Am Mittwoch, den 19.09.2007, 11:02 +0530 schrieb Jim Boykin:
> Can someone suggests a good and resonable cost voip provider with
> business unlimited plan in USA and allows simultaneous outgoing
> calling.
My experience with business unlimited is that they very well know which
customer uses more t
Hi!
Meanwhile also the location is fixed: it is happening at metalab
(http://metalab.at/) - a place for geeks.
Thus, we meet there at Thursday, 20.9.2007, 19:00 CEST (=local Vienna
time). Metalab is located next to the city hall:
http://metalab.at/wiki/Lage
Metalab is no pub/restaurant. Thus,
Previous mail did not go through. Following up...
Carlos G Mendioroz @ 16/09/2007 13:27 -0300 dixit:
> Hi,
> I'm seeing a problem using AgentCallbackLogin (Asterisk 1.2.16) where
> a call in queue while an agent is logging in results in the agent
> getting the call without properly being logged in
I have the following situation:
SIP Provider -> My asterisk -> SPA2102.
Time to time I recieve a lot of warnings on asterisk CLI with RTP Read too
short, after T.38 session established.
It hapens only in one direction when fax trasmited from provider to SPA2102.
I noticed, that asterisk did no
On Wednesday 19 September 2007 11:43:39 Carlos G Mendioroz wrote:
> Previous mail did not go through. Following up...
>
> Carlos G Mendioroz @ 16/09/2007 13:27 -0300 dixit:
> > Hi,
> > I'm seeing a problem using AgentCallbackLogin (Asterisk 1.2.16) where
> > a call in queue while an agent is loggin
Atis Lezdins @ 19/09/2007 06:05 -0300 dixit:
> On Wednesday 19 September 2007 11:43:39 Carlos G Mendioroz wrote:
>> Previous mail did not go through. Following up...
>>
>> Carlos G Mendioroz @ 16/09/2007 13:27 -0300 dixit:
>>> Hi,
>>> I'm seeing a problem using AgentCallbackLogin (Asterisk 1.2.16)
Hi all,
how can I pickup a call from a queue? Which context parameter do I
have to specify? The context that calls the queues application is
ext-queues. This is what I tried so long (777 is the extension of the
queue I want to pickup from):
exten => _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by $
On Wednesday 19 September 2007 12:11:19 Carlos G Mendioroz wrote:
> Atis Lezdins @ 19/09/2007 06:05 -0300 dixit:
> > On Wednesday 19 September 2007 11:43:39 Carlos G Mendioroz wrote:
> >> Previous mail did not go through. Following up...
> >>
> >> Carlos G Mendioroz @ 16/09/2007 13:27 -0300 dixit:
Atis Lezdins @ 19/09/2007 06:53 -0300 dixit:
> On Wednesday 19 September 2007 12:11:19 Carlos G Mendioroz wrote:
>> Atis Lezdins @ 19/09/2007 06:05 -0300 dixit:
>>> On Wednesday 19 September 2007 11:43:39 Carlos G Mendioroz wrote:
Previous mail did not go through. Following up...
Car
This might be obvious, and well known, but...
If I have 5 ready members and 5 calls in queue at once, Queue seems to
deliver them one by one, blocking while waiting for each member to
answer in turn.
Is there anyway to speed this up (other than setting auto answer ?)
I.e., I would like to have pa
On Wednesday 19 September 2007 13:03:30 Carlos G Mendioroz wrote:
> This might be obvious, and well known, but...
>
> If I have 5 ready members and 5 calls in queue at once, Queue seems to
> deliver them one by one, blocking while waiting for each member to
> answer in turn.
>
> Is there anyway to
Hi,
I am trying to use an Avaya 4602 phone, which I just updated from a
very old SIP software to the latest I could find on avaya's site
(032207). The upgrade went fine and it gets registered on the Asterisk
server.
Now, a couple of glitches, though.
