Hello,
For those interested in overcoming the problem with
some NATs and Firewalls in regards to tftp. I found a nice little tftp
server here:
http://freshmeat.net/projects/jtftp/?topic_id=87
I tried it and it works great.
Regards,
Andres.
I'll try to hack a NAT friendly tftp server on monday.
Are you still looking for it? I found one if you need it. Let me know and
I will post the info.
Andres.
--
Nicolas Bougues
Axialys Interactive
___
Asterisk-Users mailing list
[EMAIL
Don't know if this has been addressed, but why isn't there a
file_include style directive for extensions.conf?
there is...search the archives or the wikiits something like #include
filename.conf
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
From a Tilghman post:
Uncomment OLD_DSP_ROUTINES near the top of dsp.c, recompile,
install, and restart.
The newer DSP routines are used to fix a type of signalling on EM
lines.
- Original Message -
From: Stephen J. Wilcox [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January
Hi Matteo,
Send me the Ethereal SIP Trace and I will take a stab at it.
Regards,
Andres.
- Original Message -
From: Matteo Brancaleoni [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 4:50 AM
Subject: Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
Hi.
Here's the ethereal dumps (running on the * server):
one is when calling from the GS to a cisco phone,
the other is viceversa.
I dumped only the traffic between the * server and the GS
Running firmware 1.4.30
Matteo.
Il mer, 2004-01-07 alle 14:44, TeleSIP ha scritto:
Hi Matteo
Hi Terence,
I can take a look at the traces if you want. Just repeat the test using
g711ulaw and use Ethereal to capture the SIP messages and RTP stream of the
phone that hears bad sound, and if you can, of the other phone too (the one
that hears fine). Send me the captures and I will see if
Wipeout, If you want you can send me an Ethereal trace of the RTP stream
and I can do an analysis of it to determine if there is anything obvious
there. (please use G.711, and try something like counting from 1 to 20).
Regards,
Andres
- Original Message -
From: WipeOut [EMAIL
We are in the process of doing testing with the SPA and our SER servers. We
have not seen your problem and we are using 1.0.18. We have seen a nasty
G.729 codec problem when interoping with the GS Phones. I have consistently
reproduced the problem for Sipura and hopefully they will fix it.
Hi TC,
I followed your instructions. Today I managed to lock up one of our
Asterisk boxes with an incoming SIP call.
It locks up right after seeing I this line:
Dec 8 13:48:34 WARNING[4101]: File chan_sip.c, Line 456 (retrans_pkt):
Maximum retries exceeded on call [EMAIL PROTECTED] for seqno
Hi Mark,
Here are some details:
1. Asterisk is still locked, if you need anything let me know please.
2. Asterisk box locked up when processing 1 incoming SIP call (there we no
other calls at the time).
3. Seems to lock up because it receives no ACK to a STATUS 200 OK message
during call
about 100 calls and no lockup (with the
buggy phone we were able to lock it up in under 7 calls).
CVS should now reflect his fix.
And by the way, do not use firmware 1.0.4.18 on GS phones. It contains a
nasty SIP Port bug.
Regards,
Andres.
- Original Message -
From: TeleSIP [EMAIL
PROTECTED]
Sent: Monday, December 08, 2003 10:29 PM
Subject: Re: [Asterisk-Users] Asterisk freezing HELP
TeleSIP wrote:
And by the way, do not use firmware 1.0.4.18 on GS phones. It
contains a
nasty SIP Port bug.
Where do you guys come up with this Grandstream firmware that isn't
Its the VT1000
http://broadband.motorola.com/catalog/productdetail.asp?ProductID=212
We have looked everywhere for it but looks like no distributor sells it
right now.
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 07, 2003 4:50
Thats odd. We have the firmware on our Linux RH7.3 tftp server. The GS
phones can download it just fine on the LAN.
We would like that NAT-Friendly tftp though:)
- Original Message -
From: Nicolas Bougues [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 6:36
I am also looking for a NAT-Friendly tftp server too. Let me know if you
find one please.
Thanks,
Andres
- Original Message -
From: Nicolas Bougues [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 11:18 AM
Subject: Re: [Asterisk-Users] GrandStream Budgetone Phone
Running multiple boxes with RH7.3 and RH9.
