On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote:
> This is a bit of trickery, but could not resist :)
>
> This will kill a channel that is connected to SIP/201
>
> asterisk -rx "soft hangup $(asterisk -rx 'show channels' | grep SIP/201
> | awk '{ print $1 '} )"
what if there're also ch
On Wed, 14 Jan 2009 16:09:02 +0100, philipp-chemn...@gmx.de wrote:
> setting the caller ID works perfect. Detecting if a caller is or isn't
> registered is the problem. I'm using sip.
wouldnt ChanIsAvail() or regexten/regcontext settings in sip.conf assist
in this ?
--
Regards,
On Wed, 14 Jan 2009 09:24:05 +0100, philipp-chemn...@gmx.de wrote:
> Hi guys,
>
> I am trying to set the caller ID to 'Anonymous ' if the
> caller is not registered to the asterisk server. But I can't find a
> solution.
put registered users in one context which dials out, and unregistered
users
On Fri, 3 Oct 2008 12:00:16 -0800, Babcock, Michael Alex wrote:
> is it frig or fring?
>
> On Oct 3, 2008, at 11:49 AM, Tariq .. wrote:
>
> > try using Frig.. it's a great client with an SIP client.. i tried it
> > on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi-
> > Fi...
On Fri, 26 Sep 2008 15:04:30 -0400 (EDT), Ken D'Ambrosio wrote:
> So I'm confused: any ideas on how this worked when the PBX was hooked
> straight to the PSTN? Is there some SS7 signal or something that says,
> "This is an international call", when the number has no 011 preface? I'd
> hate to hav
On Thu, 25 Sep 2008 12:21:41 -0400, Jason Aarons \(US\) wrote:
> A lot of places you still can't get GSM in the US.it has
> improved...but GSM 3G coverage is lacking compared to EVDO/CDMA.
which isn't usually a problem as all 3G phones i've seen also use GSM, and
the phones switch to GSM when
On Thu, 25 Sep 2008 18:00:00 +0100, Tim Panton wrote:
> It's essentially a channel driver.
> Licensed per channel in the same way that the g729 codec is.
which would mean that us freebsd folks are going to be left out. oh well.
--
Regards, /\_/\ "All dogs go to heav
On Sun, 22 Jun 2008 08:45:10 -0500, Michael Graves wrote:
> I have a small Portech GSM gateway. It works well. It's GSM<>SIP which
> seems to me a better solution than FXO/FXS type interfaces. They make
> gateways up to 32 port for E-1 interconnect.
what did they cost, michael ?
--
Regards,
On Wed, 14 May 2008 17:06:54 -0400, Alexander Lopez wrote:
> SS7 helps carriers maximize the use of the circuits that interconnect
> them with others. Instead of using a channel and having it open for 30
> seconds as the call is setup, user gets signaling (busy, ringing, not in
> service), and cal
On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
> The best option is to put a SIP Proxy in front of your Asterisk sever
> and block REFER requests.
or just comment out the block in chan_sip.c which handles the refers.
--
Regards, /\_/\ "All dogs go to heaven."
[E
On Fri, 28 Mar 2008 06:33:42 -0700 (PDT), Vieri wrote:
> However, I can't use ringinuse=no in queues.conf
> because I'm running 1.2.27 (or is there a
> backport/patch?).
iirc, there is a patch to backport ringinuse to 1.2.x. it's on mantis
somewhere.
--
Regards, /\_/\
On Fri, 28 Mar 2008 12:09:58 -0500, Peder @ NetworkOblivion wrote:
> Any idea? If I remove call-limit on the sip.conf entries, it all goes
> back to working fine. I tried 2, 9 and 99 on the call-limit and they
> all have the same issues. I can't imagine why call-limit causes hints
> to stop
On Tue, 01 Apr 2008 13:32:28 -0400, Jared Smith wrote:
> On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
> > I call into the dialplan and try to play demo-congrats and I hear
> > nothing.
> >
> > Firewall is disabled.
