On Mon, 26 Jan 2004, Aaron Martin wrote:
I have Asterisk running with a combination of SIP and H323 clients. I am using the
OH323 module instead of the H323 one.
When the SIP clients ring each other, they can hear a ringing noise in the ear peice
to let them know that the other parties
On Thu, 22 Jan 2004, Jeff Gustafson wrote:
[...]
no, its not necessary required. in this case, check that the contents of
OS79xx.TXT if they match with your current version.
I didn't have that file because I thought it would make things worse.
:) I took the number from Settings -
On Thu, 22 Jan 2004, dkwok wrote:
Just got GS 101 phone and plugged into the network.
Got ip setup however, the following problems arise:
1. when dialing an extension, I cannot further send any key tone to
Asterisk.
2. there is no sound coming from the other end.
[gs]
canreinvite=no
On Thu, 22 Jan 2004, Jeff Gustafson wrote:
Sure: Remove the SIPDefault.cnf.xml file from your TFTP server!
It's not required, and unless you understand what it's good for, it will
keep your phone rebooting in an infinite loop.
Well, it's XMLDefault.cnf.xml on my box. It doesn't
Hi Jeff,
On Wed, 21 Jan 2004, Jeff Gustafson wrote:
Kewl, I was apparently trying to use older chan_sccp code which didn't
work.
Okay... just tried your new code. The phones keep resetting:
To be fair, one should mention that chan_sccp is actually hosted at
On Thu, 15 Jan 2004, Peter Pauly wrote:
Are there any cheap SIP phones (like the Grandstream
for example) that support power over ethernet?
SNOM-105, SNOM-200, and all Cisco phones should support PoE.
What is necessary to support SIP phones in a
Cisco Call Manager environment?
easiest
On Wed, 14 Jan 2004, Jan Czmok wrote:
It looks like chan_sccp is doing something at this pont that upsets the
7920 so that it tries to fall back to SRST mode, before finally
re-registering.
Okay, might be a reason. but what i saw on the display was:
- Registering to Callmanager
-
Hi Jan,
first of all: please don't cross-post!
On Tue, 13 Jan 2004, Jan Czmok wrote:
[...]
SKINNY OffHookMessage
SKINNY SetSpeakerModeMessage
SKINNY OnHookMessage
SKINNY DisplayPromptStatusMessage
SKINNY DisplayPromptStatusMessage
SKINNY DisplayPromptStatusMessage
It looks like
On Tue, 13 Jan 2004, Ray Burkholder wrote:
Will cisco 7910 ip phone compatible with Asterisk? I know
[...]
Cisco's site shows SIP drivers for 7960, 7940, 7912, 7905 only. If you want
to run 7910 in Skinny mode, that may work. I'll leave that up to the
chan_sccp and chan_skinny people.
The
On Tue, 13 Jan 2004, listas iPfone wrote:
I received an unit of the Symbol NetVision Phone and i will test it with asterisk
using H.323 or Skinny , somebody tested this phone with asterisk and can share
experience?
The phone itself is a bit flakey: turns off within 1 minute if it doesn't
On Tue, 13 Jan 2004, Brian Buhrow wrote:
Hello. The Cisco 7905 and 7920 phones are basically the same phone,
with the 7920 having a built-in ethernet switch. Sip and Skinny images
[...]
Umm, the 7920 is Cisco's Wireless phone. It's definitely different from
the 7905, and there's
Hi Jan,
the 7920 is on my todo list for quite a few days now, and I've had
experience similar to yours...
On Sun, 11 Jan 2004, Jan Czmok wrote:
Latest status:
chan_skinny does NOT work with 7920
chan_sccp does WORK with 7920 (!!)
Yup.
One should add that you'd better use the 0.2 release or
On Fri, 9 Jan 2004 [EMAIL PROTECTED] wrote:
[...]
Regarding the email list: A number of people have suggested creating more
email lists. I think this is not a good idea because there will be even
more cross posting than there is now between -dev and -users.
That's a very valid point.
[...]
On Sat, 10 Jan 2004, Arnd Vehling wrote:
is it possible to use Asteriks for translating SIP to H323 and vice versa?
I am looking to implement the following Setup
SIP UAC - SIP-Server - SIP/H323 Proxy - H323 Server - H323 UAC
Basicly i want SIP fones to talk to H323 fones and and SIP Fones
Hi,
On Sun, 11 Jan 2004, Adthrawn wrote:
I'm after two very specific ringtones for the 79xx's...
