i've just lost $2000 dollars or so on my first commercial asterisk
installation ..
i'm running a PIV class server, three Digium Wildcard FXO cards, and
10 Grandstream Budgettone SIP phones. The system was to be a PBX
for a small company. After over 2 months of pissing about, the client has
had his
tion error?
>
> Tan
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan
> Caruana
> Sent: 03 October 2003 11:49
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Budgettone + G729
>
>
> hi there ..
>
hi there ..
I asked sometime ago regarding getting a Budgettone
working with Asterisk over G729.
My system is quite simple, Asterisk server with 1 G 729 license
installed, and 10 Grandstream phones. Only one of them needs
G729, because it's on a remote link via an ADSL bridge. The
rest run happily
what i'm asking is what is the key sequence
you have to dial for the transfer ..
it was something like *7# if I remember
well, I know I had it working, but the client
lost the paper I wrote it on for him, and I can't
trace the email I got it from!
cheers
Dave
- Original Message -
From: "
hi great gurus of asterisk :)
could somebody remind me the key combination to send a call
into the parking queue ?
while you're at it, are there any other key combinations I should know??
eg. put a call on hold etc.
thanks
Dave
___
Asterisk-Users ma
it's 512k/128k actually ...
:)
Dave
- Original Message -
From: "Ing. Angel Gomez" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, September 04, 2003 9:23 PM
Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems
..
> WipeOut . wrote:
>
> >>now .. i have
has anyone got G729 and SIP working together?
some config examples would help :)
since I need to do this at a client where I don't
really have internet access, or the will to root
around mailing lists with the client breathing down
my neck!
thsnk
Dave
- Original Message -
From: "WipeOut .
mber 04, 2003 5:53 PM
> > Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo
problems
> > ..
> >
> >
> > > Hello,
> > >
> > > Have you succeded to use flash key to do call transfert?
> > >
> > > Regards,
> > >
dstream + SIP + Echo problems
..
> Hello,
>
> Have you succeded to use flash key to do call transfert?
>
> Regards,
>
> Daniel
>
>
> Dave Alan Caruana a écrit:
>
> >well .. good news :)
> >
> >i've just put in
> >txgain=1.0
> >rxg
well .. good news :)
i've just put in
txgain=1.0
rxgain=1.0
in my zapata.conf
and upgraded the Grandstream Budgettones i'm using to version 81
of the software and all seems fine .. there is still an echo but after
the first couple of seconds of call it vanishes, as the echocancelling
kicks in ..
Redhat 9 works fine unless you really need G729
working on H323
in which case the only solution seems to be
chanh323, which
only works with G729 support on Redhat 8 .. I found
out the
hard way :)
cheers
Dave
- Original Message -
From:
Ernest W.
Lessenger
To: [EMAIL
I tried specifying rxgain & txgain,
copied the format some some message on asterisk-users
Result was asterisk bombed out & didn't even load
due to not being able to understand the config file ..
what's the exact syntax that works??
cheers
Dave
- Original Message -
From: "Fredrik Hedberg"
hi ..
i have the exact same problem you have
..
seems to be related to Budgettone phones in my
prob.
I *tried* selling an asterisk exchange to a
client
and today he phoned telling me he is very
unsatisfied
& I risk being thrown out .. suggestions would
be
welcome! i've tried *everything* t
maybe because your email seems ot be
encoded within an attachment?
try sending plaintext!
cheers
Dave
- Original Message -
From: "Armand A. Verstappen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 25, 2003 7:02 PM
Subject: Re: [Asterisk-Users] Why doesnt anyone reply
my error .. the cards are X100P which is why I wrote FXO.
The Grandstream phones are on a LAN, the * server connects to the phonelines
via the X100P cards. When I call from the Grandstream phones onto the PSTN
there is a VERY big amount of echo, ie. I can hear myself in the earpiece.
cheers
Dave
anything that supplies a reasonably straight 5v should work ..
do not send *more* than that into the phone which is what
most unregulated supplies will do ..
the supply which comes with the grandstream seems to
be a nice switchmode one.
cheers
Dave
- Original Message -
From: "WipeOut ." <
you have to make /etc/zaptel.conf
and /etc/asterisk/zapata.conf
match on the same type of signalling ..
should work then :)
cheers
Dave
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 08, 2003 12:24 AM
Subject: [Asterisk-Users] Newbie Issue
>
could you send me the exact syntax for rxgain / txgain?
