Richard Scobie wrote:
Is there a way to configure the IAX jitterbuffer to get the benefit of
trunktimestamps, while not having any jitterbuffering (reducing delay)?
My SVN asterisk systems use the following topologies:
1) PolycomSIP -> *1 ->IAX-> *2 -> H323 Gateway
2) PolycomSIP -> *1 ->IAX-
Rich Adamson wrote:
I've noticed that if I enable the jitterbuffer in iax.conf with Asterisk
1.2.0 that Asterisk stops responding to incoming DTMF frames for calls
between Teliax and my server. I've used "iax2 debug" and Ethereal to
confirm that Teliax is, in fact, sending the frames.
I onl
Kanuri, Seshu (Company IT) wrote:
Iain Barker Wrote:
-
Our experience with over 10 or more participants
in a single Asterisk conference was that quality
degraded quite rapidly.
Is this really true as there were many in this list
who had confirmed that
Julian Lyndon-Smith wrote:
We had to move from a old * server to a new one in a hurry (hardware
failure). The old server was a dual pentium 700 with 512MB ram running
fedora core 2, the new one is a single 3GHz Pentium with 1gb ram.
The same number of people are connected to the new server as
Yair Hakak wrote:
Hello all,
forgive me if this is a simple question, but does bridging a SIP
channel and an IAX channel that use the same codec (say, alaw) involve
transcoding? i'm trying to figure out what kind of hardware i'll need,
and i'm going to be using SIP endpoints and IAX trunking
Rich Adamson wrote:
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes have
Johan Nordström wrote:
Try send an email to [EMAIL PROTECTED] I did regarding an other
issue and recieved an answer from [EMAIL PROTECTED] .
Johan
There are two major products that come out of Berkeley: LSD and UNIX.
We don't believe this to be a coincidence. -- Jeremy S. Anderson
Paul Bel
Adam Robins wrote:
So, then these would be the same ports defined in RTP.conf?
If you're using RTP. You haven't told us what you're doing, other than
using speex. We can only guess that you're using asterisk, and that
you're using either IAX2 or one of the RTP-based VoIP protocols..
Alessandra Grasso wrote:
Hi
Which instruments of monitoring I can use in order to test application
IAX?
wow, that's a vague question!
Ethereal can decode most IAX frames.
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Kris Boutilier wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Clive
Sent: Thursday, July 07, 2005 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX2 Trunking - CVS-Head
Is anyone successfu
Lee Azzarello wrote:
I have Asterisk running in Xen virtual machines. Unfortunately, this
kind of virtualization makes a real time clock impossible, which in turn
makes ztdummy or a Zaptel driver impossible to load, which also makes
MeetMe conferences impossible.
As an alternative, I have downl
Raul Elizondo (wizardteam) wrote:
Two asterisk, one in a 2.4.x and another one in a 2.6.x are connection ok
using IAX, before i upgraded one of them (the one with 2.6.x) Trunk was
working ok. Since i upgraded the one that its now 2.6.x, i m getting this
message:
chan_iax2.c:5067 socket_read: m
Roy Sigurd Karlsbakk wrote:
can someone please tell me how much memory asterisk requires? I'm
running 1.0.7 and after just a couple of days uptime, this is the
process as reported from ps axfv
13766 ? S< 0:15 0 605 1365974 1265468 61.0 \_ asterisk -vvvg -c
It would appear to require about 1.4GB
Wiley Siler wrote:
On a similar note, I have a server with 1GB of memory that seems to
never release the memory back to system use.
The system is AAH 0.9. Dual AMD Athlon.
This system does IAX out ot my voip providers and has 2 TDM400 cards
in it for connection to my POTS lines.
I never have m
Roy Sigurd Karlsbakk wrote:
hi
can someone please tell me how much memory asterisk requires? I'm
running 1.0.7 and after just a couple of days uptime, this is the
process as reported from ps axfv
13766 ? S< 0:15 0 605 1365974 1265468 61.0 \_ asterisk -vvvg -c
It would appear to require about 1.4
On May 7, 2005, at 11:44 PM, Bryce W Nesbitt wrote:
Are these things possible?