- The phone's web server is not working ... so
Hello,
I've been using for a long time asterisk-perl-0.08 for prepaid card
applications, and I've identified a problem with the last releases of
asterisk-1.2, installed with Trixbox.
The command get_variable() raises a signal SIGPIPE when it is called
(whatever the variable to get).
I made tests
Hi All,
Can anyone tell me how the below can be happening?
-- SIP/205-08439ee0 is ringing
-- SIP/405-084468f8 is ringing
-- SIP/405-084468f8 is ringing
-- SIP/405-084468f8 is ringing
-- SIP/405-084468f8 is ringing
Where, according to A*k, its ringing the same SIP device at t
On Wed, Sep 19, 2007 at 12:25:35PM +0200, Benoît Mérouze wrote:
> Is there any reason this can be fixed in the asterisk-perl-0.10 (not
> yet included in Trixbox)?
> Or is this more an issue from Asterisk (since Asterisk 1.2.19 or 1.2.20)?
Why not give it a shot?
Install asterisk-perl's modules t
Thanks All. However, I was to use it with call files. Will GROUP work there?
~Jim
On 9/19/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
> Try:
>
> http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit
>
> On Wed, 19 Sep 2007, Jim Boykin wrote:
>
> > Is there a way to limit simultaneous c
Adrian,
You are right about last-come-last-known registration. I guess the
phone is sending multiple 180 messages. A SIP debug trace will help
identify this.
Raj
On 9/19/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
> Hi All,
>
> Can anyone tell me how the below can be happening?
>
>-- SIP/20
Hi Guys,
Where can I configure in asterisk if it should use strict routing or
loose routing ?
Thanks.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Band
Hello all,
Has anyone use the Supermicro PDSME+ in combination with the TE110P
successfully?
My experience so far is not very good.
I am running trixbox 2.0 but:
1) with zttool I am getting IRQ Misses. Don't seem to have IRQ conflict,
but I am now running my SATA HD in DMA. And I am not able t
Hi,
Thanks for the answer.
> Yes, res_snmp seems to be sensitive to the specific version of net_snmp.
> I wrote some notes on this - see
> http://www.voip-info.org/wiki/view/Asterisk+monitoring
>
> Basically I ended up installing netsnmp from source, and things started
> working.
what do you mea
Hi there,
I experience the same problem here with asterisk 1.2.24 on
an E1 Line, only 2 of 3 sms are sent, this happens always and
is reproducable.
Did someone find out more about the problem ?
Especially I do not see how I could add a wait to the dialplan
as somebody suggested because there s
On Wed, 2007-09-19 at 13:01 +, kido wrote:
> Has anyone use the Supermicro PDSME+ in combination with the TE110P
> successfully?
> My experience so far is not very good.
If you're having a motherboard compatibility issue with a Digium card
under warranty, you should contact the Digium support
On Tuesday 18 September 2007 18:43:08 Steve Totaro wrote:
> Dovid B wrote:
> > I have a client that has an interesting request. He wants to have
> > people call in to a conference room and all be able to talk however
> > they should not hear each other. There should be admin access to for
> > one u
Dear all
I have one requirement in my company i need video confrancing
now i have codian ( MCU ) but it is possible with Asterisk ???
is there any opensource MCU available on net
Regards
satish patel
-
Need a vacation? G
How would we be able to determine the reasonable cost for an unlimited
plan for an unspecified business? If the business was General Electric,
I would bet they would consider $1M/month very reasonable for unlimited
service. A plan for a corner shop might be reasonable at $19.95/month,
typical for t
Hi Jim,
FWIW we'll be talking about AsteriskNow among other things on the VOIP
Users Conference this Friday at 12:30 PM. If you can't listen or phone
in live, you can check out the archive recordings later:
http://www.VoipUsersConference.org
http://www.VoipUsersConference.org/topics.php - agenda
Hi,
Thanks very much for your reply. I would like to add some information which
may provide a little more clarification on this matter. The LAN network that
we presently have consists of the Asterisk PC and two User PC's (This network
is not connected to the internet). To confirm that Asteri
We use the MAC of the phone (all lower case) with a -a, -b, -c, etc
tacked onto the end of the MAC to specify the line appearance.
