Any and all help is appreciated at this point. Thanks for the tip. This is
the only thing I have not been able to get working and ironically it is the
most important.
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mitchel
Constantin
Sent:
Title: Asterisk Installation
http://www.voip-info.org/tiki-index.php
All you will need is a network card in each
system.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Tuesday, March 29, 2005 11:54
PMTo:
Version 0.71 - 30. march 2005.
. Fixed a memory leak, and optimized performance drastically.
Download from here: http://ipswitchBoard.thorben.dk
IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a
FREE Windows.NET application which gives you:
Unattended/attended
Look at www.voip-info.org, you'll have a lot of answers there.
Yves
[EMAIL PROTECTED] wrote:
Dear User
I am new to the ASTERISK not even tried to install it yet just want to know
can I user ASTERISK as a VOIP without any Hardware DIGIUM Card I just want
to install ASTERISK as a IAX
_
Emotikony a pozadi programu MSN Messenger ozivi vasi konverzaci.
http://messenger.msn.cz/
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Hello,
I would like to test the capabilities of the various hardware that I
run AstLinux on:
- Soekris Net4801 (266mhz Geode)
- 1ghz P3
- 1ghz Via C3
- 2.5ghz Celeron
- 3 ghz x 2 Xeon
What I would like to do is use * on the higher end machines to pound as
many calls as possible (probably
Hey,
Most of the time you dont need a big machine to test a small machine.
Just make sure there is no transcoding on the sending end.
I did all the tests you mentioned (Except for the jitter buffer) on a
dual xeon and a via c3.
That took me about 2 months fulltime (its a lot harder than it looks),
Title: Message
I am
having a similar problem, at least trying to access the dynamic user on a second
asterisk machine that pulls from mysql. Are you getting anything in your debug
log? I'm using the same layout as the sample sip users table from the wiki, the
only difference being I added
Ed Greenberg wrote:
Anybody using a Sipura 3000 for FXO with Asterisk?
Mine is working except for one small nit...
When a call comes in from the PSTN, the Sipura answers it and then
passes it on to Asterisk, which plays extension ring tone.
I'd prefer for the POTS line to stay on-hook while the
Hello all,
I had tested about ten VoIP providers, but no one gave me the quality
I was looking for.
My calls, depending the hour of the day, have diferent quality.
Sometimes I felt cuts in the conversation or lost the sound on one of
the end point.
All of the providers I tested had any kind of
Take a look at http://ebills.sourceforge.net/
I uses latex to create nice pdfs.
regards,
klaus
Christopher Snell wrote:
On Tue, 29 Mar 2005 09:53:03 +1000, Rod Bacon
[EMAIL PROTECTED] wrote:
What I would like to know is has anyone found an open-source billing
platform that performs basic billing
Somewhere in the Wiki I read that the best way to adjust the rxgain and
txgain is to dial a type 102 milliwatt test line.
This line is usually found in xxx-958- or xxx-959- ranges.
I'm in area code 323 in Los Angeles.
Does anybody know the test number here??
The number assigned
On Tue, 22 Mar 2005 10:45:42 -0800, Trevor Peirce [EMAIL PROTECTED] wrote:
Mike Dent wrote:
Hi,
the topic says it all really.
Does the Sipura 3000 detect and report UK clid correctly?
Yes it does
Jason
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Hi,
Can anyone provide me with the call progress frequencies and cadences
for the Egyptian PSTN.
I need to make the TDM card zaptel driver to be able to detect busy,
ring, dialtone and congestion tones coming from the PSTN.
Also correct me if I'm wrong, once I get this information I code it into
hi all,
can any 1 pls tell me the context i shld add on sip.conf for
Audiocodec MP108 8 fxs please.
i want to add 2 phone on MP108 port assign extention and dial each other,
can`t get a dialtone only busy signal.
Thnx ppls
Imran
___
Asterisk-Users
Thanks Friends
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yves
Sent: Wednesday, March 30, 2005 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Installation
Look at www.voip-info.org, you'll
Title: Asterisk GLIB_2.0 Error
Hello everybody,
I'm trying to install spandsp_0.0.2pre11 on Debian with a 2.6.6 kernel. I followed every instruction I could find, and compilation did not produce any errors, but when I start Asterisk I get following message:
WARNING[27090]: loader.c:301
Is it possible to have 2 (working) iax2 phones behind port restriced nat?