1) Set the local Asterisk jitterbuffer size, but only for a particular
connection. I'd like to force Asterisk to use a particularly large
buffer in certain cases. Should I expect this to work?
[general]
jitterbuffer
Stefan de Konink wrote:
On Wed, 27 Apr 2005, Joseph wrote:
How can proprietary protocol be open protocol?
If the protocol is fully documentated and this documententation is
available to anyone you can speak of a open protocol. It is not an open
'standard', because it is only supported by Di
youssef ouadou wrote:
Aucune connexion n’a pu être établie car l’ordinateur cible (server)
l’a expressément refusée at System.Net.Socket.Connect(EndPoint
remoteEP) at IPS.listener.Start()
Hmm, I think you have two incorrect languages here; first, asterisk is
written in C, and your errors seem t
Chris Mason (Lists) wrote:
I'm sure this has been debated before, I'd like to get peoples input. I see
the hard drive as the single most likely point of failure on an * PBX. How
reasonable would it be to run the OS and config files from a CF card, mount
the /var/partition on a hard drive for the CD
D/DTX implementation through zaptel
cards
Steve Kann wrote:
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using zaptel with asterisk towards
PSTN.
TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not
even a valid idea.
Doing VAD on audio coming _from_ the TDM
] On Behalf Of Steve
Underwood
Sent: Tuesday, April 12, 2005 9:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel
cards
Steve Kann wrote:
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using zaptel with asterisk towards PSTN.
TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not
even a valid idea.
Doing VAD on audio coming _from_ the TDM world certainly is something
you might want to do
Steve Underwood wrote:
Gustavo García wrote:
Hi everybody,
GIPS have different products, not only codecs:
* Voice enhancements: packet loss concealment algorithms, noise
concealment,
jitter buffer, agc, aec (can be used with any codec)
* Codecs: iLbc (free), ISAC, G711 Wideband...
You can in
C F wrote:
It realy depends what you are trying to accomplish, if all you want to
do is add more extensions that happen to be offnet using VoIP, then
you could just add analog extensions, and use FXO in * and then IP
phones in the remote offices.
On Tue, 29 Mar 2005 13:31:16 -0800, Sean Ke
Eric,
If you want to optimize speex, I'd suggest the following:
1) Re-compile the speex library with SSE optimizations; add --enable-sse
to the configure line used for compilation.
2) Reduce the "complexity" from 4, to 2 or 3 in codecs.conf. You won't
notice the difference in quality.
3) L
Roman Zhovtulya wrote:
As far as I know speex is an adaptive codec, i.e. it will
automatically adjust to the conditions and provide the best quality
possible.
Therefore, there should be no need to configure that manually.
Could anyone correct me if I'm wrong?
It can do this (this is Variable Bi
Aldo Bergamini wrote:
[EMAIL PROTECTED] is believed to have said:
Dear all,
I have tried to run an asterisk instance together with XLite on a single
machine (a PowerBook).
The intent is to take advantage of IAX connections to easily cross NATs
while traveling.
While the IAX setup proved 'easy'
Gareth Blades wrote:
g729 is a commercial codec and requires a license to use. You can
purchase a license for use with Asterisk. See
http://www.voip-info.org/wiki-Asterisk+G.729+licensing
g723 is also a commercial codec but Asterisk does not support it other
than in passthru mode.
I have not heard
Peter Svensson wrote:
On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote:
I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an
Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover
cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of
th
Geoff Nordli wrote:
Hi Everyone.
On the Linux 2.6 kernel do I need to recompile the kernel in order to
compile the zaptel modules?
No. You shouldn't need to do this for 2.4 either.
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Harold Fletcher wrote:
Steve:
But how will that work for incoming calls? Assume that three phones
have registered with an Asterisk box from inside a NAT, then * knows
that these three users can be found at x.x.x.x port 4569.