I think that might help towards my problem
becuase i'm having to turn the handset volume all the
way up ..
thanks
Dave
- Original Message -
From: "WipeOut ." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 05, 2003
I tried putting in
txgain=100%
rxgain=100%
and zaptel wouldn't load telling me I had wrong parameters in my zaptel.conf
i'll try again with txgain=5.0 but my setup is at a client so each time a
day passes
and i have to go round to the client just to try things out ... it's a bit
annoying!
my 2c
ly put a check
> in that winner was non-zero before comparing it to o->chan:
>
> if (winner && winner == o->chan)
>
> Adam
>
> Dave Alan Caruana wrote:
> > I have an asterisk installation at a client, it's quite simple.
> > Basically it'
hi ..
I have an asterisk system with three TDM100P (single port FXO) cards
and 10 Grandstream 100 phones connected to it ..
1st question:
when i phone out
or receive a call from one of the SIP phones onto the PSTN, there is
a LOT of local echo in the handset .. the PSTN end of the call does not
t;
> if (winner && winner == o->chan)
>
> Adam
>
> Dave Alan Caruana wrote:
> > I have an asterisk installation at a client, it's quite simple.
> > Basically it's an asterisk downloaded from CVS about
> > a week ago, with 3 Zaptel FXO cards (the digi
me on #asterisk so I can login. Be sure you're generating cores
> and running on very latest CVS.
>
> Mark
>
> On Thu, 31 Jul 2003, Dave Alan Caruana wrote:
>
> > I have an asterisk installation at a client, it's quite simple.
> > Basically it's an aste
I have an asterisk installation at a client, it's quite simple.
Basically it's an asterisk downloaded from CVS about
a week ago, with 3 Zaptel FXO cards (the digium ones)
and 10 Grandstream Budgettone SIP phones ...
Every now and then, especially when a call is ringing
and not picked up immediatel
oh ok ;) just understood!!
call transfer is something the phone does
and asterisk picks up, not some sequence
you send directly to asterisk, hence from
the Grandstream manual :)
thanks very much for pointing it out!
cheers
Dave
- Original Message -
From: "Dave Alan Caruana&quo
Sip phones on the system are Grandstream Budgettone 100's.
Was assuming it wouldn't be phone specific :)
they have flash key which is meant to send a DTMF.
thanks for the help with the dial string.
Dave
- Original Message -
From: "Low, Adam" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
just three "stupid" questions I need to ask ..
1. what's the sequence to press on a SIP phone to transfer a call to another
extension.
2. what's the same thing if you want to hold an incoming call, speak to the
other extension, then pass the call?
3. what's the extensions.conf syntax to dial two
Is there any such thing is a userguide for asterisk from an enduser point
of view ie. what to do to transfer a call etc ? I've looked through all
the official documentation and nothing exists, and trying to install an
ASterisk at a client can't even explain how to transfer a call to another
extensi
hi ..
i require to interface asterisk to a 60 line analog PBX in a hotel.
I was thinking of giving Asterisk a couple of PBX lines interfaced
through cards, and then place outgoing calls through SIP/H323 and
a DSL connection.
analog extension lines <--> analog pbx <-->asterisk <--> SIP --> termin
problem)
> You're trying to detect inband dtmfs from the codec stream.
>
> Martin
>
> On Tue, 15 Jul 2003, Dave Alan Caruana wrote:
>
> > hi ..
> >
> > I have finally managed to get Chan_H323 & G729 working
> > flawlessly, thanks to some help fr
the probable reason is that you do not have the right
codecs so your call cannot go through. Your best option
is to switch on the trace option and see exactly what is
happening. If you are using chan_h323 the command
is
h.323 trace 3
for oh323 you have to change the "libtracelevel" in the
oh323.c
hi ..
I have finally managed to get Chan_H323 & G729 working
flawlessly, thanks to some help from Jerry McNamara.
For those out there who are stuck with the same problem
the procedure is :
1. install on RedHat 8.0 and nothing else (RH9 doesn't work!)
2. Install asterisk, zaptel etc. the normal way
;16>
G.711-uLaw-64k{sw} <17>
G.711-ALaw-64k{sw} <18>
SpeexNarrow-5.95k{sw} <19>
SpeexNarrow-8k{sw} <20>
SpeexNarrow-11k{sw} <21>
SpeexNarrow-15k{sw} <22>
SpeexNarrow-18.2k{sw} <23>
G.723.1{sw} <24>
GSM-06.10{sw} &l
I think the problem is in your extensions.conf
..
try this out :
[incoming]exten => s,1,Wait,1exten =>
s,2,Dial(extension number)exten => s,3,Hangup
substitute which extension it's meant to ring
..