Interesting you ask, since I just had an incident concerning this. I
have an IAXy and got an IAX hardphone which I tested at home behind
the same NAT. Using IAX soft clients before in this situation, they
would
do you really have
[specialized]
[specialized]
it is twice try removing one entry
Jason
On Wed, 23 Mar 2005 02:37:42 +, Scheda [EMAIL PROTECTED] wrote:
Hey, I'm currently using the GotoIf application to set it so if
certain caller ID's call my number, it will transfer it to my cell
El 30/03/2005, a las 7:40, Dominic Lu escribió:
Hello,
If purchase the codec from GIPS, how difficult it is to implant it in
Asterisk? What the cost will be?
Our company has two Asterisk, one in headquarter and the other in
branch office. We only need the communication between them. We are not
MDS wrote:
I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
months, no problems with my grandstreams. I'm fairly familiar with the
ins and outs of asterisk...
If you are going to use CVS HEAD, you _must_ stay up to date. There have
been a large number of SIP-related fixes in
Mike Miller wrote:
They're both running on 192.168.1.100
Sorry -- I probably should've clarified that.
Yeah. that would have helped! For some reason, they were not only
running on the same machine, but sharing the same port number, which
shouldn't really be possible...
But in any case, if you
[cellphone]
exten = s,1,Flash
exten = s,2,Dial,Zap/2/9729796243
exten = s,4,Congestion
I never done this, but I believe you are missing a final part.
If you do the same thing on a regular phone, the scenario would be this :
1- you are connected with the remote person
2- you hit
Skype uses the wideband version of iLBC. I beleive that this wod be
very interesting in *, but I've also read that the wideband version is
not freely available for use.
Michael
On Wed, 30 Mar 2005 13:49:48 +0200, adria vidal wrote:
El 30/03/2005, a las 7:40, Dominic Lu escribió:
Hello,
If
On March 30, 2005 05:24 am, Obihuan wrote:
My calls, depending the hour of the day, have diferent quality.
Sometimes I felt cuts in the conversation or lost the sound on one of
the end point.
All of the providers I tested had any kind of trouble.
Sounds like the trouble is on your end then.
That's what I thought would happen, but after about an hour and 100 or
so incoming calls, it was still ringing the agents in the order that
they were listed in the agents.conf file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Dennick
Sent:
Hi,
I am trying to compile Asterisk on Solaris. I have tried on a number
of different platfroms, Solaris 8 on sparc and Solaris 10 on X86 and
have run into a number of problems. The voip-info wiki talks about
working installs, but I am not having much luck.
Environment:
gcc 3.4
gmake
ginstall
Hi,
I have strange behavior on outgoing calls (I can receive calls and I can
make outgoing calls to ISDN lines ok (035778421 and 5778421 for instance -
03 is area code).
I use latest bristuffed Ast. under Suse 9.2.
My zapata.conf and zaptel.conf are at the end of mail.
Any help, advice - I
On Wed, March 30, 2005 13:04, Ezabi said:
Hi,
SNIP
Ezabi
Ezabi, (a.o.)
I am assuming that you aren't using a threaded email reader, as you would
be aware of what replying to a message in order to start a new thread -
which I am assuming you did, judging from the results - would do to the
[EMAIL PROTECTED] wrote:
Hey,
Most of the time you dont need a big machine to test a small machine.
Just make sure there is no transcoding on the sending end.
I did all the tests you mentioned (Except for the jitter buffer) on a
dual xeon and a via c3.
That took me about 2 months
Using which strategy? Remember, if you change strategies and reload,
it'll forget where it was and start over.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees
Sent: Wednesday, March 30, 2005 6:43 AM
To: Asterisk Users Mailing List -
I installed the AGI perl library then put the
following script in a file called
/var/lib/asterisk/agi-bin/send_clid.agi,
updated my [incoming] context with exten =
s,1,AGI(send_clid.agi) and did a restart now.
use Asterisk::AGI;
my $agi = Asterisk::AGI-new();
my %input = $agi-ReadParse();
my
Hi,
I'm not the kind of
Linux guru and I was wondering how I could start automatically the Zaphfc
script.
What I mean is that
before starting asterisk, I have to type : make load from the zaphfc directory
in order to load the zaptel driver.
