Excellent answer. Thank you very much.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Andreas Frackowiak
> Sent: Saturday, May 15, 2004 1:32 AM
> To: [E
Line one should be red/green, and black/yel are sometimes used to remotely
power devices or can be line #2. Also, red is ring, green is tip "right,
red, ring" in telco parlance.
At 17:17 5/16/2004, you wrote:
Ahhh this could be my problem! I just checked which
wires on the RJ11 cable had a volta
Did just that today, minor to no improvement, any other ideas perhaps?
At 14:00 5/16/2004, you wrote:
Thanks Jason...will check.
At 06:00 5/16/2004, you wrote:
Are you running the latest cvs-head on both boxes if not update and try again
Jason
___
Asteris
Hi Guys,
Thanks for all the replies :-). Really appreciate
it.
Just downloaded the latest cvs of asterisk and it
compiles fine now.
The latest source of say.c has the variables declarations at the start of the
correspondingfunctions as Juan said.
BTW, downloaded the PWLIB 1.6.6-1 and
I believe that the Janus patch 2 compatible Makefile was "rolled back" from CVS. The current Readme int he CVS (last time I checked) is wrong. I think the Readme never got rolled back. I've been anxiously awaiting the official "Janus patch 2" release of chan_h323, but have not seen it yet. Sti
What does 'cat /proc/interrupts' tell you?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Deepak
Malhotra
Sent: Sunday, May 16, 2004 11:09 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] (no subject)
i did that bit no luck.
- Original Message -
On Sun, 2004-05-16 at 22:20, Yap Teong Eng wrote:
> Hi All,
>
> I am using PWLIB-1.6.6-1 and Openh323 1.13.5-1 and running a RH7.3
> machine
> and I am unable to compile asterisk due to these errors.
>
> say.c: In function `powiedz':
> say.c:1633: parse error before `int'
> say.c:1636: `i1000E6
i did that bit no luck.
- Original Message -
From: "Todd Lieberman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, May 16, 2004 6:47 PM
Subject: RE: [Asterisk-Users] (no subject)
> You'll also have to modprobe the x100p
>
> /sbin/modprobe -k wcfxo
> /sbin/modprobe -k zaptel
>
Hi,
I’m wondering if it’s possible to get
Caller ID information from a Call Pickup… specifically on a Cisco 79xx
SIP handset.
I’ve setup a speed dial line on my 7940 to dial
*8 so I can quickly pickup a call, but because the 7940 initiates the call the
information on the screen is “
Thanks for the reply. All of the SIP phones and the Asterisk server are
on the local network (192.168.1.x) on the same side of the router (and
yes the router does have a NAT firewall). I would think the XLite to
XLite would work but it doesn't (yet). I am seeing errors on the
console when the S
Hi All,
I am using PWLIB-1.6.6-1 and Openh323 1.13.5-1 and
running a RH7.3 machine
and I am unable to compile asterisk due to these errors.
say.c: In function `powiedz':say.c:1633: parse
error before `int'say.c:1636: `i1000E6' undeclared (first use in this
function)say.c:1636: (Each undec
On Sat, 2004-05-15 at 12:22, Michael Welter wrote:
> I've gotten several "Power alarm on module 1, resetting" since I
> installed a quad FXS TDM400 card. Dell 400sc.
>
> Does your motherboard have the A-B-C-D LEDS above the keyboard/mouse
> connectors?
I suppose you plugged the power c
You'll also have to modprobe the x100p
/sbin/modprobe -k wcfxo
/sbin/modprobe -k zaptel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, May 16, 2004 9:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] (no subject)
Hello
Any working examples of configuration files is highly appreciated.
http://www.voip-info.org/wiki-Asterisk
:)
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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Hello
I am trying to configure Asterisk on RedHat Linux 9 with one X100P one port card
and one USB one port FXS card. I can modprobe wcusb but ztcfg always return
ZT_CHANCONFIG failed on channel 2: No such device or address (6)
error message.
Also unable to config outgoing call using SIP SoftPh
On Sun, 2004-05-16 at 15:15, Jorge Verastegui wrote:
> Hi
>
> Please help!
