Does Asterisk support Radius accounting?....
-----Mensaje original-----
De: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] En nombre de
asterisk-users-request@lists.digium.com
Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m.
Para: asterisk-users@lists.digium.com
Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs
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Today's Topics:
1. RE: FXO Cards in Australia (Gonzalo Servat)
2. Re: IAX2 connectivity problem (qualify=yes) (Philipp von Klitzing)
3. RE: FXO Cards in Australia (Bryan Nolen)
4. RE: FXO Cards in Australia (Gonzalo Servat)
5. Re: Meetme : Zaptel ztdummy errors (Andrew Thompson)
6. SIP soft phone registration (Steve Murphy)
7. Re: DTMF (Sean P. Robertson)
8. VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk
(Steve Murphy)
9. Re: NuFone International Calls (marrandy)
10. Re: Meetme : Zaptel ztdummy errors
(firedude@shorelinuxsolutions.com)
11. iconnecthere incoming (firedude@shorelinuxsolutions.com)
--__--__--
Message: 1
Subject: RE: [Asterisk-Users] FXO Cards in Australia
From: Gonzalo Servat <gs@webtastic.com.au>
To: asterisk-users@lists.digium.com
Organization: Webtastic
Date: Tue, 18 Nov 2003 00:17:30 +1100
Reply-To: asterisk-users@lists.digium.com
On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote:> Yes, echo problems do still exist, I would suggest testing it before
> going live.
Yeah, so I've heard.
> A couple of points to note:
> 1) Using soft phones seems to compound the issue
So the echo problems are not so bad when using software phones?
> 2) A faster CPU seems to help (I upgraded from a PII300 to a Athlon
> 2200)
> 3) When dialling in/out over the ISDN DTMF won't work (at least I
> haven't seen the patch which purportedly allows it to work) when you
> use the isdn4linux patch.
This is specific to the NetJet card once again, right? Time to go
hunting for the patch...
> 4) Without the above kernel patch you will hear DTMF tones instead of
> the other persons voice when they talk. They don't hear the tones or
> notice anything wrong.
Hmm, not good. Since we want to run a small IVR the DTMF tones are kinda
needed.
> In short, if you can live with the above problems, then you can get
> away with it, from what I know now, I would suggest getting a
> chan_capi capable device, though I haven't tried that yet.
The NetJet is supposedly CAPI capable. Have you tried installing this?
--> http://www.junghanns.net/asterisk/page1.html
> I am about to switch from a netjet card to a TE4xxP card as soon as
> possible, I have a OnRamp 10 being installed tomorrow. This is largely
> to increase the number of incoming lines, but partly to resolve the
> above issues, and also partly to try to resolve long running
> reliability issues which may in fact be related to the TDM400P anyway.
> In which case I will be looking for a T1 channel bank some time soon
> :(
Argh, the fun never stops :)
> PS, I have a brand new Traverse Netjet card available (it was to be
> used for a dial-up ISDN internet account) which is no longer needed.
How much do you want for it? If you can confirm whether the capi channel
driver works with it and reduces the echo problem, I'll be interested.
Thanks for your help.
Regards,
Gonzalo
--__--__--
Message: 2
Date: Mon, 17 Nov 2003 14:28:14 +0100
From: Philipp von Klitzing <klitzing@pool.informatik.rwth-aachen.de>
Subject: Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes)
To: asterisk-users@lists.digium.com
Organization: AEGEE
Reply-To: asterisk-users@lists.digium.com
Hi!
> Try "qualify=3000" which will increase the time between checks..
> Although it sounds like there is more to this problem than just
> increasing the time..
That's not really what I want to do - quality is really bad if you go
above 2000, so it makes sense to keep it at this. I can schedule a
reload, of course, e.g. once an hour, but that can't be the correct way
to do this... is there really no-one else out here who has seen this??
Cheers, Philipp
--__--__--
Message: 3
From: "Bryan Nolen" <reveng@arc.net.au>
To: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] FXO Cards in Australia
Date: Tue, 18 Nov 2003 00:48:34 +1100
Reply-To: asterisk-users@lists.digium.com
Re: these problems with the NetJet Cards: have people spoken with
Traverse about them? I have found them to be most helpful with any
problems (mainly with the Pulsar PCI ADSL cards)
Try talking to guy@traverse.com.au ?