One thing you MUST remember is that a sip.conf entry is NOT an
extension. Extensions are totally different from sip.conf entries.
sip.conf entries are DEVICES.
Us
I am missing DTMF digits on a PRI with 1.4.11
I added dtmf logging in logger.conf. I can see that if I
enter "205" I dont see the 2 but all I see is "05".
I have added the dsp.c patch that was recently added to bugs but
that doesnt seem to help my situation.
What can I do to provide more informa
Snip
>>Subject: Re: [asterisk-users] Interesting Conference Request - Can
this
>>be done ?
>>
>>Dovid B wrote:
>>> Hi List,
>>> I have a client that has an interesting request. He wants to have
>>> people call in to a conference room and all be able to talk however
>>> they should not hear each o
It might be simpler to have the person record the message then attach it
to an extension. The audience calls the number/extension and listens to
the broadcast.
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International: (305) 704-7249 Fax: (815)301-9759
Yahoo I
Interesting. I called Digium support. Very friendly guys but they were
unable to tell me if it was a hardware compatibility. They only
suggested an RMA, but it is an incompatibility issue, that won't help.
That is why, I asked for your experience.
Thanks
Jared Smith a écrit :
> On Wed, 2007-0
Dear all
what is softswitch what is difference between asterisk and
softswitch ??
regards
satish patel
-
Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings,
and more!___
Their really isn't many differences. A true softswitch will usually
never speak to an end users device directly.
/b
On Sep 19, 2007, at 10:02 AM, satish patel wrote:
Dear all
what is softswitch what is difference between asterisk
and softswitch ??
regards
satish patel
__
Asterisk is a PBX. A softswitch is more or less a fully featured
telephone switch, usually one that is extensively application-driven
(more so than traditional big-iron switches) and multiprotocol. A
softswitch implements full PSTN interconnection and Class 5 end-user
features, among other thi
On Wed, 19 Sep 2007, Brian West wrote:
> Their really isn't many differences. A true softswitch will usually
> never speak to an end users device directly.
Well, it does serve end-subscribers logically, but no, they are usually
not "directly" connected in any meaningful sense; DLCs and GR.
With signaling gateways to non-SIP telephony interfaces, it generally
corresponds temporally to alerting feedback received on the other side
(i.e. PRI). Because one 180 Ringing message causes only one ringback
tone, achieving multiple ringback tones requires successive
retransmission.
But ev
Kate,
Are the IP interfaces on those NICs on the same subnet?
The simplest way to do this is to pin static routes to various SIP
destinations in your kernel routing table over one interface or the
other, e.g. something like:
route add -host w.x.y.z gw ethX
Then you can send a call to SIP p
Hello all,
I am getting the following error in /var/log/syslog. I have got 2 B410P
cards in this box.
Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes
(z1=0153, z2=00d3) TRANS
Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space
left (z1=0013, z2=001
In article <[EMAIL PROTECTED]>,
Jim Boykin <[EMAIL PROTECTED]> wrote:
> Thanks All. However, I was to use it with call files. Will GROUP work there?
You will have to call out using a Local channel, rather than directly to
a Zap, SIP or IAX channel. You can then perform the GROUP check in the
dialp
How would you play this sound during a bridged call?
Jim Boykin wrote:
> Thanks Matt, just minimal volume to suite comfort noise.
>
> Jim
>
> On 9/19/07, Matt Riddell <[EMAIL PROTECTED]> wrote:
>
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>> Jim Boykin wrote:
>>
>>> Where do I
Erik Anderson wrote:
> All - I've been wrestling with how to best structure the sip device
> accounts on a new asterisk server I'm deploying. All of the sip
> devices (currently only Linksys SPA941s) will reside on the same
> subnet as the server, and I have already set up a decent automatic
> pro
Hi
in asterisk 1.4, I need to cancel the password check and allow users
enter in the mailbox without entering password.