How can I do that
automatically. This can
On Mon, 21 Mar 2005 15:03:14 -0400, Francisco Moreno
[EMAIL PROTECTED] wrote:
Now, when I dial from any of the ext. to '0' It actually matches the
'0', plays the goodbye message, but doesn't hangup but gets directly to
the 'pasvalide' context. Same thing happens when I dial to the ext. 1002
Hi,
What happened to asterisk @ home 0.7 that the dialout-default macro no
longer works?
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Hi Pros,
Please advice whats the purpose of ZAP, if i have softphones and ATA
186 with PSTN trunk, wht ZAP will do ?
do i zap to route calls internal softphone to softphones ?
thnx a lot
Ronny
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I checked and checked and. When there was no hope left. I found out that my
PSTN provider had removed the crc4 without telling.
Everything works just fine...
Thanx for the help.
Rikard
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard
Sent:
Dominic Lu wrote:
Hello,
If purchase the codec from GIPS, how difficult it is to implant it in
Asterisk? What the cost will be?
Our company has two Asterisk, one in headquarter and the other in
branch office. We only need the communication between them. We are not
satisfied with current
SOLVED!
By updating my CVS head just now, my Polycom IP 600 works great!
Thank you!
Mark
MDS wrote:
I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
months, no problems with my grandstreams. I'm fairly familiar with the
ins and outs of asterisk...
I don't think you want both dynamic and defaultip set
But that should not cause what you describe. I hvae seen other issues
with head. Perhaps checkout the latest?
On Mar 30, 2005, at 12:29 AM, MDS wrote:
I googled and googled but could not find anything regarding this
problem.
I have Asterisk
ZAP are channels which connect through hardware boards installed within
the * server.
If only using softphones feel free to not use;-)
On Mar 30, 2005, at 7:32 AM, iMRAN wrote:
Hi Pros,
Please advice whats the purpose of ZAP, if i have softphones and ATA
186 with PSTN trunk, wht ZAP will do ?
do
How would I go about giving sip users multiple contexts? For instance
right now I have them all in: from-sip-internal
Is there a way I can (for sip users) also include say my [dial-911]
[dial-local] and [dial-longdistance].. bearing in mind that I want to
have different sips allowed to do
On Wed, 30 Mar 2005 08:29:39 -0500
Matt [EMAIL PROTECTED] wrote:
Hi,
What happened to asterisk @ home 0.7 that the
dialout-default macro no
longer works?
___
EVERYONE
This is NOT the [EMAIL PROTECTED] list group.
Please go to:
Hey Chris,
What type of phone are you using for testing? I found a big difference
when I switched from a cheap testset to a better phone. The only
problems I get with voipjet is when people talk over each other - but
I'm not sure how to fix that but everything else has been very good.
J
Sure Robert, but you are going to get a lot of cross posted questions
for certain topics (though I agree this one was totally about
[EMAIL PROTECTED] so should have gone to the sourceforge forum).
Post here if it is a straight asterisk question but on sourceforge
Hi everybody,
GIPS have different products, not only codecs:
* Voice enhancements: packet loss concealment algorithms, noise concealment,
jitter buffer, agc, aec (can be used with any codec)
* Codecs: iLbc (free), ISAC, G711 Wideband...
You can include in asterisk voice enhancements and use
Its a very very bad idea to do this on production boxes. Especially if
you are trying to see how far you can go, and then you cross that tiny
border :)
Your production calls will not like an idle cpu% of 0% and a load of 500.
zoa,
Bicom Systems wrote:
[EMAIL PROTECTED] wrote:
Hey,
Most of the
Richard Reina wrote:
I installed the AGI perl library then put the
following script in a file called
/var/lib/asterisk/agi-bin/send_clid.agi,
updated my [incoming] context with exten =
s,1,AGI(send_clid.agi) and did a restart now.
use Asterisk::AGI;
my $agi = Asterisk::AGI-new();
my %input =
On Thu, 24 Mar 2005 14:19:20 +, Gavin Hamill [EMAIL PROTECTED] wrote:
Hullo :) Can someone help me untangle a bit of a mess?
I'm trying to set up a demo * server to show off how useful it can be to our
business (as an IVR system and VoIP backup if our ISDN30s fail). I've not
been able to
Produktentwickler(-in) in Java
Ihre Aufgabe:
Produktentwicklung in Java
Anbindung an eine Datenbank (SQL)
Asterisk und VoIP Affinität
Ihr Anforderungsprofil:
Mehrjährige Berufserfahrung in der (Java-) Softwareentwicklung
Erfahrungen in der Umsetzung
Create serveral contexts, e.g. from-sip-group1, from-sip-group2, etc...