-Bryan
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
> Gonzalo Servat
> Sent: Tuesday, 18 November 2003 12:18 AM
> To: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] FXO Cards in Australia
>
>
> On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote:
> > Yes, echo problems do still exist, I would suggest testing it before
> > going live.
>
> Yeah, so I've heard.
>
> > A couple of points to note:
> > 1) Using soft phones seems to compound the issue
>
> So the echo problems are not so bad when using software phones?
>
> > 2) A faster CPU seems to help (I upgraded from a PII300 to a Athlon
> > 2200)
> > 3) When dialling in/out over the ISDN DTMF won't work (at least I
> > haven't seen the patch which purportedly allows it to work)
> when you use
> > the isdn4linux patch.
>
> This is specific to the NetJet card once again, right? Time to go
> hunting for the patch...
>
> > 4) Without the above kernel patch you will hear DTMF tones
> instead of
> > the other persons voice when they talk. They don't hear the tones
or
> > notice anything wrong.
>
> Hmm, not good. Since we want to run a small IVR the DTMF
> tones are kinda
> needed.
>
> > In short, if you can live with the above problems, then you
> can get away
> > with it, from what I know now, I would suggest getting a chan_capi
> > capable device, though I haven't tried that yet.
>
> The NetJet is supposedly CAPI capable. Have you tried installing this?
> --> http://www.junghanns.net/asterisk/page1.html
>
> > I am about to switch from a netjet card to a TE4xxP card as soon as
> > possible, I have a OnRamp 10 being installed tomorrow. This
> is largely
> > to increase the number of incoming lines, but partly to resolve the
> > above issues, and also partly to try to resolve long
> running reliability
> > issues which may in fact be related to the TDM400P anyway.
> In which case
> > I will be looking for a T1 channel bank some time soon :(
>
> Argh, the fun never stops :)
>
> > PS, I have a brand new Traverse Netjet card available (it
> was to be used
> > for a dial-up ISDN internet account) which is no longer needed.
>
> How much do you want for it? If you can confirm whether the
> capi channel
> driver works with it and reduces the echo problem, I'll be interested.
>
> Thanks for your help.
>
> Regards,
> Gonzalo
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--__--__--
Message: 4
Subject: RE: [Asterisk-Users] FXO Cards in Australia
From: Gonzalo Servat <gs@webtastic.com.au>
To: asterisk-users@lists.digium.com
Organization: Webtastic
Date: Tue, 18 Nov 2003 01:10:06 +1100
Reply-To: asterisk-users@lists.digium.com
I'll be speaking to Guy tomorrow about this. Guy is certainly a helpful
& friendly guy and I'm sure he'll be keen to hear about these echo
problems.
Regards,
Gonzalo
On Tue, 2003-11-18 at 00:48, Bryan Nolen wrote:> Re: these problems with the NetJet Cards: have people spoken with
> Traverse about them? I have found them to be most helpful with any
> problems (mainly with the Pulsar PCI ADSL cards)
>
> Try talking to guy@traverse.com.au ?
>
> -Bryan
>
> > -----Original Message-----
> > From: asterisk-users-admin@lists.digium.com
> > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
> > Gonzalo Servat
> > Sent: Tuesday, 18 November 2003 12:18 AM
> > To: asterisk-users@lists.digium.com
> > Subject: RE: [Asterisk-Users] FXO Cards in Australia
> >
> >
> > On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote:
> > > Yes, echo problems do still exist, I would suggest testing it
> > > before going live.
> >
> > Yeah, so I've heard.
> >
> > > A couple of points to note:
> > > 1) Using soft phones seems to compound the issue
> >
> > So the echo problems are not so bad when using software phones?
> >
> > > 2) A faster CPU seems to help (I upgraded from a PII300 to a
> > > Athlon
> > > 2200)
> > > 3) When dialling in/out over the ISDN DTMF won't work (at
least I
> > > haven't seen the patch which purportedly allows it to work)
> > when you use
> > > the isdn4linux patch.
> >
> > This is specific to the NetJet card once again, right? Time to go
> > hunting for the patch...
> >
> > > 4) Without the above kernel patch you will hear DTMF tones
> > instead of
> > > the other persons voice when they talk. They don't hear the
tones
> > > or notice anything wrong.
> >
> > Hmm, not good. Since we want to run a small IVR the DTMF
> > tones are kinda
> > needed.