I tried this:
exten => 99,1,Set(LANGUAGE()=es)
exten => 99,n,VoicemailMain([EMAIL PROTECTED],s)
exten => 99,n,Hangup
and this:
exten => 99,1,Set(LANGUAGE()=es)
Maybe I find the problem,
It could be cause debug is enabled. Tomorrow I will change debug to disable
and I will tell you the results.
Regards.
VoipCrazy
2007/9/19, voip crazy <[EMAIL PROTECTED]>:
>
> Hello all,
>
> I am getting the following error in /var/log/syslog. I have got 2 B410P
> ca
On Tue, Sep 18, 2007 at 08:12:36PM -0400, Alex Balashov wrote:
> If you have to resort to such measures to get people to work for you
> in a motivated fashion, you're doing something very, VERY wrong.
Of course they are: they're telemarketing.
Cheers,
-- jr 'rimshot' a
--
Jay R. Ashworth
rrgv wrote:
> Hi
> in asterisk 1.4, I need to cancel the password check and allow users
> enter in the mailbox without entering password.
>
> I tried this:
>
> exten => 99,1,Set(LANGUAGE()=es)
> exten => 99,n,VoicemailMain([EMAIL PROTECTED],s)
> exten => 99,n,Hangup
>
> and this:
> exte
On Wednesday 19 September 2007 19:03:18 Mark Michelson wrote:
> ur VoiceMailMain call. Change it to this
> and see if it helps:
>
> exten => 99,n,VoiceMailMain([EMAIL PROTECTED])
>
> In other words, put the 's' at the beginning of the argument as opposed
> to a separate option.
I think, it
On Wed, Sep 19, 2007 at 11:15:25AM -0400, Alex Balashov wrote:
>
> Asterisk is a PBX. A softswitch is more or less a fully featured
> telephone switch, usually one that is extensively application-driven
> (more so than traditional big-iron switches) and multiprotocol.
Hmmm, Still describes As
Satish,
It depends on your goals. FreeSWITCH is approaching an official
release. Beta 1 is out now and various other tweaks in trunk. But
its really up to you to evaluate your need and compare which fits
your needs. I see them as complementary to each other so its really
up to you.
On 9/18/07, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
>
> > However, in Test #3 the call will fail. Why?
>
> Because Asterisk will attempt to use ulaw in preference to G.729 if
> possible, and the other endpoint offered to support ulaw. The format(s)
> supported by the eventual call destination
On Wed, 19 Sep 2007, Tzafrir Cohen wrote:
> Sounds like both you and bkw know what the difference is but don't
> really know how to explain it...
Well, admittedly, the difference is difficult to reduce from an
essentialist perspective; it's more empirical and descriptive, in
terms of the "s
Atis Lezdins wrote:
> I think, it is deprecated in 1.4, and should work at the end.
That is correct.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
___
Sign up now for AstriCon 2007! Sept
A real softswitch uses TDM
(http://en.wikipedia.org/wiki/Time-division_multiplexing) and Asterisk
uses a psuedo TDM driver (zapata).
Anthony
Tzafrir Cohen wrote:
> On Wed, Sep 19, 2007 at 11:15:25AM -0400, Alex Balashov wrote:
>
>> Asterisk is a PBX. A softswitch is more or less a fully fea
Asterisk isn't a big iron switch.