Nope.
Asterisk would then know that these three users have registered fro
Will Fletcher wrote:
Hi folks,
Ok, I've seen this question go unanswered on the mailing list, and I
assume it's because no one had the heart to break the bad news to the
guy asking, but be honest with me, I can take it. At this time it's
flat impossible to have multiple IAX phones behind a NAT
Andrew Kohlsmith wrote:
On March 7, 2005 01:23 pm, Dennis Webb wrote:
Using TDM400's here and I have tried everything to cure the echo. I
have used the Milliwatt test from the telco and from asterisk to tune
RX/TX gain via a patched ztmonitor. What happens is I experience
midcall echo. I turn
Florian Overkamp wrote:
Hi Steve,
-Original Message-
I am having a problem with periodic breaks in audio over an
IAX trunk.
The interruption only happens in one direction, and (I think) only
with clients built on the open source libiax.
Codec is irrelevant, and jit
Rod Bacon wrote:
I have looked through the archives, and can only find old references
to this problem that appear to be no longer relevant, so I thought I'd
ask again.
I am having a problem with periodic breaks in audio over an IAX trunk.
The interruption only happens in one direction, and (I t
That's exactly the point.
I think it's 90% likely that Andres' project uses iaxclient, and 98%
likely that it uses libiax2. If this is the case, he needs to comply
with the licensing terms of these libraries. Specifically, he would need
to make the source code to his version of these libraries p
Andres,
Is your client based on libiax2? It is based on iaxclient?
-SteveK
[EMAIL PROTECTED] wrote:
Hi
We just developed an IAX web client. We are currently testing it,
and we hope to be ready to market around 15 of this month. It is based
on our own OCX, that has a lot of funtions to do/pr
Andrew Thompson wrote:
Mark Kidd wrote:
if i pick my line up and the system plays back my voice saying hi how
may i
help you automaticaly.
You're looking for something along the lines of an immediate or
hotline mode.
Basically, whether or not you can do that depends on the equipment you
are us
On Feb 12, 2005, at 9:10 PM, Michael Giagnocavo wrote:
Hello,
I understand that only the destination of a call should do jitter
buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no
transfers), PhoneA and PhoneB need to perform their own jitter
buffering,
and Asterisk will just fo
To be totally honest:
I wrote the thing.
I don't think it's ready to go into HEAD, until the core people can at
least agree on the overall structure of the implementation and integration..
There's at least one major fork that one could take with it's
architecture (basically, whether it should be
Michael Graves wrote:
Hi All,
I was just reading through Info Week while on a flight and happened
upon an brief piece about a new VOIP security intiative worked up by a
handful of the usual suspects; Alcatel, SMU, NIST, Symantec, etc. All
of this begs the question of can't we get just do this as a
Peter Svensson wrote:
On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:
how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on
the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer fo
dorian logan wrote:
Hi,
Regarding these high bandwidth CODECs - is it possible to upgrade
asterisk to record at a higher quality bit rate too
Yes, with development.
- is Asterisk based on a 8Khz system.
Yes, presently.
We would like to stream calls from SIP phones to the internet at a
higher qu
Doug Meredith wrote:
Mark Eissler <[EMAIL PROTECTED]> wrote:
AFAIK, trunk=yes is not a global option. You set it within a context.
Also, using the jitter buffer with trunk=yes is not recommended since
its broken right now.
The jitter buffer itself is broken, or only in combination with
tr
david wrote:
Hi,
Where could I download the soxmix please? I want to mix two .gsm
files into one.
Regards.
Try typing "download soxmix" into google.
The answer is result #3.
-SteveK
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On Jan 28, 2005, at 1:25 PM, Chamberland-Larose, Guillaume wrote:
Hi,
I was messing around with FireFly last night and got asterisk to crash
hard. It looks like the bug is a division by zero in chan_iax2.c.