> > Everything is on the 192.168.1.X network for this simple configuratio
On Fri, 21 Dec 2007 08:50:28 -0500, David Boyd wrote:
> On Fri, 2007-12-21 at 18:41 +0800, Dinesh Nair wrote:
> > the attached log with verbose=6 and debug=6 refers.
> >
> > we've got a sangoma A104 (no hwec) with PRI ports 1 & 3 loopbacked to
> > each other.
the attached log with verbose=6 and debug=6 refers.
we've got a sangoma A104 (no hwec) with PRI ports 1 & 3 loopbacked to each
other. we're trying to have txfax send out on one of those pri ports with
rxfax listening on the other side, hence having asterisk send a fax to
itself. we however have b
On Sat, 1 Dec 2007 00:42:43 -0500, John Bittner wrote:
> Anyone see an issue on asterisk 1.2 that it will not accept the invite
> from a Cisco gateway. If I turn off voice service voip signaling
are you sure you've got ulaw enabled for that peer in sip.conf ? and the
invite trace shows that the c
On Tue, 4 Dec 2007 20:45:08 -0500 (EST), Alex Balashov wrote:
> This sounds like the app_rxfax module has a dependency on some other
> module which implements T.30 handling, and that this module is either
> not loaded, or that its symbol table is not being shared in the
> monolithic core.
there's
On Tue, 27 Nov 2007 09:40:56 -0500, Matt wrote:
> This is a dual NAT situation. PIX on Asterisk side, and Netgear on
> phone side. HOWEVER.The Asterisk box has it's own IP but it is
> being tunneled through the PIX.I guess the PIX must be messing
> something up?
could you post a 'si
On Wed, 10 Oct 2007 12:54:42 -0500, Russell Bryant wrote:
> I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc.
>
> Another proposal has been using 1.5 to indicate that it is a release
> candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the
> release candidates for t
On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan & Company, LLC wrote:
> Though still in the proof-of-concept stage, my project "AstSee" from
> http://www.astsee.com/ might be fun to play with if you're using
> linux/XWindows. There are screenshots there.
that may be so, but without source,
On Wed, 27 Jun 2007 09:08:21 -0500, Matthew Fredrickson wrote:
> You fixed your clocking then. That was what I was thinking of. Make
> sure that your Dialogic card is also pulling timing from the Digium
> card as well. What version of zaptel drivers are you running?
on a related issue, usi
On Thu, 26 Apr 2007 12:39:32 -0400, Dave Miller wrote:
> Dave Miller wrote on 4/26/07 11:46 AM:
> > We upgraded our asterisk server to 1.2.18 last night to pick up the
> > security update. Since then, any calls coming in on IAX2 links get
> > dropped if they try to enter a MeetMe conference room.
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
> I
> think it can be done by using the dialplan and the database to store the
> statistical information but maybe there is an easier way that integrates
> better with asterisk!?
i dont think you'd even need a database with statistics. jus
On Tue, 17 Apr 2007 20:55:44 -0400, Jason Aarons \(US\) wrote:
> Anyone configured a E1 PRI in Kuala Lampur, Malaysia with TM Malaysia?
> What signaling did they provide, framing, formatting?
we have many times for our customers. E1 EuroISDN with CCS, HDB3, CRC4.
works great out of the box.
--
On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote:
> The phone no longer registers with asterisk, although it displays the
> little icon as though it has, and it doesn't even seem to try to pass
> calls to asterisk...
>
> So, I would avoid 3.06330904 20-11-06 RM-49
i've got an E61 runnin
On Mon, 16 Apr 2007 10:48:40 +0100, Mark Reardon wrote:
> 1) But how do I inject them into the SIP channel.
> 2) How do I time the injection so that the correct message is played at
> the correct time.
take a look at the L() option to Dial().