/b
On Sep 19, 2007, at 11:08 AM, Tzafrir Cohen wrote:
On Wed, Sep 19, 2007 at 11:15:25AM -0400, Alex Balashov wrote:
Asterisk is a PBX. A softswitch is more or less a fully featured
telephone switch, usually one that is extensively application-driven
(more
Actually in 1.4 the s option should be at the end.
on your CLI type "core show application VoiceMailMain"
[Synopsis]
Check Voicemail messages
[Description]
VoiceMailMain([EMAIL PROTECTED]|options]): This application
allows the
calling party to check voicemail messages. A specific mailbox, a
On Mon, Sep 17, 2007 at 06:48:37PM -0500, Lacy Moore - Aspendora wrote:
>What you will find is that asking gui questions in asterisk is about like
>asking about the New York Giants in a New York Jets forums. They are both
>NFL teams and both from New York. However, they both have thei
So, is a pure VoIP switch by definition not a softswitch, despite whatever
other characteristics it might have?
On Wed, 19 Sep 2007, Anthony Francis wrote:
> A real softswitch uses TDM
> (http://en.wikipedia.org/wiki/Time-division_multiplexing) and Asterisk
> uses a psuedo TDM driver (zapata).
The SME+ has a PCI bridge chip btw the 133MHz and 100Mhz slots so you might
want to try moving it to a slot on the other side of the bridge. I believe
that card will work in any of the slots.
-Original Message-
From: kido [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 19, 2007 6:01
Alex Balashov wrote:
> So, is a pure VoIP switch by definition not a softswitch, despite whatever
> other characteristics it might have?
>
> On Wed, 19 Sep 2007, Anthony Francis wrote:
>
>
>> A real softswitch uses TDM
>> (http://en.wikipedia.org/wiki/Time-division_multiplexing) and Asterisk
>>
On Wed, 19 Sep 2007, Anthony Francis wrote:
> IMHO asterisk is a softswitch, it may not be a very high capacity one
> (right now) but it can be and if you don't mind splitting your physical
> trunk calls over multiple machines it works very well as a call routing
> engine, you just need to have
Perhaps I'll be a little more amicable when someone finds a way to bring
at least five or six DS3s into Asterisk.
On Wed, 19 Sep 2007, Alex Balashov wrote:
> On Wed, 19 Sep 2007, Anthony Francis wrote:
>
>> IMHO asterisk is a softswitch, it may not be a very high capacity one
>> (right now) bu
Now see... the fact that it can't handle 6 DS3s doesn't mean it's not a
softswitch... just that it's not a carrier-grade softswitch.
N.
Alex Balashov wrote:
> Perhaps I'll be a little more amicable when someone finds a way to bring
> at least five or six DS3s into Asterisk.
>
> On Wed, 19 Sep
On Wed, 19 Sep 2007, Jay R. Ashworth wrote:
> On Tue, Sep 18, 2007 at 08:12:36PM -0400, Alex Balashov wrote:
>> If you have to resort to such measures to get people to work for you
>> in a motivated fashion, you're doing something very, VERY wrong.
>
> Of course they are: they're telemarketing.
On Wed, 19 Sep 2007, SIP wrote:
> Now see... the fact that it can't handle 6 DS3s doesn't mean it's not a
> softswitch... just that it's not a carrier-grade softswitch.
Well, if it's okay that it's a "small to medium office-grade"
softswitch, then yes, I concur wholeheartedly and retract the
With zaptel that will be impossible, asterisk can do GR303 not sure
how well.
/b
On Sep 19, 2007, at 12:04 PM, Alex Balashov wrote:
Perhaps I'll be a little more amicable when someone finds a way to
bring
at least five or six DS3s into Asterisk.
__
Cesc Santa wrote:
> Hi,
>
> I am trying to use an Avaya 4602 phone, which I just updated from a
> very old SIP software to the latest I could find on avaya's site
> (032207). The upgrade went fine and it gets registered on the Asterisk
> server.
>
> Now, a couple of glitches, though.
> - The phone'
I have two Asterisk Systems. One on of those, when I execute this:
Dial("SIP/teleglobe-007931d0",
"SIP/[EMAIL PROTECTED]|60|oL(400752:6:3)")
... It causes Asterisk to immediately read out the time limit of the call
(66,792 minutes), as soon as the other end answers, even though we are
On Wed, 19 Sep 2007, Brian West wrote:
> With zaptel that will be impossible, asterisk can do GR303 not sure how
> well.