I reproduced it and here are some infos I got from gdb:
[Switching to Thread 245775 (LWP 23
Robert Rozman wrote:
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 25, 2005 2:44 PM
Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
(IAXPhone):
Robert Rozman wrote:
- Original Message -
From: "Steve Kann" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 25, 2005 3:56 PM
Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Pho
Alberto Martínez wrote:
Hello.
I am looking for a iax java client which could be used with our
interface written in java to make iax connections with asterisk.
Does anyone know something we could use?
get iaxclient from iaxclient.sf.net CVS..
Check out tkphone and the "iaxcli" program.
You can m
Brent Goran wrote:
We have many IAXy devices in the field now.
In all cases, in iax.conf, we have "qualify=yes", so that using "iax2
show peers", we can see whether or not the device is currently online.
In some cases, the IAXy device and/or Asterisk are not communicating
their qualification, be
Daniel Joos wrote:
Whenever I take an inbound call I am getting the following errors:
NOTICE[4719]: channel.c:1698 ast_set_write_format: Unable to find a
path from speex to gsm
NOTICE[4719]: channel.c:1731 ast_set_read_format: Unable to find a
path from gsm to speex
What typically generates this
On Jan 23, 2005, at 9:11 AM, Bruno Hertz wrote:
On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote:
seem like some kind of port issue...
Actually, the fatal issue is that asterisk's chan_oss or chan_alsa
grabs the sound device, so iaxclient can't do so.
Probably. Both try to set up listeners o
Robert Rozman wrote:
Hi,
if I change Iaxphone settings to qualify=yes it says it's unreachable.
Can iax2 clients be monitored with qualify option ? Is this problem related
to iaxphone ?
Anyone sucessfully using iax qualify feature ?
It was just fixed in iaxclient-cvs this week, I think..
Don't k
John Hammen wrote:
Hi All,
I was wondering if there is any way to encrypt IAX traffic? I am aware
of the ability to use md5 or RSA for authentication, but I'm talking
about the packets themselves, after authorization has already
occured...
Forgive me if this is documented somewhere, but I all I co
Andrew Kohlsmith wrote:
On January 19, 2005 12:23 pm, Paul Fielding wrote:
I think you might want to clarify that Best audio quality is in relation to
other highly compressed codecs. Certainly my (albeit limited) experience
is that g711 is much more clear than g729. Compared aga
Roy Sigurd Karlsbakk wrote:
Hi
Is it possible to somehow monitor/log packet loss and/or jitter in
RTP? I want to know how things look if someone complains about audio.
ethereal can do some of this for rtp, I think. At the very least, if the
endpoint supports RTCP (most do, except for asterisk), i
Roy Sigurd Karlsbakk wrote:
Good day all
We have 2 asterisk servers connected to each other via IAX2 using ilbc.
Each call we make goes up to 25kbit and each one there after 25kbit as
well
bit rate is 1bps, giving 1667 bytes/sec
packetization is 20ms, giving 34 bytes per packet
Actually, iLBC
On Jan 19, 2005, at 2:49 PM, Stewart Nelson wrote:
The MOS (Mean Opinion Score) scale is:
5=Excellent; 4=Good; 3=Fair; 2=Poor; 1=Bad.
Some values, taken from "Carrier Grade Voice over IP" by
Daniel Collins:
G.711 4.3
G.729 4.0
G.729AB3.9
GSM(full rate) 3.7
The above scores
Paul Fielding wrote:
So far in my playing with Asterisk
I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream
102), and ATA adapters (Grandstream 286, Digium IAXy).
I've also got a Vonage line, using a
Linksys ATA.