--
Regards, /\_/\ "All d
On 02/25/07 22:16 Doug Lytle said the following:
My experience from yesterday shows that zaptel.c has been renamed to
zaptel-base.c. This prevents the Sangoma Setup script from patching
zaptel. The fix (Found by Googling) was to rename every instance of
ok, the sangoma scripts on freebsd
On 02/25/07 22:16 Doug Lytle said the following:
zaptel-base.c. This prevents the Sangoma Setup script from patching
zaptel. The fix (Found by Googling) was to rename every instance of
ok, the sangoma scripts on freebsd do not patch the zaptel-bsd source in
any way, so this shouldn't affe
On 02/25/07 06:26 Darrick Hartman said the following:
Kristian is working with Sangoma to get wanpipe supported once again in
Asterisk.
is there a reason why wanpipe stopped working with asterisk ?
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED]
On 02/22/07 06:04 Michelle Dupuis said the following:
I'm working on calls coming in to an asterisk box as H.323, and going out as
SIP to a remote device (a VoiceMaster). The remote device is refusing the
calls with SIP error 406 (Not Acceptable).
I have attached the SIP debug output below.
On 02/01/07 02:15 Olle E Johansson said the following:
both channels should act the same unless there's a configuration that's
giving wrong information
to chan_sip, like you having a username= or defaultip= setting.
how does a username= entry in sip.conf affect dialling behaviour when the
On 01/28/07 18:52 Florian Overkamp said the following:
Nokia seems to have done something like this in their E-series (E60 etc)
with Avaya and Cisco. Anyone have a lowdown on the technical stuff there ?
i think that's a FMC (fixed mobile convergence) client which both avaya and
cisco wrote f
in a simple dialplan like follows:
[firstcontext]
include => secondcontext
include => thirdcontext
include => fourthcontext
[fourthcontext]
_03X.,1,Goto(${EXTEN:2},1)
_X.,1,DoSomething()
_X.,2,Hangup()
the Goto() for exten _03X. seems to start the search for the jump within
firstcontext, thu
On 01/05/07 06:18 Zoa said the following:
It used to be a problem to have very big iax2 trunks (e.g. > 100 channels).
anyone remember why this was so, and if a bug was opened on this for 1.2 ?
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED]
On 12/29/06 06:04 Hans-Jürgen Brand said the following:
Found problem
xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't
know how to change this at xlite
have you tried nat=yes in sip.conf for the peer ?
--
Regards, /\_/\ "All do
On 12/23/06 09:51 Leo Ann Boon said the following:
I would love to hear how others are using the results from show
translation in system dimensioning. So far, I feel that dimensioning an
Asterisk box is still mostly guesstimation :). Currently, I'm using the
30MHz per call rule to dimension.
On 11/16/06 06:06 David Gagnon said the following:
Which version are you using? There was a problem in 1.2.12.1 with the page
application. Update to 1.2.13.
what was the problem ?
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED](0 0)
On 10/11/06 21:15 Joseph said the following:
I quits on my as well, when I try to make a second call.
There is a bug report on it:
http://bugs.digium.com/view.php?id=7972
this seems like a configuration error within FreePBX and isnt really a bug
in asterisk.
--
Regards,
On 09/20/06 15:06 Dinesh Nair said the following:
On 09/19/06 16:59 Steve Langstaff said the following:
I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124&nbn=4
thanks for the link,
how
On 09/19/06 16:59 Steve Langstaff said the following:
I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124&nbn=4
thanks for the link,
however, on 18th may 2006, kpfleming's note says, "This should be
the situation
Asterisk <-- SIP ---> SIPGW <--- SIP Phone
SIP Phone is trying to call asterisk dialplan:
exten => 0224577501,1,Answer()
exten => 0224577501,2,Playback(demo-instruct)
exten => 0224577501,3,Hangup()
however, asterisk 1.2.12.1 (on FreeBSD 6.1) sends back a "488 Not
acceptable her
On 09/13/06 07:22 Ronald Wiplinger said the following:
I need h.264 and tried therefore svn checkout
http://svn.digium.com/svn/asterisk/trunk asterisk
asterisk supports h.264 in passthru mode. we've tested this with eyebeam
video SIP clients without problems.
