or H.323 ,
IAX seems to be the answer to many many NAT problems in this
Out-Of-Available-IP's World but what does the big guys (cisco , etc)
Hello Guys
Is Saturday and just to get my Actually poor Asterisk knowledge a
little more rich
How far is IAX to be a Industry Standard I mean like SIP or H.323 ,
IAX seems to be the answer to many many NAT problems in this
Out-Of-Available-IP's World but what does the big guys (cisco ,
Hi,
I have * 1.0.0. Everything works well except moh.
I followed the instruction in
http://voip-info.org/wiki-Asterisk+config+musiconhold.conf. I use the
default mp3 from *.
The problem is that the music is really slow. Seems like it didn't get the
right rate to play.
Any one having this
On Sat, 30 Oct 2004 14:37:35 -0600, Voip Business
[EMAIL PROTECTED] wrote:
How far is IAX to be a Industry Standard I mean like SIP or H.323
Frank Miller, the author of the IAX specification document is doing
further work on the specification with the intend to eventually submit
to the IETF. He
I thought this might be of interest to the list.
http://www.convergence.com.pk/iax2/trunked.html
Wasim, you should tell us about those things here. Anyway, keep up the
good work ;-)
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam
[EMAIL PROTECTED] schrieb:
Digital Enhanced Cordless Telephones. The system is more or less similar
to GSM (the mobile phone) that all the speech is transmitted digitally.
Also you can have (similar to GSM) unnoticeable switching from one base
station to another. Most wireless phones in Europe
One could use SCP with certificates for authentication and avoid all the
issues with FTP and it's vulnerabilities.
At 07:55 PM 10/26/2004, you wrote:
Richard wrote:
Hi Kristian,
I'd like to use ftp because of several advantages it has. For
example,
ability to change the time stamp and reload
Hi all!
I'm using asterisk (great!) with an hisax isdn card and it works well, but
during normal phone calls (no voip) there is a delay between my voice
going out and the the voice coming in... this delay increase (starting
with 1 sec and going to 4-5 secs) with the time of the call.
anyone help
Ok, I have an odd problem. I'm running Asterisk 1.0.1 and a setup like
this:
Analog phone -- Adtran channel bank -- Asterisk -- IAX --
Asterisk -- PRI
For local calls it works perfectly. For long distance calls, it creates
an echo. I don't have any problems when doing this with local or LD:
On Fri, 29 Oct 2004, Michael Giagnocavo wrote:
The only thing wrong with RedHat as far as asterisk is concerned is that
they do something goofy with their kernels and all you need do is recompile
a kernel from source. IMHO, you should always compile a kernel for your
specific hardware.
The chart is good, but I think it makes a mistake for iLBC:
Isn't iLBC 13.something kbps?
Also, since iLBC uses 30ms frames (when used with asterisk, at least),
it has slightly lower overhead. Approx 2/3 as much overhead.
(not that I'm a big iLBC fanboy or anything.. -- I still prefer a free
On Fri, 2004-10-15 at 22:54, James H. Thompson wrote:
www.voxilla.com is usually one of the first places to get the new sipura products,
at least this has
been true in the past.
Jim
James H. Thompson
[EMAIL PROTECTED]
If you are in USA go ahead and try them but if you are outside of
In sip.conf, you should have the mailbox=MB# and host=ipofSER.
And in ser.cfg, look for method==NOTIFY and do a t_relay() if in location.
Don
- Original Message -
From: Yair Hakak [EMAIL PROTECTED]
To: Asterisk Users List [EMAIL PROTECTED]
Sent: Saturday, October 30, 2004 10:44 AM
The phone has a web interface. Couldn't you just use an expect script
to change it?
John Baker
Karl J. Vesterling wrote:
One could use SCP with certificates for authentication and avoid all the
issues with FTP and it's vulnerabilities.
At 07:55 PM 10/26/2004, you wrote:
Richard wrote:
Hi
Hi,
I've searched everywhere but I have not been able to get an answer to the
following problem:
I get the following notice (appropriate parts are taken out for security
purposes) after long periods of registration using an spa 2000 with
registration set to 3600 seconds and a proxy failover set
On Sat, 30 Oct 2004, dean collins wrote:
No those cards are used to connect wirelessly to a base station.
Basically like an 802.11x card connects to a base station.
What Remco means is that he wants someone to release software that would
sit on your pc. All a PCI card to connect to base stations
My bad... I thought he was attempting to upload config files for
asterisk systems.
Yes, an expect script would work just fine...
At 11:21 PM 10/30/2004, you wrote:
The phone has a web
interface. Couldn't you just use an expect script to change
it?
John Baker
Karl J. Vesterling wrote:
One could
This is how I do it - I know the 2.6 kernel is supposed to have an
easier way, but I've not seen/read how to do it yet.
That did it for CVS head on a knoppix distro. Thanks!
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Looks like it's still incorrect in the first blue paragraph of the section on
FXO (it's fixed in the second blue paragraph). Also, the last paragraph of that
section twice still calls the channel # 2.
Now on to my next confusion... The section on contexts under dislplans mentions
a context
If the phone is behind a NAT firewall, it
would require extra configuration on the firewall. Depending on the
circumstance, it is not always be possible to make such a change.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Karl J. Vesterling
Sent: Saturday,
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