A dog barking, and a horse either galloping or neighing.
[...]
I do recall, you had to set the sample length to a divisible, something
like 800? And there was a maximum sample length too...
On Mon, 5 Jan 2004, Philipp von Klitzing wrote:
Am I correct by stating that as of now none of the VoIP protocols (SIP,
MGCP, H.323, Skinny) supports such a silent ring feature? Would SIMPLE
solve this? Naturally I wouldn't like those calls to show up in my list
of unanswered calls. For the
On Mon, 29 Dec 2003, H S wrote:
I am living oin Germany and having two ISDN BRI Lines available. Capi
driver!
I need a Sip Gateway and a H 323 Gateway.
About H.323, there should be a full implementation of H.450.
There is no such thing like a full implementation of H.450.
Innovaphone's
On Thu, 18 Dec 2003, Christopher J. Wolff wrote:
I have three cisco 7910 phones connected to * through skinny protocol. When
one of the phones is called, and the phone is ringing, you can hear what's
going on in the room even though the caller hasn't answered. It's crazy and
very hard to
On Sun, 21 Dec 2003, Ludovic Drolez wrote:
We have Cisco 7912 phones, and the doc says that I can create up to four speed
dial buttons on my phone using the Cisco CallManager.
Does anyone knows which protocol is used to configure speed dials (Is it
documented somewhere) ?
Did someone tried
On Tue, 16 Dec 2003, Pavel Zheltouhov wrote:
Does asterisk work with CCM as gateway ?
When I trying call asterisk,I totally can't hear any sound.
When call ohphone - works good.
10.0.1.219 is CCM, 10.0.1.207 asterisk.
[...]
Tested with latest cvs asterisk.
Maybe asterisk h.323 channel
On Wed, 17 Dec 2003, Victor Medrano wrote:
i did with cisco callmanager with smdi integration .
and h323 .
works very well .
You got SMDI working with CCM?
How?
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Asterisk-Users mailing list
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On Thu, 4 Dec 2003, Andrew Gillham wrote:
Well as far as I can tell, the only version I have on the box is 2.4.22-1.
I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux'
symlinked
to that directory in /usr/src.
Are you saying my /usr/include would be skewed? Since I
On Tue, 2 Dec 2003, tony banks wrote:
I have 1 IP 7940 with the following Firmware versions
App Load ID:
P00303011201
Boot Load ID:
PCO303010001
Version
3.1(12.1)
Sounds like a Skinny image to me (and an old one, around CCM 3.1, too.)
You can either get a SIP image from cisco and
On Wed, 19 Nov 2003, Florian Overkamp wrote:
[...]
add a link to the FAQ on the bottom of every message, much like the
subscribe/unsubscribe instructions. Then again, nobody seems to read those
either :-P...
Not sure about what mailing list footer you get, but I do only get this at
the bottom
On Sun, 16 Nov 2003, Josh J. Zuerner wrote:
[...]
For example, if I currently dial 1000400 on my * SIP phone, the MW lamp on legacy X
400 is flipped on by the PBX. If I dial 1001400 on my * SIP phone, the MW lamp on
legacy X 400 is flipped off.
Does this dialing capability already exist?
On Tue, 11 Nov 2003, Matthew Enger wrote:
I have it working on distinguising just the local numbers of our 4 B channels
and the number assigned to the group. I have ordered an '100 in-dial range'
here in Australia and should have it available to me by the end of next week, I
can let you know
On Mon, 10 Nov 2003, Chad Cowan wrote:
This Polycom phone seems to be one of the best on the market for sound
quality and features. I have seen on the list that some people have gotten
the IP 600 to work with Asterisk. Does anyone have the details of how to
get this working i.e. XML phone
On Sun, 2 Nov 2003, Florian Overkamp wrote:
At 15:07 1-11-2003 -0600, you wrote:
Last I checked skinny firmware would try to connect to a host that would
resolve to CiscoCM1
Actually that is just a last-resort. Before that it will try and find the
callmanager by looking for some special
On Wed, 5 Nov 2003, Dave Weis wrote:
On Wed, 5 Nov 2003, Charles Hatchette wrote:
I am new to Asterisk and Digium card implementation issues. My VAR is
strongly recommending using Apple hardware and Yellow Dog Linux for my
telephony project, because of his familiarity with this OS. Is the
On Tue, 14 Oct 2003, Sales wrote:
Can anyone point me to some online documentation showing how to reset a
CP-7960 to factory default settings. I have some that are configured for
Callmanager and I want to get them back to generic default config. Any info
is appreciated.