--
Regards,
On 09/11/06 18:36 Paco Brufal said the following:
Hello,
Here in Spain there is a VoIP provider (Telefonica) that only works
if when you make an outgoing call, the SIP headers are like this:
INVITE sip:@telefonica.net SIP/2.0
But Asterisk is sending this:
INVITE sip:@sbc.ngn
On 08/26/06 23:52 Crazy Boy said the following:
Hi friends,
I did music on hold. How can we implement music on call transfer? I am
unable to find any tutorial about setting up music on call transfer,
i'm not exactly sure what you're intending to do, but MoH is already active
and played
On 08/27/06 13:23 Rushowr said the following:
Set(TIMEOUT(absolute)=seconds)
Change seconds to the number of seconds you want to allow a call to last
alternatively, look at the L() option to Dial.
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED]
On 08/24/06 09:02 El Flynn said the following:
Just wondering -- has anyone used the SIP phone feature on the Nokia
E60/61/70 phones? We're trying to see if this would be an OK phone to
get for the company, particularly since we're already running Asterisk.
SIP works well with asterisk, with
On 08/16/06 23:35 Robert Michel said the following:
I think the BCM chip is for the GSM stuff, for GUI and applications
the XScale chip - so for running asterisk, the XScale will be the
processor.
why would you want to run asterisk on the phone ? ideally, it should be
running a softphone an
On 08/17/06 14:56 Olle E Johansson said the following:
Don't do it within chan_sip, do it within the dialplan by using
playback with the no answer option before you dial out...
yes, that will force early media and cause sip_write() to force send a 183.
thanx, this should work. i'll test it o
On 08/15/06 23:30 Michael J. Tubby B.Sc (Hons) G8TIC said the following:
I suspect your problem is with the softphone implementation...
definitely, the SIP spec iianm says that UACs should play a ringing tone
when the 180 is received.
Occasionally calls which go from 100 -> 180 without goi
i'm not sure if this is a -users or a -dev question, but am sending it here
anyways. discussion could move to -dev if chan_sip.c code needs to be
amended/explained.
first up, all this on asterisk 1.2.10 on freebsd 6.1.
here's the beef:
from a particular sip softphone we're playing with, we
On 07/29/06 02:49 Miles Scruggs said the following:
http://forum.4psa.com/showthread.php?t=455
Take it for a ride around the block and tell them what you think. As
powerful as the config files, and command line interface is, there is
is there anywhere we can take a look at screenshots wit
On 07/27/06 03:28 Phil Jordan said the following:
Jul 26 20:05:22 DEBUG[16371] channel.c: Scheduling timer at 160 sample
intervals
Jul 26 20:05:22 DEBUG[16371] channel.c: Avoiding initial deadlock for
'IAX2/phil-5'
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Called IAX2/phil
Jul 26 20:05
On 07/26/06 14:58 Phil Jordan said the following:
Before I get round to posting my configs for critique, is this a BSD
port issue? I see stuff around on the net re the BSD port, to the
no, it isn't a BSD port issue. many people run asterisk from ports with
ACDs without any problems. in yo
On 07/18/06 04:03 Fredrik Emil Jensen said the following:
the packet too, but when the firewall/router loses its table (usually it
will timeout after xx sec/min) you will only be able to dial outgoing
can't you use qualify to get the nat device to keep the mapping ?
--
Regards,
I am not for the billing part, as its sip based, and its educational calls
only. I mean between sip.edu community and my educational institute. So
practically any sip uri should be able to be dialed from the website. I
dunno I am just asking the ideas for the group.
Regards,
Dinesh
Hi Marnus,
That is a good idea, I didn’t think
of thatJ
thanks
Dinesh.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marnus van Niekerk
Sent: Thursday, July 06, 2006 4:59
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject
uri and dial dip uri 2 and
bridge the call? Do I need any special sip api for this? Any ideas will be niceJ. Does this
webpage has to be on asterisk server running on the machine? Or can it be
passed as a string to the server from the webserver?