None of the devices I've connected
to
Denis Galvão - iSolve wrote:
Em Seg 17 Jan 2005 19:13, Steve Kann escreveu:
I've already replied, asking for a trace.. If you get the trace, and
send it, we can look at what is actually happening:
What would really help, though, is a packet trace of the call. The
best w
Bruno Hertz wrote:
Hi folks
last weekend, I tried Windows Messenger first time and was stunned by
the little latency it gives. Until now, I've been using softphones on
Linux exclusively, like iaxcomm, linphone and sjphone, and they all give
me about 1, at times even 2 secs delay. Whereas Messenger
Denis Galvão - iSolve wrote:
Em Seg 17 Jan 2005 16:47, Dan escreveu:
Hi Denis,
Same problem with version 0.9.8c
After one minute aprox, delay disappear.
Any ideas!?
What's very strange is that I cannot reproduce this behaviour, trying
with different
Denis Galvão - iSolve wrote:
Two more information:
1. I've played with all suported codecs, same problems for all of them.
2. After aprox. 1 minute of conversation the delay problem doesn't occur, or
better, it is very less(some miliseconds) than the begining(10 seconds) of
a call.
Any ideas!?
On Jan 16, 2005, at 2:53 PM, Dan wrote:
Hi Steve,
- Original Message - From: "Steve Kann" <[EMAIL PROTECTED]>
On Jan 14, 2005, at 2:03 PM, Dan wrote:
Hi,
\> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> I dont have problems when calling PSTN extensions, and calling
>
On Jan 14, 2005, at 2:03 PM, Dan wrote:
Hi,
\> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> I dont have problems when calling PSTN extensions, and calling
> VoceMail, EchoTest, etc. The problem is related with the
conversation
> between two DIAX Softphones.
Between 2 DIAX phone and the delay is in o
Gilad Ben-Yossef wrote:
Justin Carlson wrote:
what is wrong with running asterisk with the -pg flags at startup?
Which is exactly what I suggested:
Since VoIP is a real time activity, simple "nice" really isn't
enough. What you should do is mark the Asterisk proccess as a real
time task for the
[EMAIL PROTECTED] wrote:
I built the CVS version of the Speex library - v1.2 it calls itself.
Asterisk seg faults trying to use codec_speex.so.
I'll have a look to try to fix it, but thought I'd just ask if anyone else
knows what needs to be done?
Hmm, should be OK if you're using the latest s
Waldek wrote:
Steve Kann wrote:
For CBR, there is a 1:1 correlation between "quality" and bitrate.
The speex documentation has a table which describes this:
here:
http://speex.org/manual2/node9.html
..
Of course, this table isn't helpful unless you also know this:mapping
b
Matt Schulte wrote:
I'm assuming asterisk does not have a SIP jitter buffer in place? Any
ideas on how to help with this going over a data T1 where VoIP is shared
with regular traffic? Problem is when people are downloading the voice
is jittery, even lossy.
Where do the calls go?
If it goes <*>
Esben Stien wrote:
I'm trying to figure out why I get delay using the echo test
application. I'm using the asterisk cli so I got no external factors
that could interfere. I'm getting close to half a second delay
speaking into the microphone and hearing it out through my speakers.
I'm doing lots of
Waldek wrote:
Hi,
I am looking how to setup speex codec in codecs.conf for 8 Kbps and 6
Kbps.
In config file are many parameters for setting.
I don't know what is need to change for narrowbad like 8 Kbps and 6
kbps.
Any suggestion?
[speex]
;0-10
quality => 4
;0-10
Kristian Kielhofner wrote:
Jean-Michel Hiver wrote:
I was wondering what could be pros and cons of ztdummy vs proper
timer device (i.e. X100P).
I am going to set up an asterisk server in europe (to do trunking, to
save bandwith) and I was wondering if it'll be OK to get it going
with ztdummy.
Steve Murphy wrote:
--On Tuesday, December 21, 2004 9:37 AM -0700 Steve Murphy
<[EMAIL PROTECTED]> wrote:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and
IAXPHONE on
a windows (XP) machine on my network, and I'm ru
On Dec 20, 2004, at 7:13 PM, Shoval Tomer wrote:
Thanks Steve,
See my answers inline
Calls from GS to IAXphone ring, and once I answer the call in
IAXphone,
I hear a very load noise.