- unplug ethernet
On Thu, 21 Aug 2003, Low, Adam wrote:
I've been developing all sorts of applications for use on our 79xx handsets but am
having great difficulty with formatting, I just can't seem to be able to produce a
line feed between lines on the stuff actually displayed on the phone. Has anyone
else
On Wed, 13 Aug 2003, Devon Henderson wrote:
[...]
We have agents who work both from home and from the office.
Some agents are
always in the office, some are always at home, and some
alternate between
the two.
[...]
I guess my big question is: is it possible to have extensions mapped
Hi again.
On Mon, 11 Aug 2003, Rainer Jochem wrote:
I've played around a little bit and discovered the following:
with
services_url:
http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php?user=1234amp;pin=1234;
the phone tried to get
GET /xmlservices/vm/index.php?user=1234?pin=1234name=...
On Sat, 9 Aug 2003, John Todd wrote:
The Ciscos should be able to do this via a XML Service using the RTPTx:
and RTPRx: URI schemes. The tricky part is just pushing the RTPRx to the
listening phone.
XML service specifications are here:
On Wed, 13 Aug 2003, James Sizemore wrote:
Leave off the softkey xml tags, This should get you working
You can telnet to the phone and type debug http and you will get better
errors.
While that seems to work on the 7960 SIP firmware, it's not quite
satisfactory. Are softkeys officially
On Fri, 8 Aug 2003, cwitte wrote:
There was a thread a few months ago that tossed around some ideas for
using a cisco phone for intercom or paging. I don't have any ip
phones, and wondered if anyone had any luck getting intercom or paging
to work on the cisco units.
The Ciscos should be
Hi Bruce,
On Wed, 6 Aug 2003, Bruce Ferrell wrote:
[...]
Is there a difference between what asterisk is and a softswitch? Can
someone explain it in small words and phrases for me?
Sure, there is. First of all, * is much cheaper.
But technically, * does much more than a soft switch:
AIUI, a
On Fri, 8 Aug 2003, Dave Cotton wrote:
The x100p does get the CID in France. It is now a question of how to break it down.
I changed callerid.c line 278 to :-
ast_log(LOG_NOTICE, Got this:- %s\n, cid-rawdata);
and the result on August 8 at 10:06 from 0490233081 was:-
File callerid.c,
On Fri, 8 Aug 2003, Maik Schmitt wrote:
[...]
I just tried to use it with our 7960 (sip-version).
I've set the services_url in SIPDefault.cnf to
http://xxx.xxx.xxx.xxx/xmlservices/vm/index.phpamp;user=1234amp;pin=1234;
It didn't work with ?user=...pin= cause the phone then tried to
get
Hi again,
this just popped into my eyes:
On Fri, 8 Aug 2003, Maik Schmitt wrote:
I just tried to use it with our 7960 (sip-version).
I've set the services_url in SIPDefault.cnf to
http://xxx.xxx.xxx.xxx/xmlservices/vm/index.phpamp;user=1234amp;pin=1234;
you mean:
Hi again,
struggling with localization issues (so the script is not German only)
took me a week longer than expected. (Did anybody ever get PHP's gettext
extension working??)
But finally, I've wrapped something up:
On Thu, 24 Jul 2003, Dave Packham wrote:
I would like to see your code...
On Mon, 4 Aug 2003, Luciano Ramos wrote:
The gsmfr codec works great with asterisk,
all the g7* codec don't.
huh??
g711 works fine, too (both A and ยต). But maybe you don't count them as
codecs...
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Asterisk-Users mailing list
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On Wed, 30 Jul 2003, Benjamin Miller wrote:
I have not had time to complete an Unified Messaging component to
voicemail, but I would see this as an admiral goal. Most modern
voicemail systems have some kind of way to delete or mark the voicemail
as read when the message is deleted or read
On Fri, 25 Jul 2003, Yifang Dai wrote:
[...]
I've successfully flashed 2 7940 with sip image, they can now talk to
asterisk pbx, call each other, vm etc.
Cool.
Now I'd like to get them talking to CCM via asterisk through the oh323
channel.
extensions.conf
exten = 6107,1,Wait,2
exten =
On Fri, 25 Jul 2003, Kelvin Chua wrote:
yes, i agree, we never really felt the need to use unity, *'s vm is
functionally ok with callmanager
(except for the message waiting indication, or is there?) can *'s vm send a
MWI to the callmanager?
Not yet.