> >
> > > In short, if you can live with the above problems, then you
> > can get away
> > > with it, from what I know now, I would suggest getting a
chan_capi
> > > capable device, though I haven't tried that yet.
> >
> > The NetJet is supposedly CAPI capable. Have you tried installing
> > this?
> > --> http://www.junghanns.net/asterisk/page1.html
> >
> > > I am about to switch from a netjet card to a TE4xxP card as soon
> > > as possible, I have a OnRamp 10 being installed tomorrow. This
> > is largely
> > > to increase the number of incoming lines, but partly to resolve
> > > the above issues, and also partly to try to resolve long
> > running reliability
> > > issues which may in fact be related to the TDM400P anyway.
> > In which case
> > > I will be looking for a T1 channel bank some time soon :(
> >
> > Argh, the fun never stops :)
> >
> > > PS, I have a brand new Traverse Netjet card available (it
> > was to be used
> > > for a dial-up ISDN internet account) which is no longer needed.
> >
> > How much do you want for it? If you can confirm whether the
> > capi channel
> > driver works with it and reduces the echo problem, I'll be
interested.> >
> > Thanks for your help.
> >
> > Regards,
> > Gonzalo
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
========================================WEBTASTIC
ABN 30 087 960 619
PO Box 3024
Willoughby North, NSW, 2068
Phone: +61 (02) 9499 2452
Fax: +61 (02) 9499 2618
Email: gs@webtastic.com.au
Web: http://www.webtastic.com.au
========================================
--__--__--
Message: 5
From: "Andrew Thompson" <asteriskuser@aktzero.com>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] Meetme : Zaptel ztdummy errors
Date: Mon, 17 Nov 2003 09:09:30 -0500
Reply-To: asterisk-users@lists.digium.com
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--__--__--
Message: 6
From: Steve Murphy <murf@e-tools.com>
To: asterisk-users@lists.digium.com
Organization: Electronic Tools Company
Date: 17 Nov 2003 07:14:25 -0700
Subject: [Asterisk-Users] SIP soft phone registration
Reply-To: asterisk-users@lists.digium.com
Hello--
I've installed a few X-Lite softphones on windows machines, and am
playing with the settings. I've written before about this, and since
have discovered that the X-Lite has an option to turn off registration,
which I have set, because I bumped into a letter from Mark, saying that
Asterisk doesn't do registrations yet. And, without such on the X-Lite,
I can still dial out, and get calls. And, best of all, no 60-second
repetition of the same error message over and over and over to fill up
the logs and the screens.
My question: What is this feature? What is SIP registration?
murf
--__--__--
Message: 7
From: "Sean P. Robertson" <spr@netxusa.com>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] DTMF
Date: Mon, 17 Nov 2003 09:24:45 -0500
Reply-To: asterisk-users@lists.digium.com
I think that you are thinking of SIP INFO messages if you are expecting
to see something in the SIP messaging. RFC2833 is sent as part of the
RTP packets so you are not going to see a plain text 1,2,3,4,etc in a
trace when using it.
http://www.faqs.org/rfcs/rfc2833.html
Sean
----- Original Message -----
From: "Scott England" <scott@homelan.com>
To: <asterisk-users@lists.digium.com>
Sent: Monday, November 17, 2003 5:58 AM
Subject: [Asterisk-Users] DTMF
> I am trying to connect to a vocal server from an asterisk server. A
> call is received via iax2 to my asterisk server. I then initiate a SIP
> connection to the vocal server. everything works great except dtmf
> doesnt work. A cisco 5300 can connect to this vocal server and do dtmf
> without a problem. I have my dtmf set to rfc2833 in the general
> section of the sip.conf . I can confirm that the channel is in rfc2833
> during the call via show channel. With SIP debug though I dont see any
> event for dtmf. I do see dtmf in IAX though.
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--__--__--
Message: 8
From: Steve Murphy <murf@e-tools.com>
To: asterisk-users@lists.digium.com
Organization: Electronic Tools Company
Date: 17 Nov 2003 07:34:02 -0700
Subject: [Asterisk-Users] VOIP phonesets vs. cheap Analog touch-tone
sets with Asterisk
Reply-To: asterisk-users@lists.digium.com
Hello--
I've been asked an interesting question, and I'm too ignorant to answer
it authoritatively (yet). Can anyone help me?