Regards,
Dinesh
Birlasekaran
Network Engineer
On 07/04/06 00:16 Jean-Denis Girard said the following:
It should be working. What happens exactly: is this an installation
problem, or what ? Can you try running Firefox from an xterm, there
should be some messages, eg.
it dies with FATAL ERROR: No connection to "network client" in a popup
On 07/03/06 15:41 Jean-Denis Girard said the following:
MozPhone no longer depends on any external libraries (libiaxclient is
statically compiled in, and jslib is now included). So install is very
simple, like any other firefox extension. It is correct that newer
cant seem to get it to work
On 07/03/06 12:51 Tzafrir Cohen said the following:
Web pages, evenwith javascript, are still very limited. For instance,
they cannot establish UDP communication on their own with other places.
An arbitrary TCP connection is also not so trivial.
presently yes, however this will soon change as
On 06/29/06 01:18 Jeremy McNamara said the following:
why not setup a listen only meetme for the 'listeners' and talk only for
the 'talker'?
isnt the Page() application used for stuff like this ?
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED]
On 06/29/06 05:17 Tzafrir Cohen said the following:
But it's not a "web phone" by any means. Writing a soft phone in HTML and
javascript is practically impossible.
with the amount of interest in AJAX, DHTML and the much hyped Web 2.0, this
may soon be a possibility as the browsers open up m
On 06/29/06 04:41 Forrest Beck said the following:
Here is a firefox plugin that connects to asterisk via IAX protocol.
http://moziax.mozdev.org/
works only on windows, right ?
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED](0 0) htt
On 06/25/06 19:01 Roy Sigurd Karlsbakk said the following:
seem to pass all SIP and RTP traffic through their own servers... See
http://karlsbakk.net/asterisk/gizmo-project.php for details
interesting. but isnt Gizmo an open source client ?
--
Regards, /\_/\ "All
On 06/20/06 18:20 Matt said the following:
It seems 1.2.9.1 does not correct this behavior... can I correct it
somehow?
matt, i believe i've already sent this to the list.
the bug at http://bugs.digium.com/view.php?id=6897 has the fix for 1.2.x as
agent-endcall.patch. apply that, and hittin
On 06/23/06 01:22 Andy Brezinsky said the following:
< Protocol Discriminator: Q.931 (8) len=47
< Call Ref: len= 2 (reference 15996/0x3E7C) (Originator)
< Message type: SETUP (5)
> Protocol Discriminator: Q.931 (8) len=10
> Call Ref: len= 2 (reference 48764/0xBE7C) (Terminator)
> Message
On 06/13/06 22:49 Colin Anderson said the following:
Although this may have changed in the newer 1.2.X series of Asterisk, I
believe that Asterisk does not support SMP from the perspective of
isnt asterisk multithreaded ? on a proper OS thread implementation, threads
can migrate across CPUs,
On 06/12/06 21:11 Matt said the following:
What version of Asterisk are you running, that you are able to dial *2
and the * isn't hanging up like it is for me?
because i wrote and applied the patch ? :)
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED]
On 06/12/06 20:42 Dinesh Nair said the following:
i would
think that 1.2.9.1 would also have this patch applied.
not it doesnt. my patch was only committed for trunk, though mantis does
have the patch that works on 1.2.x as well.
--
Regards, /\_/\ "All
On 06/12/06 20:21 Matt said the following:
AHHH! We use the Xfer button on our Aastra 9133is to do transfers
for some reason (see another post I just made) when I hit * queue
calls disconnect.
take a look at http://bugs.digium.com/view.php?id=6897 which solves this
problem. also, since
On 06/03/06 22:10 Kevin P. Fleming said the following:
- Michiel van Baak <[EMAIL PROTECTED]> wrote:
Then the svn automerge thingie Kevin wrote for the asterisk
svn tree is automerging changes to the 'common' tree to all
the server trees.