Not well enough to actually use, as far as I can tell.
I *suppose* one can put an AS5800 or something in front of Asterisk
and take a few DS3s there, and thus end up wi
On Wed, 19 Sep 2007, Douglas Garstang wrote:
> Anyone got any idea what might be causing this? Maybe one was compiled
> in 32 bit mode and an Integer value is overflowing? How do I check this?
This is certainly possible, but you might save yourself some time
debugging by posting this to the A
Dear all
Which one would be best for large production enverment
freeswitch or Asterisk and which on would be stable and fuctional ???
Regards
Satish Patel
-
Shape Yahoo! in your own image. Join our Network Research Panel today!
Dear all
I have want to connect 13000 SIP phone with asterisk so it
will be supportable with asterisk and at a time 200 call would be active so
asterisk will support 13000 registration request ???
other wise what about freeswitch it would be fine or SER proxy
fo
Dear Cohen;
But as I remembered, they told you something about the
syslog, I did not understand it, could u please
explain it for me and it relation with our problem?
Regards
Bilal
> Dear Benjamin;
>
> OK friend, things are clear. But now I came to the
> same original issue that you asked about
Running 1.4.11, and during an established SIP call, we often get audio drop
outs if another call comes in. Anyone else see this happening? Incoming
calls ring both some local SIP phones, and also some other servers via IAX
trunks.
--Brent
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Sign up
On Wed, 2007-09-19 at 13:11 -0500, Brent Torrenga wrote:
> Running 1.4.11, and during an established SIP call, we often get audio drop
> outs if another call comes in.
This is usually an indication of some type network problem. The first
step is to figure out *why* the audio sounds bad. I sugg
Hi Jerry,
Please post your Zapata.conf configuration.
Joseph
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Wednesday, September 19, 2007 8:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dtmf issues on PRI and 1
Can anyone recall a bug where asterisk crashes after a callback agent
presses the ackcall key in 1.2?
The last logged item was before safe-asterisk restarted was "delaying
member connect for 2 seconds"
We have seen it two times on a heavily loaded server (1.2.12.1), but
cannot find anything
You may wish to inquire on the developers' list, to people that keep these
types of issues in check by way of the official Digium bug database and
manage the process of fixing them.
On Wed, 19 Sep 2007, Damon Estep wrote:
> Can anyone recall a bug where asterisk crashes after a callback agent
>
Hi,
Is there a list of all the extension states as sent by the
manager interface? (I know I could look them up in the source
but that involves some "backtracing".)
The ones I know are:
-1: no hint for the extension
0: registered && idle
1: busy
4: unreachable, not registered
8: ringing
I've
Atis Lezdins wrote:
> This is available starting from 1.4, see UPGRADE.txt:
>
> * ... The new behavior, enabled by setting autofill=yes in queues.conf
> either at the [general] level to default for all queues or
> to set on a per-queue level, makes sure that when the waiting
> callers are con
Ok, so I'm no rpm expert, but Asterisk 1.4.11 comes with an asterisk.spec
file. Running rpmbuild against it yields errors, the first one being that
the 'Copyright' tag is unknown, and that I need a License tag instead.
Fixed that, and...
Processing files: asterisk-CVS-1
error: File not found: /tm
Understand, however when you are asking about a version that is 1 year
old the dev list answer is always the same, upgrade to the latest 1.4
and test... not an option in this case.
I was just looking for a hint from someone who may have had a similar
experience in the past.
-Original Message-
Hi,
I am trying to set BLF on SNOM phones.
With call-limit=4 in sip.conf and hints in the extensions.conf a call to the
extension correctly shows state as InUse (show hints) and BLF works. When
call is originated from the extension the associated state remains Idle, so
no notification and no BLF.
