Does any one knows of an Windows based SIP video phone???... Thanks...
-----Original Message-----
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Sent: Tuesday, January 11, 2005 9:27 AM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users Digest, Vol 6, Issue 142
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Today's Topics:
1. How to mark a user for a conference (Jagan Mohan)
2. Re: fax e-mail spandsp (Nils Segerdahl)
3. Re: Vmail.cgi - "Hrm, can't seem to open
/var/spool/asterisk/voicemail .... (Frank Kostin)
4. Re: Analogue RAS Server (Niksa Baldun)
5. Re: Analogue RAS Server (Paradise Dove)
6. Zaptel config (ismaelg)
7. Re: Zhone channel bank issues (James Freire)
8. Re: Weir long distance behaviour... (Francois Meehan)
9. RE: Generic modem question (Rich Adamson)
10. Re: Zaptel config (Tzafrir Cohen)
11. RE: asterisk one number service (Eric Hall)
12. internal caller id on analog phones connected to zap
(Shoval Tomer)
13. sip to h.323 (sai latha)
14. Re: Vmail.cgi - "Hrm, can't seem to open
/var/spool/asterisk/voicemail .... (Jon Radon)
----------------------------------------------------------------------
Message: 1
Date: Tue, 11 Jan 2005 17:13:57 +0530
From: Jagan Mohan <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] How to mark a user for a conference
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=US-ASCII
Hi All,
I would like to mark a user so that all users other than marked
user hear music-on-hold till the marked user joins the conference.
I took a look at
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe, but could not get
sufficient info.
I'm using meetme for conferencing.
Could anyone point me to a url which has the configuration details
using meetme.
Thanks,
Jagan
------------------------------
Message: 2
Date: Tue, 11 Jan 2005 12:55:57 +0100
From: Nils Segerdahl <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] fax e-mail spandsp
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Matt Riddell wrote:
> Brian Dingman wrote:
>
>> Anyone care to pass on a makefile that works. This is what my
>> makefile.rej looks like:
>
> [SNIPPED]
>
> Really it's not that hard. Open two console windows. In one open
> that patch. In the other open the Makefile.
>
> If you look at the patch you can see what lines need to go into the
> Makefile and where. (the + symbol means add this line, and the lines
> without +'s show what is before and after the section you need to
> change).
>
> If you have any problems, drop me a line off-list and I'll help you
> out (but it's worth your while to at least have a try).
Hi,
Remember that make requires that the indentation in the Makefile is done
with tab and not spaces.
I you cut and paste there is a risk that the indentation is converted to
spaces.
/Nils
--
Nils Segerdahl
----------------------------------------------------------------
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Upsala Science Park, 751 83 Upsala Mobil: (+46) (0)703 55 65 03
http://www.upsys.se Fax: (+46) (0)18 56 80 49
----------------------------------------------------------------
------------------------------
Message: 3
Date: Tue, 11 Jan 2005 04:12:53 -0800 (PST)
From: Frank Kostin <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Vmail.cgi - "Hrm, can't seem to open
/var/spool/asterisk/voicemail ....
To: Mike Dent <[EMAIL PROTECTED]>, Asterisk Users Mailing List -
Non-Commercial Discussion <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Hi, Just doing a "chmod" OK
Halas, not a specialist in cgi and/or perl how to run that automatically
into script preferably for specific box b4 list msg's
Anyone really smart could help ?
Thanks
Mike Dent <[EMAIL PROTECTED]> wrote:
Yes, its the permissions on the wav/gsm files:-
-rwx------ 1 root root 330 Nov 16 23:48 msg0000.gsm
-rw-r--r-- 1 root root 231 Nov 16 23:48 msg0000.txt
-rwx------ 1 root root 3244 Nov 16 23:48 msg0000.wav
-rwx------ 1 root root 385 Nov 16 23:48 msg0000.WAV
-rwx------ 1 root root 13794 Nov 16 23:51 msg0001.gsm
-rw-r--r-- 1 root root 216 Nov 16 23:51 msg0001.txt
-rwx------ 1 root root 133804 Nov 16 23:51 msg0001.wav
-rwx------ 1 root root 13646 Nov 16 23:51 msg0001.WAV
-rwx------ 1 root root 2310 Nov 17 09:41 msg0002.gsm
-rw-r--r-- 1 root root 216 Nov 17 09:41 msg0002.txt
-rwx------ 1 root root 22444 Nov 17 09:41 msg0002.wav
-rwx------ 1 root root 2336 Nov 17 09:41 msg0002.WAV
-rwx------ 1 root root 20460 Nov 18 11:48 msg0003.gsm
-rw-r--r-- 1 root root 217 Nov 18 11:48 msg0003.txt
-rwx------ 1 root root 198444 Nov 18 11:48 msg0003.wav
-rwx------ 1 root root 20210 Nov 18 11:48 msg0003.WAV
they are not readable by the web process. Anyway I have not fixed it
yet, so please let me know if you do.