However, CCM supports MWI notifications
On Thu, 24 Jul 2003, Jeremy McNamara wrote:
Siggi Langauf wrote:
Are you running CCM 3.3(2), too?
No idea, I avoid dealing with CCM at all I fought tooth and nail to
stop them from wasting money on it, but they wouldn't listen to me.
Same thing here: they're probably going to pour
On Wed, 23 Jul 2003, Yifang Dai wrote:
I wish! My company just spend a lot $$ on the shinny CCM phone system, so I
don't think I can change that easily... But if I can get asterisk to
talk to CCM via h323, and prove it's usefulness, I might have a chance
to use * in the branches...
Well,
On Wed, 23 Jul 2003, Ronnie Earle wrote:
I'm sure asterisk would make a great stand alone voice mail server.
Basically I want to get rid of our voice mail system and replace it with
*, but the problem is we use a cisco cluster with skinny clients. So I
was thinking the way to contact a *
On Wed, 23 Jul 2003, Troy Settle wrote:
Funny. I just subscribed to this list to ask the exact same question.
The application I have in mind though, would be a little more intense. What
I would like to create, is a unified messaging center for voice, fax, and
follow-me service (home,
On Thu, 24 Jul 2003, Jeremy McNamara wrote:
Siggi Langauf wrote:
We're running Cisco CallManager 3.3 with Cisco 7940 and 7960 Skinny
phones. * is registered to the CallManager as an H.323 gateway (using the
chan_oh323 driver, chan_h323 didn't work with the Cisco cluster).
chan_h323
On Thu, 17 Jul 2003, Yifang Dai wrote:
On Thu, Jul 17, 2003 at 06:50:41PM +0200, Siggi Langauf wrote:
There's not much to it: just configure * as an H.323 gateway in
CallManager for the appropriate extensions.
Thanks! I'll need to read more about how to accomplish this in CCM :)
Trivial
On Wed, 23 Jul 2003, Michael Manousos wrote:
Is it possible to use 2 B channels simultaneously
with either I4L or CAPI drivers?
We use AVM A1 (Fritz) PCMCIA with I4L driver and
AVM B1 PCMCIA with CAPI driver.
Sure.
I have an I4L setup that does this, and it's working fine, even with both
B
On Thu, 17 Jul 2003, Yifang Dai wrote:
On Wed, Jul 16, 2003 at 12:32:41PM +0200, Siggi Langauf wrote:
Has anybody tried Cisco 7960G? Or 7940?
sure, using them all the time here (the Skinny version, which requires
Cisco CallManager which in turn connects to asterisk via H.323
On Wed, 16 Jul 2003 [EMAIL PROTECTED] wrote:
Has anybody tried Cisco 7960G? Or 7940?
sure, using them all the time here (the Skinny version, which requires
Cisco CallManager which in turn connects to asterisk via H.323).
The hardware (same as for the SIP version, in fact you can convert
On Wed, 11 Jun 2003 [EMAIL PROTECTED] wrote:
our Asterisk pbx is sitting behind a normal analog hardware pbx, we have
to dial 9
to take an outside call through the hardware pbx, our fxo interface is
also connected
to one of the extensions of it. we can make calls to internal hardware
pbx
. Thanks in advance!
And while I'm at it: thanks for writing the driver in the first place!
Let me know if I can do anything to help...
Cheers,
Siggi
(unchanged fullquote follows...)
Siggi Langauf wrote:
After some more analysis of my dropped fragment problem, things look
like
After some more analysis of my dropped fragment problem, things look
like this:
Cisco 7940 phone -- RTP -- chan_oh323 -- Asterisk
(running, eg., VoiceMailMain)
That RTP connection was negotiated via H.323 on a third machine running
Cisco CallManager
Hi,
trying to build the h323 channel driver that comes with asterisk works
fine, but only as long as I use openh323-1.11.7.
Unfortunately, that setup seems to have a bug which misguides one of the
audio streams. (So while * can hear me, the phone remains silent.)
I suppose that bug is fixed at
On Tue, 10 Jun 2003, Jeremy McNamara wrote:
trying to build the h323 channel driver that comes with asterisk works
fine, but only as long as I use openh323-1.11.7.
Unfortunately, that setup seems to have a bug which misguides one of the
audio streams. (So while * can hear me, the phone
-- Forwarded message --
Date: Wed, 11 Jun 2003 01:10:16 +0200 (CEST)
From: Siggi Langauf [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
On Tue, 10 Jun 2003, Jeremy McNamara wrote:
If you would have followed the build
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