Displaying 20 results from an estimated 29 matches for "jerjer".
2006 Feb 07
3
alternative to realtime?
hi
I recently spoke to mr McNamara on IRC, and he mentioned there was a
"far better way to do realtime-stuff than the usual realtime in
asterisk, and that this was GPL". He failed, however, to ever mention
how this could be done, so I just wonder if someone else might know... ?
roy
--
Roy Sigurd Karlsbakk
roy@karlsbakk.net
---
In space, loud sounds, like explosions, are even
2003 May 21
0
to jerjer or not to, i.e. not the question was ( chan_oh323.so: Segmentation Fault)
a) jerjers been doing a lot commendable work for *
b) support is not mandatory, and i agree with royk it should not be
withheld based on political viewpoints, that's pointlessly draconian
c) choice is always good, so people should have the option of oh323 or
h323, let them decide, and not limit them, u...
2007 Jun 09
0
Asterisk Users Conference Friday: New Asterisk Book and a visit from JerJer of Nufone
Oops, I had some problems and was offline unable to remind you about
the conference yesterday.
LISTEN to recent recordings: http://x2z.eu/astusers.htm (Flash
player, will autostart)
THIS WEEK: Stephan Winterberg and Stephen Boche tell us more about the
new book, whick looks like a great effort.
A surprise visit from Jeremy, one of the pioneers of our community who
started Nufone when someone
2004 Jan 12
4
Asterisk 0.7.0
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
Mark
p.s. there was no 0.6.0 release.
2003 Nov 24
3
Cisco to asterisk termination with h323 and g729 finally works.
Hello,
I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323.
I tested both oh323 from inaccessnetwork and JerJers chan_h323.
I used 1.12.2 version of oh323 and 1.5.2 version of pwlib.
After latest changes from JerJer chan_h323.c works ok when receiving traffic
from ciscos. I havnt found any audio problems although I didnt send much
traffic.
Latest oh323 has some problems with proper g729 codec recognition...
2007 Jul 04
1
Asterisk TV will go live this Friday
...be
back with us on the conference, now that I've had a chance to play
with their speech recognition product.
I think we're starting to get some great info from the user community
and I would like to see more service providers show themselves. I
think it's great to hear what people like JerJer of Nufone to say.
We've had some bleeding edge stuff on past conferences as well, with
Jay from Adhearsion and I'm waiting for some video content from him as
well.
Digium has been great about following the evolution of these community
projects and providing time for people like Russell to...
2003 Oct 16
0
Use of the "hint" modifiers - examples, anyone?
...ome real-life examples of how
to use this perhaps very useful tool. I understand the point of the
tool, but I need to get some actual configs to look at before I think
I'll figure it out. Even if my particular equipment doesn't support
it, there may be other ideas I can get from it. (JerJer - maybe SCCP
could use that data if there is an SCCP command of similar nature to
the SIP SUBSCRIBE command - that would be pretty handy for those 7914
operator stations.)
Searching through the source gives tantalizing hints (no pun
intended) in pbx.c, but no actual real-life samples. Can som...
2003 Oct 26
1
NuFone International Calls
Does anybody know how to do an international call using NuFone. I realise
this isn't really the place to ask, but NuFone appears to be closed for
the weekend and would like to have a try at this before tomorrow. I
assumed it would be '011' for an international line followed by country
code but that doesn't seem to work.
I am getting:
-- Executing
2005 Mar 04
1
chan_h323 & codecs
Hi,
Can anyone confirm that if I want to do h323 proxying that I do not need
codecs installed? For example if the codec being used is g723.1, I don't
need the codec installed locally because there is no compression or
decompression being done on my server; the incoming traffic is simply
being sent out on another h323 channel (h323 in->h323 out). Is this correct?
Thanks,
Chetan
2005 Jun 27
1
Re: teliax [Was: LiveVoip is Bankrupt]
...:29 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]
On 2005-06-27, Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote:
> http://www.nufone.net. I've been using them for the past 18 months
with zero
> technical hassle. Jerjer and Shido6 hang out on IRC. Nufone is not a
"hand
> holding" VOIP provider. You are expected to have some clue. This has
turned
> away a number of people but as I said, they Just Work.