unrelated to asterisk obviously, but is there s
On 05/19/06 18:57 Chris Hastie said the following:
Yes, I have these. The modules load, but ztcfg complains
"ZT_CHANCONFIG failed on channel 1: No such device or address (6)" and as I
said, it doesn't appear that the card has been recognised by the kernel.
could you try the X100P in anther sys
On 05/19/06 16:30 Chris Hastie said the following:
I've just received an OEM Wildcard X100P FXO card. Installing into my
FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at all. Since
have you downloaded, compiled and installed the zaptel-bsd drivers ? if you
haven't, instructions
On 05/18/06 18:45 Sebastian Kayser said the following:
So although the Zap interface is used for both types of "external" calls
(snom -> POST, snom -> PSTN) the ringing indication to my snoms fails
for calls to the PSTN.
we've got the following:
E1 PRI --- Asterisk ---+--- FXS Gateway --- Ana
On 05/17/06 04:00 Noah Miller said the following:
only one registration. You can register from multiple devices, but
only the one that has most recently registered will receive calls.
Put another way, when the second device registers it will unregister
the first device.
exactly as you've put
On 05/11/06 19:46 Josué Conti said the following:
functions informs "to transfer" and the transference is ok. However, if
the agent tries to effect an attended transference the ATXFER, knocks
down the call. All the agents of this queue are with canreinvite=no in
i'm guessing that the feature
On 05/02/06 20:50 Josué Conti said the following:
To activate the transferences of calls in asterisk, I effected:
SIP.CONF in sip of the agent I qualified canreinvite=no, so that
asterisk monitors this transference.
EXTENSIONS.CONF I qualified the parameters tT in the command Dial
FEATURES.C
On 04/29/06 20:15 Josué Conti said the following:
Dinesh the agents they receive a call and this call will have to be
transferred, them uses only functions "hold" and "trnsf" in device
i'm not sure how the polycom's hold and trnsf buttons are mapped, but us
On 04/29/06 10:06 Josué Conti said the following:
is that if the agent transfers the call, for another user and this user
takes care of the call, the status of the agent in the "show agents" is
of that it the same continues speaking (talking to zap) with circuit
how are you performing the t
On 04/25/06 05:58 Sangoma Techdesk said the following:
At Sangoma we do quite a lot of back-to back T1 and E1
connections. T1 is not a very fussy connection, as the baud
rate is only about 750 kbps.
In our experience, for error free communications you can use
the following rules of thumb:
On 04/21/06 05:35 Sean Kennedy said the following:
I have a vague memory of reading about this somewhere, but searched @
the wiki AND through google aren't turning up anything useful.
take a look at http://bugs.digium.com/view.php?id=6897
there's a patch there for 1.2 with another for trunk w
On 04/16/06 10:51 C F said the following:
use feauters.conf and the application map section.
i may be wrong, but that's not the same as background music during a call.
iianm, using playback() or background() in features.conf turns off the call
audio and plays the selected file.
--
Regards
On 04/14/06 20:05 Matt said the following:
When it patched the
zaptel source... if I have usecallerid=yes on then it crashes... if I
turn usecallerid=no then it is fine.
we've tested the sangoma A101, A102 and A104 cards with usercallerid=yes,
and it hasn't crashed. this is on FreeBSD though
On 04/08/06 11:26 [EMAIL PROTECTED] said the following:
I'm pleased to announce the release of Astmanproxy 1.20, the fast,
flexible proxy server for Asterisk's Manager Interface. Astmanproxy
we've just started using astmanproxy, and i'll soon be submitting a couple
of patches which address
On 04/08/06 11:26 [EMAIL PROTECTED] said the following:
I'm pleased to announce the release of Astmanproxy 1.20, the fast,
flexible proxy server for Asterisk's Manager Interface. Astmanproxy
we've just started using astmanproxy, and i'll soon be submitting a couple
of patches which address
On 04/05/06 13:17 Avi Miller said the following:
I had a similar problem connecting Asterisk to an Avaya IP403 via
OOH323: In the end, I removed all the disallow=all and allow=
lines in Asterisk. This seems to have allowed the two systems to
overcome the codec negotiation problems they were h
has anyone managed to get these three beasties to work together ? we're
using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft
netmeeting default from windows xp.