Martin Roy wrote:
> I just did a clean install of Fedora Core 4 on a PC with a TDM400
> installed. I installed zaptel 1.2.20.1 and asterisk 1.2.24 correctly.
> I did make config for both to have zaptel and asterisk start when I
> boot the computer. My main problem right now is that zaptel do
On Wed, Sep 19, 2007 at 02:54:17PM -0700, Douglas Garstang wrote:
> Ok, so I'm no rpm expert, but Asterisk 1.4.11 comes with an asterisk.spec
> file. Running rpmbuild against it yields errors, the first one being that
> the 'Copyright' tag is unknown, and that I need a License tag instead.
>
> Fix
I'd like to know why the spec file is even included at all then?
I think we'd prefer to build our own, rather than trust someone elses build.
On 9/19/07 3:22 PM, "Tzafrir Cohen" <[EMAIL PROTECTED]> wrote:
On Wed, Sep 19, 2007 at 02:54:17PM -0700, Douglas Garstang wrote:
> Ok, so I'm no rpm exper
On 9/19/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> here because we are actually specifying the IP Address of the Asterisk
> server, but I am willing to try anything to fix this problem. The two user
> pc's are setup on workgroups, so I do not believe that there is a domain
> available t
http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi-box-is-official-named-airave-300451.php
The above is quite interesting, it would be interesting to see if it
uses sip, which I have no reason to believe otherwise, and if it does,
can it be hacked to talk to Asteirsk? In whi
I tried what you sent me and what Barry sent me and it seems to have
fix the problem. I don't know what was wrong but I rebooted the
server after doing both command even if it gave me no messages and
now everything seems to work fine.
Thanks
Martin
On 19-Sep-07, at 6:21 PM, Philipp Kempgen
Hi Michael,
On Sun, 2007-09-16 at 15:10 +0200, Michael Kamleitner wrote:
> thx very much Nasir & Philipp, I'm gonna try this tomorrow when I'm
> back at the server...
>
> however, I wonder if this behavior has changed recently, as I swear
> [ ;) ] that this script has been working before...
yes
Where are MAXFILES or SYSMAXFILES?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Jacob
Sent: Tuesday, September 18, 2007 11:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Linux limits
safe_asetr
I've tried to put a call file to /var/spool/asterisk/outgoing/
to make an outgoing video call, but not succeeded.
I could hear the audio, but no video.
The asterisk version is 1.4.10, with videosupport=yes
The client is eyebeam 1.5.7, with h263 support.
Here are some debug messages.
It shows the
I just did a clean install of Fedora Core 4 on a PC with a TDM400
installed. I installed zaptel 1.2.20.1 and asterisk 1.2.24 correctly.
I did make config for both to have zaptel and asterisk start when I
boot the computer. My main problem right now is that zaptel doesn't
load at startup so
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Hash: SHA1
Martin Roy wrote:
> zaptel correctly and everything seems to be fine from there. I was
> wondering what I have to change to make sure zaptel load correctly at
> startup?
Did you do: chkconfig zaptel on ?
Barry
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Vers
Hallo Group!
My Name is Guenther Sohler and I registred to this group, because
I think asterisk could be interesting for me.
I have got a small server at home running linux.
It does NAT and a Firewall. There is an intranet with my home PC
and a hardware SIP phone.
This SIP phone registers at muj
hi,
and first off all ... welcome!
now it would be handy if you provide us with the name of your phone for
ex 'a linksys spa942' or somthing
kr,
Jan de Coster
Guenther Sohler wrote:
> Hallo Group!
>
> My Name is Guenther Sohler and I registred to this group, because
> I think asterisk could be
My Phone identifies as
USer-Agent: ALL7950 02.09.23
I suppose its AllNet 7950
Hope this helps :)
Original-Nachricht
> Datum: Thu, 20 Sep 2007 08:36:59 +0200
> Von: Jan De Coster <[EMAIL PROTECTED]>
> An: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Betreff:
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