They all freeze every few days and always when processing an inbound SIP
call. Sometimes it will freeze 5 times in a row when processng a SIP call
from the same user.
We have been unable to reproduce this so no bug has been opened.
The only suspicious
I sure will try. (I am working on it right now)
Thanks,
Andres.
- Original Message -
From: TC [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 4:47 PM
Subject: Re: [Asterisk-Users] Asterisk freezing HELP
The only suspicious thing is that they always seem
A good rootkit will also modify the date and time of the replaced binaries
so they will look the same as the original.
Try to replace your ps command with that from a trusted RH9 machine. If
it works ok then you must do a clean install to get rid of the rootkit.
- Original Message -
The i2004 does not handle SIP natively. There is an addon-software for a PC
that will do SIP with an i2004. But the software is not available publicly.
- Original Message -
From: Alexander Romanov [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 9:00 PM
Subject:
Hi,
I would like to know if those using the Sipura SIP
units with Asterisk have found them to be stable. I ask because the
Grandstream units simply have not improved their stability considerably, and we
are now in search of an alternate to the ATA186. We want to know if
anybody has seen
Another trick to look into is cutting down of the asterisk modules that
are loaded. Our newly deployed asterisk machine is not going to do any
IVR, and is only going to do IAX2 traffic. Because of this I was able to
trim down a lot of the modules being loaded by asterisk and decrease
it's
It seems that everyday we see these complaints about bad echo on X100P
cards.
Why can't these cards incorporate an echo canceller that a cheap $10 dollar
phone bought at Walmart can?
If we plug a cheap phone on the line there is zero echo. If we plug an
X100P on the line there is horrible
Try this:
In your [general] section:
disallow=all
allow=ulaw
allow=alaw
this forces * to only accept ulaw and alaw codecs.
- Original Message -
From: Arslan Saeed
To: [EMAIL PROTECTED]
Sent: Saturday, November 15, 2003 3:11 PM
Subject: [Asterisk-Users] Internal server error - cannot
Mark,
So far we have been unable to determine how to set the jitterbuffers
parameter in zapata.conf. We can guess that it represents a multiple of
bytes of memory be we are unsure. We have tried to set it to a value of
1000 and it makes communication impossible on the zap side. The zap side
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 26, 2003 3:11 PM
Subject: Re: [Asterisk-Users] Asterisk on FreeBSD
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD
server.
On a slower CPU linux system, Asterisk
How does the CISCO ATA sound quality, functionality and stability
compares
to the Grandstream phones?
Sound Quality using G.729: Grandstream phones are superior. They sound
perfect even with slight packet loss. The ATA will sound very good with 0%
packet loss but if you ramp it up a bit it
Also I found that ATA use silence supresion on the second phone even if
they have silence supression deactivated.
What is the setting of your AudioMode: ?
--
Juanjo sin .sig
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
If you use g729 codec The Voiceage part of the codec breaks it.
In our case we use the provided script called init.asterisk and place it
in /etc/init.d
Then we start it by executing service init.asterisk start. We use g729
all day long without issues.
regards
Martin
On Thu, 16 Oct
put a comma after "Dial"
- Original Message -
From:
John
Foster
To: [EMAIL PROTECTED]
Sent: Monday, October 13, 2003 5:27
AM
Subject: [Asterisk-Users] Extension
Dialing problem with SIP
Hi List..
I m getting this mesg while trying to dial an
LeterheadThe FAQ at digium explains how to do it:
http://www.digium.com/index.php?menu=faq#Configuration_7
- Original Message -
From: David J Carter
To: [EMAIL PROTECTED]
Sent: Friday, October 10, 2003 5:29 PM
Subject: RE: [Asterisk-Users] X100P Config
Hi,
I can see the card with a
Hi Jeremy,
The handbook says:
user: A user can place calls to or through the Asterisk server.
peer: A peer receives calls from the Asterisk server, but does not
place them
friend: A friend both sends and receives calls through the Asterisk
server. This makes the most sense for handsets or
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 09, 2003 2:57 PM
Subject: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2
Trunking confirmation?)
WipeOut wrote:
Jeremy, Can you elaborate on how using
33 matches
Mail list logo