So I've heard three recommendations for people coming from LiveVOIP:
nufone, teliax, and...
2004 Nov 23
1
CLI > h.323 show codecs shows nothing
Hello
I like to make calls to an h.323 device.
I'm using Nuphone h323.
Compiled everything okay "I Guess"
When I make a connection * SIP > h323 device, the phone is ringing and then * tells me "No one available....." and disconnect
Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing.
I try many combination in the h323.conf like.
2003 Nov 25
1
SIMPLE support in Asterisk?
Hi
Is there any work being done on implementing IM/SIMPLE support
for SIP on Asterisk? Like a presence server?
rdgs,
/Staffan Kerker
2004 Dec 16
1
Asterisk Cisco CallManager Integration
Hi,
Where can I find information on H.323 for Asterisk and/or integration with
Cisco CallManager in particular?
<http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration>
I have oh323 working on Asterisk. Since the CallManger I am working with
is running 3.3.3 I cannot use SIP...
Thanks,
Adi
2004 Jun 22
3
IAX2 Trunking help!
I'm trying to get two * boxes to talk.... no matter what variation I try
I get No Authority Found and connection refused from 192.168.1.5
I've googled, I've site searched.... to no avail.
Here is the server a configs (192.168.1.5):
iax.conf
[general]
port=5036
bandwidth=low
disallow=all
allow=gsm
jitterbuffer=yes
tos=lowdelay
register => pbx:test@192.168.2.2
[pbx]
type=peer
2004 May 16
2
Re: say.c compilation error
...this function)
make: *** [say.o] Error 1
I was using PWLIB-1.5.2 and OPENH323-1.12.2 previously and there
wasn't any of the above errors before when I compiled it. I am changing
it to the above versions i.e 1.6.6-1 and 1.13.5-1 respectively, because
I want to compile the channel_h323 code from JerJer who stated that it
must be a Janus-2 version.
I would appreciate any help rendered to me. Thank you.
Rgds
T.E.Yap
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2004 May 17
0
CAPI<->SIP broken incoming audio
...gt; don't have any other issues with that (not even SIP-IAX, where the
7960 is
>> > really bad).
>>
>> I've done some crawling over ethereal traces, and have found the
problem
>> to indeed be bad timestamps in the RTP payload from *. I was advised
(by
>> JerJer on the IRC channel, thanks!) to update to latest CVS-HEAD, but
the
>> problem has recurred. What's happening is that the RTP frames from *
are
>> all going out with the same timestamp, which is causing my
>> timestamp-sensitive 7960 to barf and ignore the incoming audio st...
2004 Sep 23
2
CallerID on Channelized T1 not working with 1.0.0
I have been using Asterisk 1.0-RC2 successfully with a channelized T1
circuit for quite a while now but after upgrading to 1.0.0 callerid no
longer works properly.
Debug output from a channel shows what actually is received through DTMF
from the carrier:
<< [ TYPE: DTMF (1) SUBCLASS: * (42) ] [Zap/9-1]
<< [ TYPE: DTMF (1) SUBCLASS: 2 (50) ] [Zap/9-1]
<< [ TYPE: DTMF (1)
2003 Aug 15
1
Asterisk H323 Trunk
During debugging of H323 trunk side (using Jeremy Macnamara's H323
driver in ~/channels/h323) a couple of things come don't quite work
as advertised...
1/ the following line in extensions.conf explicitly sets the
outgoing caller ID (required in my case for downstream GK
processing..)
exten => _1NX.,1,SetCallerID,6400047602100
exten => _1NX.,2,Dial,H323/${EXTEN:1}
what
2003 Sep 22
2
G.729A + Cisco AS5300
Hello,
I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected.
The codec list show on my cisco AS5300 for g.729 are:
g729r8
g729br8
I suspect that
2004 May 14
1
chan_capi broken incoming audio
G'day all,
I've been googling myself silly looking for help on this one but have come
up blank.
I have an AVM Fritz!Card PCI, and I'm using chan_capi v 0.3.1 with * from
CVS-HEAD-05/08/04-22:48:00. I can start * and make and receive calls on
ISDN fine but after a few hours of * uptime, on any ISDN call I make or
receive from my SIP handsets (7960 or ATA-186) I get bad audio: on the