On which side do you hear this noise? IAXphone, or GS?
I hear the noise on IAX site
Does the noise continue, or ju
Louis-David Mitterrand wrote:
Hi,
At the * console I periodically get these messages:
Dec 9 10:58:11 WARNING[-1248765008]: chan_iax2.c:5021 socket_read: midget
packet received (1 of 4 min)
Which seem pretty inocuous.
Google say (almost) nothing about that subjet.
What does it mean?
Probably
Shoval Tomer wrote:
We've bought the G729 codec for lowering SIP bandwidth usage (we're
using grandstream phones) and we're quite happy with it up until I tried
using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations.
Weirdly enough, calls from IAXphone to the GS phone work just fine.
S
Ernest Raspberry wrote:
Hello,
I'm experiencing some problems with running IAX2 protocol on quite
reliable link with G729A codec. My customer has 2mb FR link to the
Internet used in about 20%. Ping statistics:
50 packets transmitted, 49 received, 2% packet loss, time 49496ms
rtt min/avg/max/mdev =
Jason Becker wrote:
Chris TenHarmsel wrote:
Hi all,
I've done some minimal searching on this topic, but haven't come up
with anything conclusive. Right now we're using WAV to store
voicemail messages, that then (for the most part) get sent to users in
email when they have new voicemail. The reaso
Now that the cover is blown, what's the ETA to stop getting these.
I've gotten 270 already:
[EMAIL PROTECTED]:/home/stevek $ tail -1 .procmail.log |grep Spencer|wc
27013798886
eco.webway.se (not the only sender) is sending me one message every 56
seconds for about 4 hours now.
Now
WipeOut wrote:
Hi,
Didn't get any opinions on the log file I mailed onto the list over
the weekend so I am continuing to try and track the cause for the
dropped calls..
I have a feeling that its to do with IAX being way too sensitive when
it comes to packet loss.. Since it is going across the i
On Nov 18, 2004, at 8:23 PM, Adam Fineberg wrote:
Having some trouble with segfaults and sound quality all of a sudden (since
I recompiled from the latest source) when 2 iaxComm clients connect. First
off immediately after the server reports:
<>
<> -- Attempting native bridge of IAX2/[EMAIL PR
Sean Kennedy wrote:
[EMAIL PROTECTED] wrote:
Gentlemen and ladies of the Asterisk community.
I am considering implementing asterisk based IAX solution for a business
that handles a lot of sensitive data. Internal security will be no
worse than before as they plan on connecting to their current PBX
You might have better luck with app_conference (see the wiki) under
UML.. It is probably a little more tolerant of loose timing and scheduling.
-SteveK
nils toedtmann wrote:
Hi *,
[this goes to [EMAIL PROTECTED] and
[EMAIL PROTECTED]
i try to setup an VoIP conferencing server within a UML u
codecs, including at least G711, PCM, Speex and iLBC. The first two
with a simple concealment algorithm (I think one is described in an
RFC), and the latter via their built-in interpolation algorithms.
-SteveK
Message: 1
Date: Mon, 01 Nov 2004 10:30:41 -0500
From: Steve Kann <[EMAIL PROTEC
Whisker, Peter wrote:
[sorry about previous mis-post]
I have an * switch at home and one in the office. Both similar new CVS head
versions and both with chan_sip2 built in:
Asterisk CVS-HEAD-10/12/04-17:43:26
Asterisk CVS-HEAD-10/13/04-12:53:52
One is on a T1 connection and the other is on 576k/288
[EMAIL PROTECTED] wrote:
Hello
IAX really isn't the 'one and only' perfect signaling protocol because
many people forget one thing
IAX has one technical issue (by design) which makes it difficult to ever
get accepted by the big boys, a real big problem for carriers who have
big loads on their syste
Public Dump wrote:
Is
asterisk capable of sealing (some amount) of losses that occur on IP
based channels before it routes the Calls to a TDM channel (BRI, E1,
etc.) to limit quality loss if IP loss occurs ?