Mike
On Mon, 10 Jan 2005 08:00:13 -0800 (PST), Frank Kostin
wrote:
> Hello everybody,
> I was trying to install a web interface to my Voice Mail, Vmail.cgi
> I can log on it, list messages, but no play with the following error msg;
>
> "Hrm, can't seem to open
> /var/spool/asterisk/voicemail/default/234/INBOX/msg0001.WAV"
>
> Remark: playing the message msg0001.WAV directly OK
> Any smart guy up there could help ?
> Thanks,
>
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>
>
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Message: 4
Date: Tue, 11 Jan 2005 13:32:58 +0000
From: Niksa Baldun <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Analogue RAS Server
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
I don't think it's possible. Asterisk would have to emulate analog
modem, and I believe that feature is not (at least yet) implemented.
Daniel Niasoff wrote:
> Hi,
>
>
>
> Does anyone have any idea how to set up Asterisk so that it can act as
> an Analogue Remote Access Server. I've looked around and as far as I
> can see it will only act as an ISDN Ras server.
>
>
>
> Thanks
>
>
>
> Daniel
>
>------------------------------------------------------------------------
>
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Message: 5
Date: Tue, 11 Jan 2005 16:38:25 +0330
From: Paradise Dove <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Analogue RAS Server
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=US-ASCII
> I don't think it's possible. Asterisk would have to emulate analog modem,
does anybody know if there ia any works on emulating analog modems
(not specially to work with asterisk).
something like Steve's spandsp for fax.
------------------------------
Message: 6
Date: Tue, 11 Jan 2005 14:10:11 +0100
From: ismaelg <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Zaptel config
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello all,
I am having a lot of problems with zaptel channels,
I have got an TDM02B, and I don't know how setup /etc/zaptel.con and
/etc/asterisk/zapata.conf for use it on asterisk.
Some one could help me with this configuracisn?
My problem is about the type of signalling
Thanks,
Regards.
Ismael Gil.
------------------------------
Message: 7
Date: Tue, 11 Jan 2005 08:13:03 -0500
From: James Freire <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Zhone channel bank issues
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=US-ASCII
Hi Michael,
You might want to check the voltage settings on the FXS side of
things. Also, are you using the correct signalling? (ground start,
loop start, etc.)
In the Zplex users guide, on page 41 you will see 2 sections on TTLP
and RTLP. That might be of some help to you.
Hey... You have caller ID working on that thing??? How did you do that?
Let me know if you need a PDF copy of the manual
-James
On Mon, 10 Jan 2005 20:55:13 -0500, Michael Lyszczek
<[EMAIL PROTECTED]> wrote:
> On Mon, 10 Jan 2005 12:51:49 -0500, Michael Lyszczek
> <[EMAIL PROTECTED]> wrote:
> > Anyone have any issues like this....I am fwding broadvoice to zaptel,1
> > with my t100p and the t1 goes to a zhone zplex10b.. I can ring
> > extension 1, which is pair 1 of the channel bank, but it doesnt
> > recognize offhook and it keeps ringing the phone after I pick up.
> > Also, its like each ring is like a seperate call as far as the
> > callerid history goes. Anyone have any ideas?
> > Michael Lyszczek
> >
> _______________________________________________
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------------------------------
Message: 8
Date: Tue, 11 Jan 2005 08:15:53 -0500 (EST)
From: "Francois Meehan" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Weir long distance behaviour...
To: "Wilson Pickett" <[EMAIL PROTECTED]>, "Asterisk Users
Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain;charset=iso-8859-1
Hi Wilson,
I had both features enabled in my zapata.conf file, I will try disabling
the callprogress see if it makes a difference, what troubles me is that I
have no problems with local calls, what could be the difference with long
distance one?
I am from Quebec, Ile-Perrot near Montreal.
Regards,
Francois
>> There is a strange behavior, when we do long distance calls, it keeps
>> ringing on our end, remote callee answers the call but hear nothing.
> Look up callprogress and busydetect
>
> are you in France by any chance?
>
> Look here also
> http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf
> _______________________________________________
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>
Random Thought:
---------------
Business will be either better or worse.
-- Calvin Coolidge
------------------------------
Message: 9
Date: Tue, 11 Jan 2005 07:24:18 -0600
From: Rich Adamson <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Generic modem question
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
> > Yes, you can buy a clone. Yes, it may work currently (although I
wouldn't
> want to guess for how long). Also, the ?>?>>
> > actual cards that the X100Ps are based on have stopped being produced by
> Intel, so you're out of luck as far as a
> > replacement goes in 6 months time.
