Will these be available on the CVS? Devel or Stable?
> Hi all,
>
> in an effort to create a SIP <-> H.323 translator we've found and
fixed
> several problems in H.323 channel. These inlcude:
>
> for SIP->H.323 calls
>
> - no ringback tone
> - ringback not related to H.323 events
> - one-way
on the
Message Board archive. Any help/hints appreciated.
- SamW
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The bug seems to be fixed on the latest stable cvs branch. Can someone
confirm if this is not a bug anymore. I tested and works fine for me with a
macro being used to dial out. But I see incorrect call-times. Also ResetCDR
command do not seem to work as expected asterisk spits out a warning
mes
What is the method you are using to test the bandwidth. Can you give us a
outline how to do a bit rate measurement on asterisk.
At 04:29 PM 3/8/2004 -0800, you wrote:
Hello
all,
I'm looking for advice for codec that works
best for asterisk. Anyone has real testing with all codecs,
specially w
er extension at line 50
== Parsing '/etc/asterisk/h323.conf': Not found (No such file or directory)
== Parsing '/etc/asterisk/iax.conf': Found
Killed
- SamW
RTP stream not passed through the * server in case of SIP(Sipura) >
H323(Cisco) traffic (Grand Stream was doing OK so we suspected SIPURA).
After SIPURA firmware upgrade (What ever latest) started working
correctly. No confirmed reason but Firmware upgrade did the trick.
- S
Bug submitted, for this missing functionality. Bug ID : 1141. Thanks for
who ever contributed.
- SamW
-Original Message-
From: Philipp von Klitzing
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 25, 2004 2:45 PM
To: SamW
Subject: RE: [Asterisk-Users] cdr->dst incorrect?
the server would have done something. This
may be useful if you are having license issues with the G729 codecs.
Cheers!
-SamW
-Original Message-
From: Darren Wiebe [mailto:[EMAIL PROTECTED]
Sent: Monday, March 01, 2004 12:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G729
appreciated.
- SamW
-Original Message-
From: T. Chan [mailto:[EMAIL PROTECTED]
Sent: Friday, February 20, 2004 12:48 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE: codec negotiation prob solved
I have the same problem, most carriers out there deal with both g723.1
or
g729
Did you try to upgrade the firmware?, some issues we saw with rtp
stream, went away after a firmware upgrade.
http://www.sipura.com
-SamW
-Original Message-
From: Mark Messmore, Technical Support, University Telcom Inc.
[mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 02, 2004 1:59 PM
tle more details, to indicate this as well.
-SamW
>If you are not using Digium hardware, then you don't need libpri and
zaptel.
>Asterisk WILL build on its own.
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0.5.2.tar.gz
ftp://ftp.asterisk.org/pub/telephony/zaptel/zaptel-0.8.1.tar.gz
Which one of the 2 above should be used for a stable Asterisk build. (I
do not use digium hardware)
I am currently seeing lot of segmentation faults (core-dump) when I
running asterisk. Help is appreciated.
-
I went back on my old CDR's they are correctly recorded, so is this
feature introduced recently with a new version upgrade ?
>> instead of macro-dialout if I directly dialed through the [intern] I
get
>> the correct results. Some how asterisk think I dialed extension "s"
instead
>> of the numbe
st place, but
there might be
another reason that you want to use a macro...
**
http://lists.digium.com/pipermail/asterisk-users/2004-January/033970.htm
l
Hope it solves your problem...
Regards, Girish
>From: SamW <[EMAIL PROTECTED]>
>Reply-To: [EMAIL PROTECTED]
>To: [EMAIL PROTECTE
gt; _1NXXNXX,1,Macro(dialout,${EXTEN},60)
- SamW
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-o chan_iax2.so chan_iax2.o iax2-parser.o
-lmysqlclient -lz
/usr/bin/ld: cannot find -lmysqlclient
collect2: ld returned 1 exit status
make[1]: *** [chan_iax2.so] Error 1
- SamW
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Title: Digium X100P for $43
Digium X100P / new cards are is available on ebay for $43.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309
Hope this helps to who want to play with X100P! Are these being sold by Digium ? I don't know ??
- SamW
I think what the license registration program does is read a unique
DISK-ID number to install the license. I think they cannot read that
number from anything else other than IDE HDD.
- SamW
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 21
lease one IDE disk installed - g729 will work.
SATA Serial ATA Disk I have no clue how it works. Is SATA considered a
IDE disk or a SCSI disk ?
This is an issue that VoiceAge need to address soon.
- SamW
-Original Message-
From: Amaury Jacquot [mailto:[EMAIL PROTECTED]
Sent: Wednesday
hardware. I use * with Cisco ATA 186 adapters. Thank
you in advance.
- SamW
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Have you tried canreinvite=yes in the sip.conf ? This is what gurus
suggested to me when I had similar issues. But that did not work for me.
May be it might work for you.
- SamW
-Original Message-
From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
Sent: Monday, January 12, 2004 9:51
eature list do not list any feature with call-conferensing @
http://www.asterisk.org/index.php?menu=features )
Thank you in advance!
- SamW
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_h323.c, Line 216 (build_alias): Keyword alaw
does not make sense in type=h323
== Adding alias "slt-h323-o" to endpoint
== Registered channel type 'H323' (The NuFone Network's Open H.323
Channel Driver)
== H.323 listener started
Thank you,
SamW
Ooops, It was a type that is how I tried it. But Asterisk do not see to
have a way to force codec negotiation if possible. * always want to
tran$code. (Which cost $$ and quality)
My original config should be corrected to
Case 1
--
[sip-a]
disallow=all
allow=g729
allow=alaw
SamW
At
I always like to use * to negotiate a codec which both sip clients
support. But * do not try to go in that direction, * try to be in the
middle and try to translate(code convert) the rtp stream, which will
deteriorate the call quality and it will also cost for license if we are
translating using G
Title: Dial from command line?
I have 2 installations of asterisk. On CLI one server has Dial command. Other installation do not have Dial command on the CLI. What I am missing. How to enable dial command from the CLI.
Thanks,
- SamW
ATA and the remote end would then be able to
>negotiate a different codec.
On Mon, 2004-01-05 at 01:29, SamW wrote:
> Hello,
>
> I have been trying to get my coders to work without a conversion. I have
> read all the available asterisk documentation and support groups with
Hello,
I have been trying to get my coders to work without a conversion. I have
read all the available asterisk documentation and support groups without
any luck. Here is my issue. (Please feel free to ask questions if you do
not understand what I am talking about.)
I am using Cisco ATA-186 se
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