Also try adding the line
"immediate=yes"
to your zapata.conf
it works for me :)
PM
Subject: Re: [Asterisk-Users] OH323 + G729 + Go2Call
> I get an IVR when I use chan_h323 and Digiun's G.729.
>
>
>
> Jeremy McNamara
>
>
>
> Dave Alan Caruana wrote:
>
> >hi ..
> >i've just installed and licensed an instance of the G729 code
hi ..
i've just installed and licensed an instance of the G729 codec.
I am trying to connect through asterisk to Go2Call server ..
According to their info it involves dialling extension 729 on
voip01.go2call.com, to get the IVR.
my extensions.conf shows :
exten => s,2,Dial(OH323/h323:[EMAIL PROTEC
hi ..
I have a very simple asterisk system which is not working properly ..
basically I have an asterisk box, with a PRI-E1 card accepting calls,
and immediately forwarding them, via SIP, to [EMAIL PROTECTED]
which is a commercial calling gateway.
Sometimes it works, but very often I get this mes
regarding my previous post about SIP outgoing calls
dropping with an error 481 ..
this is my output from a SIP debug.
the call dropped occurs at the end.
Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my
control.
help :) please!!
Dave
Signal=0
Duration=250
(no NAT) to
-- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207
== Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1'
I am getting this error on an outgoing call to a SIP host.
The call just disconnects ..
is there any way around it ?
thanks
Dave
_
i'm getting Asterisk to dial an h323 call termination service ..
right now getting this message:
-- Executing Wait("Zap/1-1", "1") in new stack
-- Accepting call from '21382890' to 's' on channel 1, span 1
-- Executing Dial("Zap/1-1", "OH323/h323:[EMAIL PROTECTED]") in new
stack
5:5
I have just compiled & installed the latest oh323, on a fresh asterisk
installation
however using a previously working oh323.conf file.
When I try to dial an outbound oh323 call I get the following error :
-- Going to extension s|1 because of immediate=yes
-- Executing Wait("Zap/1-1", "1"
hi ..
when placing a SIP call to a sip host in the states
every few seconds I get an RTP codec 19 error.
I know this is related to comfort noise, and the
call goes through OK ... how can I suppress
the error message ?
Also, many times I get "Invalid CSeq Number"
back from 216.52.153.207 (which is
hi ..
when placing a SIP call to a sip host in the states
every few seconds I get an RTP codec 19 error.
I know this is related to comfort noise, and the
call goes through OK ... how can I suppress
the error message ?
Also, many times I get "Invalid CSeq Number"
back from 216.52.153.207 (which is
what is the format for an h323 entry in the dialplan?
can I use chan_h323 without compiling anything else
or should I compile oh323?
basically what's the best way :)
cheers
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
problem solved - forgot to update zaptel.conf
stupid me!
thanks guys :)
Dave
- Original Message -
From: "Dave Alan Caruana" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 30, 2003 2:55 PM
Subject: Re: [Asterisk-Users] E100P installation sheet
&
d server.
Dave
- Original Message -
From: "Tais M. Hansen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 30, 2003 2:29 PM
Subject: Re: [Asterisk-Users] E100P installation sheet
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On M
hi ..
maybe someone can help me,
I seem to have lost the sheet of paper that comes
with an E100P card and tells you how to compile
the stuff it requires to run.
I'm trying to move my Asterisk to a different
box and at this time totally stuck.
Could someone be kind enough as to mail
me a PDF of it ?
hi there..
I have an asterisk installation with a PRI-E1 card
running EuroISDN, installed on a 1GHz Intel Celeron
box with 256Mbytes RAM.
CPU usage is stuck at 100% all the time, even with
no calls going through. Is this the normal ?
Running "top" reveals that the CPU allocation is
99.6% to Asteris
nd out which way your SIP gateway wants to receive the
> DTMFs. There are three ways to do that. Read sip.conf.sample.
>
> Martin
>
> On Mon, 23 Jun 2003, Dave Alan Caruana wrote:
>
> > hi there,
> > I have an installed & working Asterisk server,
> > which
My system is an asterisk machine,
with an E1 card (functioning) and
forwarding calls to a remote SIP
address ..
when a call connects I am getting the
following error :
NOTICE[1240577216]: File rtp.c, Line 330 (ast_rtp_read): Unknown RTP codec
19 received
can anybody tell me what this means
(& h
hi there,
I have an installed & working Asterisk server,
which I am using to connect to a SIP service
abroad. Although I can hear the IVR from the
ITSP, I cannot seem to send them digits from
my phone.
