that you can't boil eggs on an
"electric" guitar.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Ian Pattison
> Sent: Tuesday, May 24, 2005 4:00 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [A
One would think if it can generate it it can survive it as well
>>> [EMAIL PROTECTED] 24/05/2005 04:07 >>>
Ian Pattison wrote:
> Hi all,
>
> Yesterday, in an attempt to take back my phone room, I pulled everything
> apart as far back as the demarc and rebuilt
pt it without being destroyed?
I've tried sever hard and soft resets to revive the module, reseated the card
and the module and moved slots to no avail. Can anyone thing of anything else
to try?
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
M
I've had similar issues with TDM400P FXS ports and it turned out to be a
voltage issue. According to spec a North American FXS port (whether from your
ATA or your telco should generate a 89V, 20Hz AC signal for ringing. Some
systems try to use a low voltage ring (usually about 45V) which won't t
Hi All,
Is there a way I can pass custom callerID info out to the PSTN or my SIP
provider? Currently all outbound calls get the callerID of one of my DIDs no
matter what I do.
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
OURCE
--dport 5060 -j DNAT --to-destination ASTERISK_SERVER
$IPTABLES -t nat -A PREROUTING -i $EXTIF -p udp -m udp -s EXTERNAL_SIP_SOURCE
--dport 1:2 -j DNAT --to-destination ASTERISK_SERVER
Hope this helps,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext
I believe it is Zaptel only that becomes a problem. I'm running asterisk on a
2.6 kernel... the only concession I had to make was to use "make linux26" when
I compiled Zaptel.
Thanks,
Ian
>>> [EMAIL PROTECTED] 30/04/2005 19:10 >>>
I was reading on the wiki about the supported kernels and I __T
opermode = "FCC";
static int fxshonormode = 0;
set boostringer=1 instead of 0 and recompile Zaptel. The FXS ports will be
forced to generate 89V ring signals from now on.
Now if I can just stop the FXO ports from dropping calls
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Ass
Is this a SIP phone?
I had to upgrade the firmware on my SIP phones to alleviate this. It seems that
the phone would actually disable it's own keypad after dialling.
Ian
>>> [EMAIL PROTECTED] 29/04/2005 09:16 >>>
I am having an issue with the asterisk system not responding to dialed
numbers dur
Hi Paul,
An RJ-45 is designed to take an RJ-11 or RJ-12 connector as well. Just plug
them in.
Ian
>>> [EMAIL PROTECTED] 27/04/2005 14:02 >>>
I just received my TDM400 card from digium with 2 fxo and 2 fxs
interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS
phones. How do i inter
Hi All,
Can anyone help me out here? I'm having some issues configuring my IPTables
firewall to properly NAT SIP and RTP packets to my asterisk server hiding
behind it.
Here are my current rules:
#Inbound SIP to HERMES
$IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 5060 -j DNAT --to
Interestingly enough I'm looking to do the same for a Canadian English
version... does anyone to collaborate on this one?
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED]
WWW: http://www.technologyassociat
Hi All,
I'm looking for a single FXS port ATA capable of doing both SIP and IAX (not at
the same time of course). Can anyone make a recommendation?
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROT
I don't know how everyone else is doing but my woes are continuing. I'm really
starting to dislike these Digium cards.
Hardware:
Digium TDM400P (REV G according to the silk screening on the board) 2xFX0,
2xFXS purchased in August/September 2004
Dell Precision 420 (PIII-733, 512MB RAM nothing fa
]> 18/04/2005 16:06 >>>
See inline responses...
On Mon, 18 Apr 2005 10:43:30 -0400
"Ian Pattison" <[EMAIL PROTECTED]> wrote:
> 2. Low ringing voltage still (~44V AC). I have used the
>boostringer=1 option when loading wcfxs, did I miss
>something at compil
I don't know how everyone else is doing but my woes are continuing.