>
> I have one X101P and TDM400P in my asterisk Box
>
> When i make a call from * to PSTN, everything goes Ok,
>
> When the PSTN hangups or * hangups, the busy tone is detected and *
> disconnects the channel without pro
Note: some lines provide
"ring" on a third wire. If that's the case, need to bridge that wire to 3 or 4
(the middle pins in 123456).
I'm sorry I can't be more
specific (or even describe why).. I'm just a lay person who's interested but now
curious .. Bumble wha ??
- Original Messag
I think he mean an integrated system like sap, witch use * as a
horicontal layer.
nicolas
Christian Hoffmeyer wrote:
> Vertical applications?- Original Message -
> From: John Vogel
> To: [EMAIL PROTECTED]
> Sent: Sunday, May 16, 2004 11:51 AM
> Subject: [Asterisk-Users] Vertical applicat
Geert Nijpels wrote:
> You mean you are doing something like:
> exten => 666,Dial(SIP/snom)
Yes i do dialing a snom trough * before i answer the line.
> or are you using the SNOM as stand alone phone (dial directly to it's IP
> address)?
No never tried because i have a isdn line (chan_capi) her
Ahhh this could be my problem! I just checked which
wires on the RJ11 cable had a voltage across them and
it was the yellow and green (3 & 4?). From what
someone posted the other day it's supposed to be
Bumble Bee and Christmas Tree.
I did have to get a technician out to fix my line when
it was fi
nicolas wrote:
Snom is a great phone, especially in conjunction with Asterisk. See for
more information and tips and tricks the following URL:
http://www.voip-info.org/wiki-SNOM+phones
This is because i bought a snom.
In what context? If the SNOM is reply-ing this
I want to send date information that comes in PSTN-Caller*ID from
Asterisk to a H.323->PSTN gateway (an AudioCodes one) on the other
side, but Caller *ID in the phone shows
00/00/00 12:00 a.m.
"" <36>.
That is No ANI, no date.
I've found a solution to ANI problem (perhaps patch h323 channel)
Hi
Please help!
I have one X101P and TDM400P in my asterisk Box
When i make a call from * to PSTN, everything goes Ok,
When the PSTN hangups or * hangups, the busy tone is detected and *
disconnects the channel without problems.
The problem occurs when the call comes from PSTN. When * hangups,
> Trunking over IAX2 sounds very interesting, but it can't "tunnel" channels
as TDMoE does, does it?. I mean. Do I need Asterisk to pick up the calls and
redial or can I pass channels as is. I still need channels to be CAS
signaled.
Its going to work the same either way you go from the dialplan st
C. Maj wrote:
On Sun, 16 May 2004, Bruno Fontana waxed:
I was trying to use TDMoE and I lasted with two problems. First of all I
can't configure the dynamic span to use CAS signalling but documentation
(by Mark) says that you can use any type of signalling (and this
includes CAS I guess).
Vertical applications?- Original Message -
From: John Vogel
To: [EMAIL PROTECTED]
Sent: Sunday, May 16, 2004 11:51 AM
Subject: [Asterisk-Users] Vertical applications?
I'm trying to market * in my area (Seattle) and would like to offer vertical
apps to my customers.
--
So do
Sorry Rich, both ends use analog bell 500 sets. The system has trunks to
Iaxtel and FWD as well, but this occurs on all lines, no matter who how or
where the calls are placed. * to PSTN, PSTN to *, * to *, all the same
prob. Will look into that CVS. Thanks.
Several thoughts come to mind, but s
Thanks Jason...will check.
At 06:00 5/16/2004, you wrote:
Are you running the latest cvs-head on both boxes if not update and try again
Jason
At 05:50 16/05/2004 -0400, you wrote:
Let me run this by the group, my inter-IAX connections are extremely
pop-hickup,missing syllable type of affairs.
Play
Important.
1. Try the phone (set)
directly on the line.. - confirm you have dialtone
2. Make sure the phone is
picking up the line from pins 3 & 4 on the RJ11 ONLY .. i.e. if your line is
using a non-standard interface (and so does your phone) this is a possible
failure - not of the card, bu
Title: Vertical applications?
Has anyone created any vertical applications, e.g. real estate, for Asterisk?