I have also noticed that the CPU usage on my
machine is up to 100% constantly and 99.9%
of that
not really asterisk related this,
but is it normal for a mail to take so long
to be resent through the mailing list server?
i'm speaking about 20 minute + delays here ..
(or it it only me ?)
cheers
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED
hi guys,
I have a little problem maybe you can help ...
I have an asterisk setup, with an E100P, and an ISDN-PRI 30 channel
line from the telco going into it .. the E1 line is OK, because plugged into
a Lucent Portmaster 4 it works OK .. plugged into the asterisk box
I just get an engaged tone, and
d need
an IP for my hosts.allow :) (& would be grateful too!)
cheers
Dave
- Original Message -
From: "Michael Manousos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 09, 2003 8:54 PM
Subject: Re: [Asterisk-Users] OH323 crashing
> Dave Alan
hi,
does anyone have a problem with OH323 crashing
with a segmentation fault whenever anything tries
to connect to it ??? are the current CVS versions OK?
Would like to speak to someone with a bit of OH323
experience, so if u're in a good mood to help,
please do :)
cheers
Dave
_
I added the line "allow G723.1" in my sip.conf general config,
and from a bridge connection which gives silence,
I have progressed to the error message below,
and the call gets rejected.
help!!
Dave
ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant
Expressa
[EMAIL PROT
one.
> If you are a developer, you can register for a G.729 codec from SJLabs.
>
> BR,
> Dan
> P.S. Have you tried X-Lite? It has G.711ulaw, G.711a law, GSM and iLbc.
>
>
> - Original Message -
> From: "Dave Alan Caruana" <[EMAIL PROTECTED]>
> To:
i've been having a problem getting two SIP phones
to bridge running through asterisk, actually one is
a SIP softphone, SJ Phone, and the other is the
Go2Call calling gateway.
Someone suggested that I don't have the right codecs.
How do I find out which codecs are installed, and how
can I install f
ent: Thursday, June 05, 2003 4:33 PM
Subject: Re: [Asterisk-Users] a little oh323 questoin
> Dave Alan Caruana wrote:
> > this might be a better dump:
> >
> > #0 0x41ec7279 in ast_oh323_new (i=0x810e538, state=0) at
chan_oh323.c:1170
> > #1 0x41ec786e in oh323_request
ead.so.0
hope u're still around to help!! (Michael, ie)
i've been away from office for 2 days ..
cheers
Dave
- Original Message -
From: "Michael Manousos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, June 03, 2003 6:17 PM
Subject: Re: [Asterisk-
39 PM
Subject: Re: [Asterisk-Users] a little oh323 questoin
> Dave Alan Caruana wrote:
> > doesn't seem to be dumping a core at all
> > if it is, can't find it.
>
> Turn it on by running:
> ulimit -c 100
>
>
> Michael.
>
>
> > Dave
doesn't seem to be dumping a core at all
if it is, can't find it.
Dave
- Original Message -
From: "Michael Manousos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, June 03, 2003 5:23 PM
Subject: Re: [Asterisk-Users] a little oh323 ques
EMAIL PROTECTED]") in
new stack
Segmentation fault
help!! :)
cheers
Dave
- Original Message -
From: "Michael Manousos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, June 03, 2003 4:28 PM
Subject: Re: [Asterisk-Users] a little oh323 questoin
> D
PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan
Caruana
> Sent: Tuesday, June 03, 2003 8:27 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] a little oh323 questoin
>
>
> hi,
> just wanted to know what's the proper syntax for an h323 extension.
>
&g
hi,
just wanted to know what's the proper syntax for an
h323 extension.
exten => 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
dials SIP extension 723 on IP
216.52.153.207,
what is the h323 equivalent of that ??
cheers
Dave
tion ..
Hi Dave,
If you have registered the SIP phone with
Asterisk, then you must have a line like:
exten => 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
---
ECTED],52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
----- Original Message -
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Friday, May 30, 2003 6:21
PM
Subject: Re:
isk, then you must have a line like:
exten => 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
- Original Message -----
From:
Dave Alan
Carua
with
Asterisk, then you must have a line like:
exten => 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207)
in extensions.conf file
Then call 555 from the SIP phone to access the
destination.
BR,
Dan
- Original Message -
From:
Dave Alan
M support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
- Original Message -
From:
Dave Alan
Caruana
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 9:01
PM
Subject: [Asterisk-Users] a beginner'
I am trying to get asterisk to dial this address
:
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice prompt
..
I have configured an extension, 1303 on
asterisk,
modifying the demo configuration :
exten => 1303,1,Dial(SIP/[EMAIL PRO
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