Then in that context, include the features you'd like for each group,
and give each sip user the correct context.
Julian J. M.
On Wed, 30 Mar 2005 09:30:16 -0500, Matt [EMAIL PROTECTED] wrote:
How would I go about giving
Well,
I thought about that but wanted to check to see if there was another way...
at the moment I have:
911 calling
local calling
international calling
long distance.
That's only 4.. but there are various combinations 911 and local...
911 and local and long distance... 911 and international
Hi, ALL:
I has installed my chan_h323 channel driver in my *.
my scenario is:
SIP UA = SER(mediaproxy) = Asterisk = chan_h323 = GNUGK = H323 EP
And my UA and EP all support codecs such as alaw ulaw G.729 at least.
I dial from UA behind NAT to H323 EP, and I answer from H323 EP too.
But I can
Matt Schulte wrote:
How do you toggle the realtime cache?
Check in the configs/iax.conf.sample file of a recent CVS download and
it should be in there.
If there were too many fields
in the table, could you foresee this being a problem?
No, because I have lots of extra company
Is there a way I can (for sip users) also include say my [dial-911]
[dial-local] and [dial-longdistance].. bearing in mind that I want to
have different sips allowed to do different things so I can't just do
includes for those in my from-sip-internal.
Just make different context for different
Gustavo García wrote:
Hi everybody,
GIPS have different products, not only codecs:
* Voice enhancements: packet loss concealment algorithms, noise concealment,
jitter buffer, agc, aec (can be used with any codec)
* Codecs: iLbc (free), ISAC, G711 Wideband...
You can include in asterisk voice
Looks interesting,
From the FAQ it looks like a 'metered' plugin for CDRs is coming but
not available yet. Is this out of date or am I missing something?
Of course you could just do the translation yourself from what I read...
On Wed, 30 Mar 2005 12:24:23 +0200, Klaus Darilion
[EMAIL
Wow, that did not take long.
As with the current case before the US Supreme Court about file sharing
and music copying, I am just writing software. What people do with the
software is not under my control.
My SS7 channel, app, stack or what ever, will be written from scratch in
C++. If it just
I compiled and installed cbmysql.From the
command line if I do a show applications should I see cbmysql in that
list? I guess what I am trying to see is if cbmysql is connected to my
mwqsql. IS there anyway. I was hoping to be able to do it from *
CLI.
just think it the other way round, group your users in different groups
acording to what you want to let them do (ie: Managers, Marketing Employees,
Salesman, etc) then create a context for each group, and include into each
of those contexts what you want to let them do.
hope this helps.
bye,
M.
I know this is slightly round about and probably not recommended...
but could I do an #include for each user... include their sip config
in there as well as:
context=sip-usersphonenumber
[sip-usersphonenumber]
include = theirsettings
include = localstuff
include = 911
?
On Wed, 30 Mar 2005
Right,
I understand the logic behind this, and normally this is what I'd do..
but in this particular instance.. some users are going to have configs
that are different then what others have... I guess the answer is NO..
you can not have multiple contexts on a sip without creating a context
and
Ok, I am still working on getting this PolyCom phone working with Asterisk.
I have been looking all over, but I have not been able to find the username
and password for the web interface on this phone.
I found some site that said it was Polycom and spip, but that does not work.
Anyone else have
Hi!
If I want to use ISDN card for connecting phones to it, that card must be HFC-S,
because of NT mode.
How about if I am connecting ISDN card to the external ISDN phone line (to local
telephone companys s-bus) when card must be in TE mode, do I still have to have
HFC-s card that I could
Matias G. wrote:
yes, ring back tone in Regional (Admin - advanced options in the web config
utility)
(this info is regarding Linksys PAP2 NA but they're almost identical)
Are you suggesting to disable the ring indicator all together on the
ATA? I don't think that would solve our problem.
Ok, I am still working on getting this PolyCom phone working
with Asterisk.
I have been looking all over, but I have not been able to
find the username
and password for the web interface on this phone.
I found some site that said it was Polycom and spip, but that
does not work.
Garrett Nelson wrote:
Ok, I am still working on getting this PolyCom phone working with Asterisk.