>
> I though that the X100P were a tigerjet chip? I'm not looking at one
right
> now but I've seen the Tigerjet branding on the real X100Ps and also on the
> TDM400 board too. Actually I have a couple of branded tigerjet "telephone
> gateway" (or something) cards that are identical [in appearance] to X100Ps
> (although I remember that in zaptel they were identified as generic) - I'm
> not using them now but they seemed to work fine.
>
> > Don't forget that the impedance on the X100P (or clone) is 600Ohms so
you
> won't be able to use it without echo outside
> > of the United States.
>
> Thats very interesting - I've certinly had echo annoyances (not major
> problems - echo canel got rid of it after a second or so) and I put it
down
> to bad quality telephone lines (probably true too).
I believe one can characterize the TigerJet name as the pci controller
chip, but the card has several other chips as well. My x100p card has
a heatsink glued on top of one of the chips so I can't see the actual
part number; I believe its the Tigerjet chip however.
------------------------------
Message: 10
Date: Tue, 11 Jan 2005 15:44:17 +0200
From: Tzafrir Cohen <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Zaptel config
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=utf-8
On Tue, Jan 11, 2005 at 02:10:11PM +0100, ismaelg wrote:
> Hello all,
>
> I am having a lot of problems with zaptel channels,
> I have got an TDM02B, and I don't know how setup /etc/zaptel.con and
> /etc/asterisk/zapata.conf for use it on asterisk.
>
> Some one could help me with this configuraciC3n?
> My problem is about the type of signalling
We wrote a simple script to do just that:
http://updates.xorcom.com/genzaptelconf
Only tested on Rapid and Debian, but should generally work elsewhere
genzaptelconf -sdv
--
Tzafrir Cohen +---------------------------+
http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
mailto:[EMAIL PROTECTED] +---------------------------+
------------------------------
Message: 11
Date: Tue, 11 Jan 2005 08:35:37 -0500
From: "Eric Hall" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] asterisk one number service
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
I have it setup to dial my sip phone and my cell at the same time. Is
this what you are looking for?
If so just add & after your dial sip command
(sip/123456789&zap/g1/6145551212)
This works for me
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ashling
O'Driscoll
Sent: Tuesday, January 11, 2005 5:47 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk one number service
I wonder does anyone have any thoughts or can give me some direction on
the following:
I have an asterisk testbed environment set up. My task is to make a
personal number service available whereby users would be given one
number (perhaps a voip number) and this number would enable them to be
reached via the pstn, pots, gsm etc....
Does anyone have ideas where I could start looking at sites to research
this or how asterisk might fit into this?. It would be great if someone
could maybe point me in the right direction.
Thanks,
Aisling.
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Message: 12
Date: Tue, 11 Jan 2005 16:30:41 +0200
From: "Shoval Tomer" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] internal caller id on analog phones
connected to zap
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="US-ASCII"
Hi,
We've got IAX softphones, GrandStream VOIP phones and zaptel connected
analog phones.
Caller id, internally, works just fine (as long as I use numeric only
callerids) for IAX and grandstream.
Is there a way to have the analog phones' LCD display show the caller
id?
These are plain old regular analog phone, that if I had callerid from my
telco would show on the screen.
thanks
Shoval Tomer,
IT Manager,
SofTov Advanced Systems, Ltd.
Office: +972-3-9230686 ext. 179
Fax: +972-3-9216642
Mobile: +972-54-8000200
------------------------------
Message: 13
Date: Tue, 11 Jan 2005 06:26:57 -0800 (PST)
From: sai latha <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] sip to h.323
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED], [EMAIL PROTECTED]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii
Hello,
Happy New Year
where u r downloaded the asterisk server please
tell me.Iam searching the asterisk server site in
google but i dint get this server u please tell me the
site for me
Is only for sip to sip or sip to h.323 please tell
me
Thank u
Bye
Sailatha
------------------------------
Message: 14
Date: Tue, 11 Jan 2005 09:27:06 -0500
From: Jon Radon <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Vmail.cgi - "Hrm, can't seem to open
/var/spool/asterisk/voicemail ....
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=US-ASCII
This issue is well documented.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20vmail.cgi
On Tue, 11 Jan 2005 04:12:53 -0800 (PST), Frank Kostin
<[EMAIL PROTECTED]> wrote:
> Hi, Just doing a "chmod" OK
>
> Halas, not a specialist in cgi and/or perl how to run that automatically
> into script preferably for specific box b4 list msg's
> Anyone really smart could help ?
> Thanks
--
Is it something someone said, was it something someone said?
------------------------------
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