Does anyone know of a PCMCIA FXO card or even a USB one? I'm looking at
building an appliance out of a machine that has USB and PCMCIA but no
PCI.
Chris Tooley
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[EMAIL PROTECTED]
Please take a look at http://www.ietf.org/internet-drafts/draft-ietf-sipping-mwi-04.txt.
The snom phone tries to use the Message-Account line, if its present;
otherwise it will take the From header:
Messages-Waiting: yes
Message-Account:
sip:[EMAIL PROTECTED]
Voice-Message: 2/8
Vic Cross wrote:
On Sat, 7 Feb 2004, John Fraizer wrote:
snip all the trace data
Here are the configs:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 66.35.64.38 ; Address to bind to
context = default ;
Hi Geert,
Jason Ross wrote:
G'Day,
I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm
having DMTF problems no matter what configuration I try. And as yet I
haven't downgraded it to see if an earlier release makes a difference
Just wondering if anyone can provide some
I have only 2 64 bits on my server's mobo :(
I was wondering if it would work because the card's interface can fit into
the 64 bits slot. I guess I have to change the motherboard.
Regards,
Soragan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]
Soragan wrote:
I have only 2 64 bits on my server's mobo :(
I was wondering if it would work because the card's interface can fit into
the 64 bits slot. I guess I have to change the motherboard.
Regards,
Soragan
If you only need one analog channel infor your Asterisk box then you are
I only have 1 server which is dual p3 1ghz.
Its mobo only has 2 64 bits pci. :(
Regards,
Soragan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Sunday, February 08, 2004 10:07 PM
To: [EMAIL PROTECTED]
Subject: Re:
I only have 1 server which is dual p3 1ghz.
Its mobo only has 2 64 bits pci. :(
Not good. Throw it away in my house's trash can and I send
you 4 Pentium 2... more details in pvt. :-)
Isamar
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Hi All
does anyone know anything about the nortel i2004 ip phone.
It is very hard to find out if the unit is h323 or sip.
Can anyone please tell me info regardsing this.
All I can find on the net is manuals and what codecs it uses. but too
many websites contradict each other. So I have no
Hi,
-Original Message-
Subject: [Asterisk-Users] Problems with ATA's locking up..
Anyone had any problems with ATA's running 3.0 software locking up?
Nope, seems rock-solid here. Can you tell more about the circumstances this
occurs with ?
Florian
Hi Olle, you mail server thinks this is SPAM, so I resend it in the
mailing-list...
CS
-Original Message-
From: Christian Stredicke [mailto:[EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 12:54 PM
To: 'Olle E. Johansson'
Subject: RE: SNOM 200 silence suppression
Hi Olle,
we do
We are using a SCSI based IBM eServer x300 for our PBX. In setting this
unit up, we used a backup machine, which
was IDE only.
The problem that we are currently experiencing is that the voicemail
prompts are coming out the system so fast that the words overlap each
other, and sometimes are
Hello All.
Many times I've done a top and found that Asterisk is pinning the CPU,
even when Asterisk isn't being used (this is on a DEV box):
2044 root 15 0 5232 5228 2608 R98.6 8.5 626:58 0 asterisk
I'm running a recent build of Asterisk on Slackware (2.4.24 kernel):
www*CLI
-Original Message-
From: Paul Oster [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 07, 2004 8:22 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Snom 200 MWI Button\
I'm trying to get the MWI button to work with my Asterisk configuration. The snom is
accepting and responding
3.0.0 have some problems. Sometimes, ata answers to invite with Not found
or Busy here. This is a strange behavior.
I'm using now 2.16.2
Regards,
Gus
- Original Message -
From: Billy Huddleston [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 12:56 AM
Subject:
Hi,
Citeren Billy Huddleston [EMAIL PROTECTED]:
Using them with * and a Cisco GW, they're on PUBLIC ip's, No Firewall or
anything, I am using re-invites. Pretty standard setup. When they lockup,
you can't ping them, or get to the http interface, and I even think the IVR
stops responding
In the mean time try running asterisk with no console. This is bug #864.
Preliminary analysis shows that after a restart now, one of the
ioctl()'s performed by editline fails with -1. Ignoring the ioctl made
the CLI non-functional. Happy to get any help I can in this regard.
Mark
On Sun, 8
Hi,
Citeren CW_ASN [EMAIL PROTECTED]:
3.0.0 have some problems. Sometimes, ata answers to invite with Not found
or Busy here. This is a strange behavior.
I'm using now 2.16.2
Hm ? I have not seen this happening yet. 2.16 has alternative behaviour
regarding flash transfers...
Florian
Could you share your 3.0.0 config?
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 2:10 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
Hi,
Citeren CW_ASN [EMAIL PROTECTED]:
3.0.0 have some
That's just it, I'm not doing anything.. Just normal use.. as far as I can
tell, they end up locking up with or without anyone using them as far as I
can tell..
Thanks, Billy
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08,
http://www.nxs.net/cisco_ata_186.htm
- Original Message -
From: CW_ASN [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 12:40 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
Could you share your 3.0.0 config?
- Original Message -
Upon reading the bug I can confirm that I was doing:
asterisk -vvvc
through a ssh session/client (the Asterisk box is headless) and then for
whatever reason I would close that session/client and come back later
and start another ssh session/client and do a:
asterisk -r
and then probably a:
I will test with TOS in a8b8. All other stuff are equal in my ata.
Regards,
Gus
- Original Message -
From: Billy Huddleston [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 2:51 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
yaboo wrote:
Hi All
does anyone know anything about the nortel i2004 ip phone.
It is very hard to find out if the unit is h323 or sip.
Neither. It uses a Nortel proprietary protocol.