I'm trying to market * in my area (Seattle) and would like to offer vertical apps to my customers. These apps will help me compete with the big guys like Cisco, Avaya, etc.
If you h
Hi!
> Grandstream v1.0.4.68 firmware
> http://www.hellofone.com/downloads.html
Hehe, the ringtones are fun, but we'll need someone to reveal how to
upload our own samples...
Can I keep iLBC frame size at 20 ms or do I need to change this to 30 ms
for better (?) operation with Asterisk? It appe
On Sun, 16 May 2004, Bruno Fontana waxed:
> I was trying to use TDMoE and I lasted with two problems. First of all I
> can't configure the dynamic span to use CAS signalling but documentation
> (by Mark) says that you can use any type of signalling (and this
> includes CAS I guess).
Well just
Hi,
Consider the following:
exten _X.,1, Dial(Zap/g1/12345&Zap/g1/678912)
we attempt to dial 2 numbers simultaneous and who ever answers get the call.
My issue here is that the cdr only contains like ,Zap/g1-1, which
doesn't tell if 12345 or 678912 answered the call.
One number could be local and
> Snom is a great phone, especially in conjunction with Asterisk. See for
> more information and tips and tricks the following URL:
> http://www.voip-info.org/wiki-SNOM+phones
This is because i bought a snom.
> In what context? If the SNOM is reply-ing this when idle, this is not
> normal behavio
nicolas wrote:
Hi all,
Got SIP response 486 "Busy Here" back from x.x.x.x
I become this message if a call is coming in and i have read this is normal
with snom.
In what context? If the SNOM is reply-ing this when idle, this is not
normal behaviour. It is however possible that you are running an
Hi,
I'm using a ZaptelBRI card. It works fine.
But I have a small problem with call logs.
The leading zeroes of the external calling party are not stored (e.g. : 0140302010
will be stored as 140302010).
Same for international numbers for which "00" will be stripped out.
I would not mind if the
Hi all,
Got SIP response 486 "Busy Here" back from x.x.x.x
I become this message if a call is coming in and i have read this is normal
with snom.
If it is so, then a right extension.conf is unuseable and snom be not the
right phone.
please help.
nicolas
_
Grandstream v1.0.4.68 firmware
http://www.hellofone.com/downloads.html
Seems to have loaded ok on my BT100..
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
On Sun, 16 May 2004 10:34:27 +0800, Lars Boegild Thomsen wrote:
>Well - the only problem I had with my Snom 200 was that after the update it
>mentioned that a new version was available - which was in fact an old
>version. I just disabled the automatic update and then no problem.
> -Original
> Let me run this by the group, my inter-IAX connections are extremely
> pop-hickup,missing syllable type of affairs.
> Played with jitter buffers, card levels (this is totally independent of any
> level setting) and about anything else we can adjust. Did the adjustments,
> undid them. Doesn't m
Another Asterisk week has gone by. A lot of changes has been submitted into
the CVS head, only a few to CVS stable.
CVS stable only changes for bug fixes now.
* Using MGCP? Please step forward!!
---
There are a number of MGCP bugs and fixes in the bug tracker that ne
Are you running the latest cvs-head on both boxes if not update and try again
Jason
At 05:50 16/05/2004 -0400, you wrote:
Let me run this by the group, my inter-IAX connections are extremely
pop-hickup,missing syllable type of affairs.
Played with jitter buffers, card levels (this is totally indep
Let me run this by the group, my inter-IAX connections are extremely
pop-hickup,missing syllable type of affairs.
Played with jitter buffers, card levels (this is totally independent of any
level setting) and about anything else we can adjust. Did the adjustments,
undid them. Doesn't matter, pro
hi,
does "sip show channels" show the real format ?
If i connect to sipgate the showing format is alaw allways,
But have gsm at the first place (allow=gsm).
nicolas
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hi,
anybody use a snoom with the g729 codec ?
For me the keys do not work if i use it.
nicolas
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Title: Asterisk-Users digest, Vol 1 #3774 - 14 msgs
Ha,
ha, looks like your rates was just a little too competitive and you went out of
business. I've tried to send you email direct and I get a confirmation
that the email was erased WITHOUT reading it. Quite an interesting
approach to acco
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