No, not yet.
See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20new
Andrew Edmond wrote:
Message
All
--
System
FreeBSD 5.2, Dell PowerEdge 2450
Asterisk
installed from ports (1.0.1)
Only
using IAX2 (VoicePulse) and SIP (clients)
Oct 31 18:34:29 NOTICE[165595136]:
chan_iax2.c:2442 iax2_read: I should never be called!
Oct 31 18:34:30
The chart is good, but I think it makes a mistake for iLBC:
Isn't iLBC 13.something kbps?
Also, since iLBC uses 30ms frames (when used with asterisk, at least),
it has slightly lower overhead. Approx 2/3 as much overhead.
(not that I'm a big iLBC fanboy or anything.. -- I still prefer a free
co
Christopher L. Wade wrote:
Kanuri, Seshu (Company IT) wrote:
Why are we allowing this A**hole to spam all of us to visit his site
which has nothing to do with Asterisk or VOIP?
Read the rest of the posts in this thread, I've already defended
myself once, do I need to do it again? [*** + * + ** =
Andrew Edmond wrote:
Message
*Community,
I
am a new VoicePulse customer, using their EXCELLENT
connect.voicepulse.com services.
I
have asterisk CVS head 10/20/04 running quite successfully, with 10 SIP
phones, voicemail, and 2 zaptel lines. The box is running Gentoo
Li
Adam Goryachev wrote:
On Wed, 2004-10-27 at 13:37, Jim Van Meggelen wrote:
People will want to pay for your expertise because you wrote (or at
least contributed to) the base platform, or language, or what-have-you.
The more one contributes, the more their credibility is established
Shawn Dillon wrote:
Thanks to all who have
helped me build and test out Asterisk
installation thus far. I needed to move my * installation to a new box
, due to
the fact my test machine would not support PCI 2.2 ( which I am told is
required to use my TDM11B).
I have * up
Stefan de Konink wrote:
Isn't this an opportunity for Digium to offer encoded G729 files for a
fixed price directly encoded from the original wav files?
I think this is an opportunity for people to use unencumbered codecs..
If even just the asterisk community got together to put half their G729
Benjamin on Asterisk Mailing Lists wrote:
This is a question I am almost too embarassed to ask but here we go ...
Is it not possible to use the i extension to trap attempts of users
misdialling numbers otherwise not in the dialplan/context?
I have seen this in so many examples and I always thought
Mike Taht wrote:
Lately I've been working in relative isolation (e.g. at home) and I
find I like the idea of sharing a virtual audio room with my fellow
programmers. (We all share thoughts currently via irc).
So I'm listening in a asterisk conference room right now (using
SJphone, everyone mic
Could be this:
From: Jon Lewis <[EMAIL PROTECTED]>
Cc: [EMAIL PROTECTED]
Subject: Re: Level 3 US east coast "issues"
On Mon, 18 Oct 2004, Grant A. Kirkwood wrote:
Level 3 experiencing widespread "unspecified routing issues" on the US east
coast. Master ticket 1086844. Anyone have more specific info
Brian West wrote:
Anyway we could talk you into releasing the source? I would love to see
wider codec support. And the ability to launch the URL sent with the IAX
call.
Brian,
The codec stuff I did, and the source is all available at
iaxclient.sf.net. Afaik, all the existing IAX softphones
Michael George wrote:
We are trying to get a softphone working on a gentoo system and are having
trouble.
kphone isn't working on the laptop, complaining about sound access
iaxcomm doesn't start up: "Fatal Error: Couldn't Initialize IAX Client"
AFA iaxcomm, this might be because your audio devic
On Sep 23, 2004, at 8:10 AM, Tom Ivar Helbekkmo wrote:
Steve Kann <[EMAIL PROTECTED]> writes:
I don't think it's in the Wiki, and it's not really documented;
Could you offer a very, very brief introduction? I've figured out,
through trial and error, that it takes a call
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