Question: If I'm going to implement a somewhat small (10-80) phone
system, and I have a choice of using VOIP phoneset (like SNOM or
Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly
what features will I kiss goodbye if I use the cheap analogs?
In other words, what features will a (more expensive) VOIP phoneset
provide, that the analog won't?
I know already that asterisk will give me these features with just plain
analog phones (&zaptel cards, of course): Voice mail, park & retrieve
&
MOH, transfer, agents, and a few others. And, if you get an analog with
a CID built in, you could have that, too? (Haven't tried that yet).
What is the justification for VOIP? just total cost reductions (if the
phone is cheap enough?)? Or are there some nicenesses that only VOIPs
can supply?
murf
--__--__--
Message: 9
From: marrandy <marrandy@chaossolutions.org>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] NuFone International Calls
Date: Mon, 17 Nov 2003 09:43:46 -0500
Reply-To: asterisk-users@lists.digium.com
On Sunday 16 November 2003 12:52 pm, wasim@convergence.com.pk wrote:
Hello Jasim.
> lag just took on a whole new definition :) --
Old mail I stored for the future for when I have time to look at
asterisk
again.
Like now :-)
Basically, I'm going over the scripts/files and resetting and testing.
This
is still a test system, but I thought I would give some feedback.
> in iax.conf put
>
> [nufone]
> type=peer
> secret=yourpassword
> context=WORLD ; <-- this bit originally stated, maynot be necessary
> host=switch-1.nufone.net disallow=all
> allow=ilbc
> trunk=yes
>
> in extensions.conf put
>
> exten => _011.,1,Dial(IAX2/username@nufone/${EXTEN}|60)
I didn't need the context=WORLD
My extensions.conf has the following for international calls. Any
comments
appreciated :-
[globals]
TRUNK=IAX2/user@nufone
[int-trunks]
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:1})
exten => _9011.,2,Congestion
I found I had echo in that I could hear my voice about a third of a
second
later.
It turned out that one of the many sets of scripts I had been testing
had a
high tx and rx gain setting.
When I changed the settings to :-
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
in /etc/asterisk/zapata.conf
the issue went away.
YMMV
> now, jerjer says you don't need CONTEXT in your iax.conf anymore, he
> handles it at his end, but this is a config from donkeys ages ago and
how > we used to do things in the elden days...
>
> - wasim
>
> p.s. for those in the know ... the super secret code name for eeks has
> been herewith changed to farfon (with two dots on the "o" of the
fon)
Ahhh...secrets...What's it got in its pockets ?
Regards...Martin
--
Even God lends a hand to honest boldness.
-- Menander
--__--__--
Message: 10
Date: Mon, 17 Nov 2003 09:58:40 -0500 (EST)
From: firedude@shorelinuxsolutions.com
To: Asterisk-Users Mailing-list <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] Meetme : Zaptel ztdummy errors
Reply-To: asterisk-users@lists.digium.com
HI guys
I do have usb-uhci. How do I build ztdummy? I think once its built I
just have to do a modprobe to load it, I just don't know how to load it.
AJ
--__--__--
Message: 11
Date: Mon, 17 Nov 2003 10:12:22 -0500 (EST)
From: firedude@shorelinuxsolutions.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] iconnecthere incoming
Reply-To: asterisk-users@lists.digium.com
Hi guys
I just registered an incoming number with iconnecthere and I'm trying to
set up incoming calls from icconnecthere on my asterisk server. I took
a
look at john todds sample sip.conf and extensions.conf file but for some
reason my incoming is still not working. At this point I wish to use
iconnecthere merely for inbound calls. Also my asterisk server is
behind
nat. The following is my current sip.conf file or at least the part
that
I see as pertinent here.
[general] register=15555555555:1111@sipauth.deltathree.com/15555555555
context = sip-incoming
The pertinent part of my current extensions.conf is this
[sip-incoming]
exten => _.,1,Goto(home,s,1)
Please note that 15555555555 above has been substituted for my 11 digit
iconnecthere number and 1111 above has been substituted for my
secret/password. I have seen on the mailing list where nat = 1 seems to
be used by some to get this working and I even saw this in john's
sip.conf
file; however it was not in the general section. Can I place this in
the
general section?
Thanks a lot for any and all assistance.
AJ
--__--__--
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