Hardware:
Digium TDM400P (REV G according to the silk screening on the board) 2xFX0,
2xFXS purchased in August/September 2004
Dell Precision 420 (PIII-733, 512MB RAM nothing fancy but not doing too much
either)
Software:
Zap
That's the nice thing about VOIP devices... Ethernet is the same the world over
no matter what the phone system is like. So long as you've got comptible power
you're all set.
Ian
>>> snacktime <[EMAIL PROTECTED]> 16/04/2005 19:20 >>>
Will sip/iax devices designed for European use also work in R
If I understand your question correctly ztcgf is not a module, it's merely a
rudimentary diagnostic utility. Run "ztcfg -vv" to get info on your zaptel
hardware.
>>> [EMAIL PROTECTED] 15/04/2005 17:24 >>>
Hello everyone:
I've had X100P running on asterisk 1.0.6 for about two
months. Each time I
one had a similar experience?
A snapshot of the phone's config can be seen at
http://www.technologyassociates.ca/phone.jpg
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED]
__
My specific issue has to do with ringing on my FXS ports.
A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect
my newer GE 2.4GHz cordless I never get more than 1/2 ring (it lights up and
works fine... just can't get a ring from it). Normally I'd assume that it's a
low
I'd try turning the RXGAIN and TXGAIN down some... they're probably what's
giving you the excessive side-band audio.
Ian
>>> [EMAIL PROTECTED] 07/04/2005 16:38 >>>
Hello everybody,
my setup consists of an asterisk server with a TDM400P and a couple of
softphones (SJphones) ... everything wor
44V
Off-hook 0V6V
Ring 44V 0V
Does this make any sense to anyone?
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PR
Measurement
On-hook 107V 49V
Off-hook 11V6V
Ring drops to 44V0V
Does this make any sense to anyone?
Thanks,
Ian
>>> [EMAIL PROTECTED] 04/01/05 3:24 PM >>>
Ian Pattison
running a VPN (of
sorts) I suspect that your SDP messages are getting screwed up somewhere.
What are the asterisk NAT settings in effect for each of the SIP phones? I'd
be inclined to turn them both ON to ensure that symmetrical RTP in being
used. Also make sure that canreinvite is OFF for b
the SIP phones? I'd
be inclined to turn them both ON to ensure that symmetrical RTP in being
used. Also make sure that canreinvite is OFF for both.
- Original Message -
From: "Ian Pattison" <[EMAIL PROTECTED]>
To:
Sent: Thursday, April 07, 2005 4:49 AM
Subject: [As
s try the opposite of what I have suggested and see what it does.
Change the Dial command as under and see how that goes.
exten => _9NXXNXX,1,Dial(Zap/2/${EXTEN:1},30, Tt)
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent: We
Hi Everybody...
Continuing the litany of problems I'm experiencing with my new system I'm
getting issues connecting between SIP phones.
A bit of background... I have an asterisk server running in a central location
where I have two incoming analog lines connected to FXO ports, two analog
phone
lan. Are you using 'Answer' or 'Dial' command?
1)If you are usind Dial command, do not use T or t flags
2)DTMF mode Inband works only for Ulaw. If You use any other codecs, use
RFC2833
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I've had that in there for a while now... no result for me.
Ian
>>> [EMAIL PROTECTED] 06/04/2005 13:46 >>>
I had a simliar problem with my C-302P SIP phones until I added
"dtmfmode=rfc2833" to my sip.conf
Ian Pattison wrote:
>Hi All,
>
>I'
ppears that the keypad is being disabled after dialling.
I've attempted to contact the vendor of the phones several times and have been
unsuccessful in reaching anyone.
Anyone had a similar experience?
A snapshot of the phone's config can be seen at
http://www.technologyassociat
never a complete ring or multiple rings. When connected directly to my
incoming lines they ring normally.
Normally I'd assume this was a power problem but at 0.1 REN?? Any other
ideas I can try?
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 2
31 matches
Mail list logo