the symptoms are that calls from a SIP client to NetMeeting rings on
NetMeeting, but upon answering the call in NetMee
On 04/05/06 13:52 Dinesh Nair said the following:
On 04/05/06 13:17 Avi Miller said the following:
I had a similar problem connecting Asterisk to an Avaya IP403 via
OOH323: In the end, I removed all the disallow=all and allow=
lines in Asterisk. This seems to have allowed the two systems
On 04/06/06 04:41 Dan Austin said the following:
Chan_ooh323 just worked. The code is, to a infrequent programmer,
easy to read, extend and fix bugs.
ok, i'm not getting into a my H323 is better than yours argument, but we've
been struggling to get OOH323 working with OHPHONE. symptoms are t
On 04/06/06 19:17 Tomislav Parèina said the following:
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
on a related note, we notice that if we've set atxfer = *1 in features.conf
and blindxfer=#1, then attended transfers dont work. somehow, the Queue app
captures the '*' and hangs
On 03/31/06 08:24 Wai Wu said the following:
In Asterisk, what happens to the files when both legs of the call hangs
up? Is there a way to create a conference room on the flight? i.e.
without pre-defining the conference ID in meetme.conf.
look at the 'd' option to MeetMe.
--
Regards,
On 04/06/06 05:36 Avi Miller said the following:
If I dialled from a SIP phone on Asterisk 1 to the Phone on the Avaya,
it worked fine. If I dialled from a phone on the Avaya, the SIP phone
would ring, but the call would drop as soon as it was answered because
of codec negotiation failure.
a
On 04/05/06 21:37 Dov Bigio said the following:
- The agent transferred the call to an user (not a queue), by dialing
the atxtransfer (1) key defined in features.conf
on a related note, we notice that if we've set atxfer = *1 in features.conf
and blindxfer=#1, then attended transfers dont wo
On 04/04/06 19:20 Tomislav Parèina said the following:
Ooh323 channel driver from asterisk-addons-1.2.1 has same problem
have you managed to get this working ?
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque
On 03/31/06 23:29 Jim Houser said the following:
Looking at the TE100P I don't see it listed Q.SIG as supported. We have it
working great as PRI. Am I wrong about the Q.SIG support?
Q.SIG and the like are supported from libpri. we got it working with a
TE410P, but i'm sure getting to work
On 03/31/06 19:49 Wolfgang Zweimueller said the following:
My conclusion with Q.SIG: do not use it at this implementation
level. YMMV.
i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110
for a customer in thailand.
--
Regards, /\_/\ "All dog
On 03/31/06 00:28 Stefan-Michael. Guenther (in-put GbR) said the following:
To make it clear: We don't want to compare the three system against each
other. The asterisk server is running on a completely different hardware. We
what are the hardware and OS specs for the asterisk server ? this w
On 03/29/06 13:06 Andres said the following:
It works perfectly with other values we have tested of 40 and 60. We
currently use 60 on all our servers. It cuts down on bandwidth for a
G279 call to about 15Kbps.
with 60ms packets, is a packet loss or two noticable ?
--
Regards,
On 03/24/06 07:39 Larry Alkoff said the following:
That's how I _thought_ it worked but extens in such a created
[context_name] are not seen or used by Asterisk to dial out.
There is something missing.
have you included the new context in the context where your phones are set to ?
include
On 03/23/06 16:45 Mimmus said the following:
This is my setup:
PSTN PRI E1 --- Asterisk --- Crossed E1 cable --- Alcatel PBX
Asterisk v.1.2.1 with a Sangoma A102 card (wanpipe driver v. beta1-2.3.4)
'ztcfg -vv' gives no error.
that's probably an issue with the sangoma wanpipe drivers than it
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