I have been looking all over, but I have not been able to find the username
and password for the web interface on this phone.
I found some site that said it was Polycom and spip, but that does not
[EMAIL PROTECTED] wrote:
Its a very very bad idea to do this on production boxes. Especially if
you are trying to see how far you can go, and then you cross that tiny
border :)
Your production calls will not like an idle cpu% of 0% and a load of
500.
I could not agree more with you hence
Garrett Nelson wrote:
Ok, I am still working on getting this PolyCom phone working with Asterisk.
I have been looking all over, but I have not been able to find the username
and password for the web interface on this phone.
I found some site that said it was Polycom and spip, but that does not
Polycom and 456
- Original Message -
From: Garrett Nelson [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 30, 2005 10:24 AM
Subject: [Asterisk-Users] username/password for PolyCom IP500 web interface?
Ok, I am still working on getting this PolyCom phone
[EMAIL PROTECTED] 0.7 was a little buggy so we decided to
release 0.8 It even has a few new features.
AMP 1-10-007a
SpanDSP 0.0.2pre11
vsftpd server
If you have question about installing or configuring
[EMAIL PROTECTED] please read the [EMAIL PROTECTED] Handbook.
http://www.voip-info.org/wiki-Polycom+Phones
It's in the admin guide. User: Polycom; password: 456
Good luck.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garrett Nelson
Sent: Wednesday, March 30, 2005 7:24 AM
To: asterisk-users@lists.digium.com
I did find that in the admin guide, and it does not work. I have tried
Polycom both capitalized and not capitalized.
-Garrett
Polycom/456
Caps are important.
Sean
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Hi,
You can write a GPL'ed SS7. There is nothing protected in the SS7
design. I don't think there ever were any patents. However, if there
were they ran out long ago.
Our non-GPL SS7 (because it is commercial) stack is written as a library
in C. A modified chan_zap links it into Asterisk at
On 23:34, Tue 29 Mar 05, Mitchel Constantin wrote:
Matt,
This isn't meant as a flame, rather I'm curious about what other
people think about the following situation...maybe it's just the
philosopher in me, what happens when the load balancer fails?
Good point. Was thinking the same thing.
Title: CISCO 7970 COLOR FROZEN
Hey Everyone,
I bought a Cisco 7970 Color IP phone. I wanted to reset it back to factory defaults. I went through the sequence of holding down the pound key when the unit is powering on and then when the sequence changes to press 123456789*0#. The phone
Hi!
If I want to use ISDN card for connecting phones to it, that card must be HFC-S,
because of NT mode.
How about if I am connecting ISDN card to the external ISDN phone line (to local
telephone companys s-bus) when card must be in TE mode, do I still have to have
HFC-s card that I could
Hi Dean
I haven't found any limitations as such. It just seems a lot of people have
this impression of [EMAIL PROTECTED] as being a beginners tool. It was
fabulous for
the first couple of weeks, (as I have been running it for a couple of
weeks), but I want to see how I go about migrating up to
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Tuesday, March 29, 2005 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call-ID and Unique-ID
The Call-ID is
-Original Message-
From: Matt Schulte [mailto:[EMAIL PROTECTED]
I thought the TDM was broke on 1750's...?? I could never get passed
that NMI issue.
I don't know about the 1750s. On my 800, loading the TDM modules the first
time causes an NMI, but it seems to be harmless. Wish I
try this sir, Polycom SpIp-
Original Message -
From: Garrett Nelson
To:
Sent: Wed, 30 Mar 2005 10:01:05 -0600
Subject: RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface?
I did find that in the admin guide, and
it does not work. I have triedPolycom both capitalized and
Weird. For some reason when I googled I only got a page in the the UK.
Anyone know if I can buy the 4801 and case and add a X100P card to it?
I notice the bundle it with a Sangoma T1 card, but at the moment I need
to test a single analog line.
Thanks,
Philip
On Mar 29, 2005, at 8:58 PM, Josh
Hi good people,
A local Kenyan company wishes to improve its communication system by
embracing VoIP technology. They currently have a legacy PBX with 17
analogue trunk lines and about 900 extensions. Going by the tender
document, the main features they are looking for include:
01) Converged
I have a T1 going into *, SIP phones Grandstream Polycom IP500.