Can anyone please tell me info regardsing this.
All I can find on the net is manuals and what codecs it
On Sunday 01 February 2004 08:13 am, Martin wrote:
Hello.
I'm a bit puzzled at the moment. I have a x100P and TDM400 with 4 modules
(extensions). Asterisk CVS-02/01/04-06:55:30
Part of my extensions.conf says this:
; Zap Phone #1
;
exten = 204,1,Dial(Zap/2,20) ; Ring for
Hello.
Anyone tried this newer via EDEN processor ased on a new streamlined Nehemiah
core architecture released in October.
http://www.via.com.tw/en/Digital Library/PR031014EdenN.jsp
REgards...Martin
--
Please avoid sending me Word or PowerPoint attachments.
See
Has anybody
interfaced an asterisk system with a Panasonic KXTD switch?
I have invested in a
few digital phones for my house and would hate to throw them
away.
I'm thinking about
interfacing with the KXTD using their voice mail integration. Only issue is that
it is quite difficult to find
Hi,
How do
I get asterisk to use an alternate outbound provider in the event my primary IAX
provider goes down. I currently have an IAX provider that is having issues, so I
signed up with a sip provider for a backup. I added the sip provider info into
the extensions.conf file as the second
You will need to set priorities for each one.
For example:
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Playback(pstnallbusy)
exten =
_91NXXNXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
exten = _91NXXNXX,4,Congestion
Basically what happens
2nd question (should I use a separate email?)
I have Vonage service. Is it possible to end the call directly in my
asterisk system rather than in the Motorola V1005?
with any compatible FXO card, like digium X100P
Youness
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Asterisk-Users mailing
Dustin Knuttgen wrote:
Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you.
Read http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
The problem is well-known.
/Olle
___
I just downloaded and built the latest beta version of Speex, and it
appears to be the case that the 1.1.x versions do not work with
asterisk. Just to be sure, I built the 1.1.4 (both with and without the
fixed-point option) on two servers so I could test using the same codecs.
All calls made
Dialout redundancy using this method works perfect. I've been using this method for
some time now. I currently have two IAX2 providers and plan to get another backup as
well (In addition to me getting my Digium cards tomorrow that'll be another backup.)
That's great for outgoing calls, but...
Brian == Brian Capouch [EMAIL PROTECTED] writes:
Brian On a broader note, I would love to try to play with the
Brian very-low-bandwidth versions of Speex. I could have sworn I saw
Brian things on the bugtracker some weeks back on that topic, but I
Brian can't find them anymore.
It is bug
I got it working by configuring qualify in my iax.conf. I guess asterisk
didn't think the IAX provider was down until I added that line.
As for incoming I have an 800 number pointing to 2 local phone numbers. 1 on
voicepulse and 1 on voiceglo. This way if voicepulse is down it will route
the call
You're toll-free number automatically forwards to the next number if one is busy?
Cool. I wasn't sure if it would do that. I know VP reports a fast busy. Don't know
what Voiceglo reports.
What toll-free provider do you have out of curiosity?
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I see there were a couple of posts regarding installing Asterisk on a
OS X box. I have tried with no success. I am completely new to Asterisk
but think it is very cool. If someone could please guide me through an
install so that I can start to work with Asterisk, I would be very
grateful.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Tooley
Sent: Sunday, 8 February 2004 18:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PCMCIA
Does anyone know of a PCMCIA FXO card or even a USB one? I'm
looking at
building an
On Fri, Feb 06, 2004 at 08:18:21PM -0500, Andres wrote:
Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL
PROTECTED] for seqno 3 (Response)
So did it drop a few seconds into the call...like 5 - 15 seconds? If so
then you are having a problem with call
What is the very best motherboard I should use to set up my new asterisk
box?
I plan on installing about 8 pots lines.
And is X100P the only card available? I'm looking for multiple pots line
cards.
I'm trying to avoid irq conflicts as well as have a superstable box.
Thanks
George
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Geo_p15tt
Sent: Monday, 9 February 2004 14:25
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Motherboard and fxo suggestion.
What is the very best motherboard I should use to set up my
new
I have set up call queue for incoming calls. However, when I try to
transfer call after answering the queue to another station, the call is
hung up. The agent login into Asterisk by AgentCallbackLogin(). When the
agent's phone rings the agent pick up the call queue.
Is it normal behaviour that
On Sun, 8 Feb 2004, Soragan wrote:
I only have 1 server which is dual p3 1ghz.
Its mobo only has 2 64 bits pci. :(
Ohh.. You better give that to me then. I'll send you my Pentium 133 w/ 16
megs of ram. It works great with the X100P.
--
Vice President of N2Net, a New Age Consulting
I am trying to register SJPhone with my asterisk server but my SJPhone messages saying NON-INVITE transaction.. registration failed ... I dont have any FXO or FXS card installed. Just the asterisk server running on Linux and an SJPhone installed on my windows box. Some debug info:
Tilghman Lesher wrote:
On Sunday 08 February 2004 06:27, Steven Ringwald wrote:
We are using a SCSI based IBM eServer x300 for our PBX. In setting
this unit up, we used a backup machine, which
was IDE only.
The problem that we are currently experiencing is that the voicemail
Hi,
Is there a work around about Fax and Answering Machinedetection ?If not, where is the all process, at chan_zap.c ?Any site that could help ?
Actually, how is this working? When we originate a call, * just recognize if the line is busy and then creates a record for that call at CDR ? or not ?
Ohh.. You better give that to me then. I'll send you my Pentium 133 w/ 16
megs of ram. It works great with the X100P.
LOL, can your Pentium do web server, mail server with spam and virus
checking and ADSL router all together? If it can do without any performance
loses compare with mine, I'd be
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