Everything works great, but when I use the monitor command, or use IP
Switchboard to record a call, the call has really loud static, and you
can only make out maybe 1 or 2 words spoken. I have tried the IN-OUT,
and combined wav
I am looking for input on what an SS7 interface to Asterisk
should look like and what it will need to be of any use.
If you don't want to help then don't whine and complain about
how you don't need SS7. All comments made in jest are welcome; points
will be awarded for cheekiness and good
Bicom Systems wrote:
[EMAIL PROTECTED] wrote:
Its a very very bad idea to do this on production boxes. Especially if
you are trying to see how far you can go, and then you cross that tiny
border :)
Your production calls will not like an idle cpu% of 0% and a load of
500.
I could not agree more
How do I embed a newline into a sms message using the sms originate in * ?
Julian.
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try this sir,Polycom SpIp
-
Tried that, didn't work.
Is my phone just messed up? Is there way I can change that password through
the phone itself? Is there a way to reset the phone to factory settings? I
know how to
We recently configure an asterisk server to use with an VoIP provider
to make calls to a PSTN. We use (voipjet, nufone, diamond)
We feel that we haven't got the quality that we hope. Sometimes our
calls gets mute, or we feel communication cuts on our phone calls.
We have got an QOS
Philip Trauring wrote:
Weird. For some reason when I googled I only got a page in the the UK.
Anyone know if I can buy the 4801 and case and add a X100P card to it? I
notice the bundle it with a Sangoma T1 card, but at the moment I need to
test a single analog line.
Thanks,
Philip
Philip,
I
On Wed, 2005-03-30 at 19:10 +0300, [EMAIL PROTECTED] wrote:
How about if I am connecting ISDN card to the external ISDN phone line (to
local
telephone companys s-bus) when card must be in TE mode, do I still have to
have
HFC-s card that I could forward incoming calls from pbx to phone(s) or
On Wed, 2005-03-30 at 19:10 +0300, [EMAIL PROTECTED] wrote:
If I want to use ISDN card for connecting phones to it, that card must be
HFC-S,
because of NT mode.
Correct.
How about if I am connecting ISDN card to the external ISDN phone line (to
local
telephone companys s-bus) when card
Dunno if this matters at all but before embarking on a new project, you
might want to have a look at this:
http://www.openss7.org/
Maybe the license isn't open enough. I am but a poor peasant boy :)
Race Vanderdecken wrote:
Greetings All,
I am looking for input on what an SS7 interface
There has been some discussion about this. Apparently true Digium X100p
cards will work 3.3 volts, but some clones or other variety of X100p run at
5 volts and do not work.
check out ASTLinux if you are interested in the Soekris.
-Nate
-Original Message-
From: Philip Trauring
Steve Underwood wrote:
Gustavo García wrote:
Hi everybody,
GIPS have different products, not only codecs:
* Voice enhancements: packet loss concealment algorithms, noise
concealment,
jitter buffer, agc, aec (can be used with any codec)
* Codecs: iLbc (free), ISAC, G711 Wideband...
You can
Wilson Pickett wrote:
Quoting the wiki at http://www.voip-info.org/wiki-Asterisk+cmd+Sms
appended to the end. The telco can define a default sub address (9 in
the UK) which is used when the extra digit is not appended to the end.
It says there's a default anyway. Note smsq doesn't send one (I
It is...
Polycom
456
The setup for using new confs and app files is done through the phone
anyway. Just setup the FTP server and your files.
Then at least you should be able to get the latest app file son the
phone to ensure it works right, even if not configured correctly.
W
David,
The Makefile in your zaphfc directory contains zaptel and zaphfc modprobe's for
different systems (2.4 or 2.6 kernel, etc). Add the lines for your system to
the asterisk startup script.
eg:
#! /bin/sh
/sbin/modprobe zaptel
/sbin/insmod /usr/src/bri-stuff.0.1.0-RC4a/zaphfc/zaphfc.o
Hi
1. I wonder Asterisk can do this (refer to the following diagram) or not ?
(Can I make a call from the SIP phone to the normal phone )
2. Is the Asterisk server 2 called the PSTN Gateway ?
3. What are the hardware that I need to do that ?
Hope that anyone can help me in this
On Wednesday 30 March 2005 11:16, TC wrote:
I am looking for input on what an SS7 interface to Asterisk
should look like and what it will need to be of any use.
If you don't want to help then don't whine and complain about
how you don't need SS7. All comments made in jest are welcome;
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